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From: Paul Kyzivat <pkyzivat@cisco.com>
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References: <40FB0FFB97588246A1BEFB05759DC8A0038854EC@S4DE9JSAANI.ost.t-com.de> <4ADCCF93.2070208@cisco.com> <40FB0FFB97588246A1BEFB05759DC8A003885F8F@S4DE9JSAANI.ost.t-com.de> <4ADFB028.3010604@cisco.com> <4AE99CFE.9030205@sipstation.com>
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Cc: dispatch@ietf.org, R.Jesske@telekom.de, L.Liess@telekom.de, anwars@avaya.com
Subject: Re: [dispatch] New version for the Alert-Info URNs draft:	draft-liess-dispatch-alert-info-urns-00.txt
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Alan,

Sorry to be slow in responding. Comments inline.

	Thanks,
	Paul

Alan Johnston wrote:
> Paul,
> 
> Thanks for your comments on the draft.  See mine below.
> 
> Thanks,
> Alan
> 
> 
> Paul Kyzivat wrote:
>> responses inline
>>
>> L.Liess@telekom.de wrote:
>>> Paul,
>>> Thank you very much for taking the time to look and comment the draft.
>>> After the changes we did shortly before the submission deadline, there
>>> are some inconsistecies and open items. We are aware of some of them and
>>> probably there are some we are not aware of. For the problems we are
>>> aware of, there are alternative solutions. In some cases, we did not
>>> have a clear opinion which way to go and we would like to hear others
>>> opinions.
>>>
>>>
>>> Please find comments inline:
>>>     >     > This seems to be shaping up well. I think the 
>>> requirements are
>>> quite     > clear now.
>>>
>>> It's good to hear that we have achieved at least a smal stepp in the
>>> right direction :-).       >     > I don't fully understand what is 
>>> intended regarding
>>> combination of     > multiple alert URNs for a specific situation. 
>>> Section 4.5
>>> states:
>>>     >     > >    finally, a short Albanian auto-callback tone could be
>>> indicated
>>>     > >    Alert-Info: <urn:alert:service:auto-callback>,
>>>     > >    <urn:alert:tone:short>, <urn:alert:tone:al>.
>>>     >     > while section 7.3.6 states:
>>>     >     > >    Alert URN Indications from the same group should not be
>>> combined.
>>>
>>>
>>>
>>> To avoid the combinatorial explosion of IANA values as Adam pointed out
>>> in his comment
>>> http://www.ietf.org/mail-archive/web/bliss/current/msg01027.html  ,  we
>>> decided to abandon the usage of sub-indications for  the use cases we
>>> have in the draft and to register only discrete tokens as top-level
>>> registrations.  (With county-specidfic ring and busy tones, we almost
>>> got the problem Adam talked about.) The sub-indications are no longer
>>> used in the current version of the draft. 
>>
>> So, maybe that is the confusion. I didn't understand that from reading 
>> the text.
>>
>>> We thougt about changing the template in chapter 3 from
>>>                 alert-indication = top-level *("." sub-indication)
>>>         top-level        = let-dig [ *25let-dig-hyp let-dig ]
>>>         sub-indication   = let-dig [ *let-dig-hyp let-dig ]
>>>
>>> to just
>>>            alert-indication= let-dig [ *let-dig-hyp let-dig ]
>>>
>>> But we are not sure if everyone would agree with this, so we would like
>>> to have feedback from the WG on this issue.  
>>
>> I don't know yet. The alternative is apparently to have multiple URNs 
>> and interpret the combination. I'm having difficulty imagining how 
>> that would work. I have always assumed that in the end a URN would be 
>> "looked up" locally by the UA and resolved to *something* that it 
>> could render.
>>
>> I will keep an open mind, but I'll need a better explanation of how 
>> multiple URNs are to be combined into *something* that can be looked up.
>>
>>> There is an inconcistency concerning the alert indications "priority",
>>> "short" and "delayed": In chapter 4, they are now  top-level indications
>>> in the alert-category "service", therefore in the same group eith
>>> "auto-callback".     In chapter 7 there is a dedicated group for them
>>> "Additional Indications for PBX-tones" in the "tones" alert-category.
>>> We must have to decide which alternative to take. If we decide for the
>>> chapter 4 alternative, we must change either the combination rule or the
>>> example you complained about.
>>>
>>>     > I could find no definition of what constitutes a "group". The
>>> first     > thing that comes to mind is "service" vs. "tone", but the 
>>>     > above example     > violates that. It seems to me that you need 
>>> some notion of
>>> orthogonal     > properties in order to define what can/cannot be 
>>> combined.
>>>
>>> It's true, it's quite impossible to guess from the text what I meant
>>> here. A group consists of the indications in one sub-chapter 7.3.1 to
>>> 7.3.5. I just looked for a way to avoid combinations of contradictory
>>> URNs, e.g.   that <urn:alert:tone:internal> is combined with
>>> <urn:alert:tone:external>   or  <urn:alert:tone:us> is combined with
>>> <urn:alert:tone:de> .   If we agree that this kind of groups makes sense
>>> and we keep it, some definition needs to be added.  But maybe there are
>>> better proposals in the WG about how to avoid combinations of
>>> contradictory indications?
>>
>> It seems like what you must be doing is replacing a hierarchy with an 
>> N-dimensional matrix which is sparsely populated. That seems harder to 
>> deal with.
>>
> 
> Perhaps, or perhaps not. 

When I said "seems harder" that's what I meant. I'm not *sure* its 
harder. Mostly I'm trying to understand the implications.

> Remember that this is just a suggestion, 
> rather than a command about a ring or ringback.  It is perfectly 
> reasonable for UAs to simply ignore the Alert-Info.  So, the sparse 
> scenario, where a server puts in as much information as possible, 
> possibly via multiple URNs, and the UA uses the ones it knows and 
> ignores the rest.

Yup. But there really isn't much point if everybody just ignores the 
info. So what is important is to make clear what the *intended* behavior 
is, when not overridden by policy. I think that intended behavior isn't 
very clear if there is just a bunch of values for multiple "dimensions" 
and no notion of how they might then be turned into something 
perceptible by a person.

>> I have a sip pbx phone on my desk. I don't have any distinction for 
>> internal vs. external, and have no clue how "normal" would be 
>> different from one or the other of those. I don't really desire any 
>> distinction on any of those terms.
> 
> Remember, the use case for normal is driven by shared appearances - we 
> want to insert an appearance number parameter into the Alert-Info but we 
> do not want to convey any particular Alert, so we use normal as a no-op.

Ah. So maybe its just a matter of my not understanding what connotation 
you meant me to get from the name. You mean it to be like "unspecified".

>>> I would keep the two worlds disjunct. E.g. , I have a sip handy which is
>>> attached sometimes to the pbx and sometimes to my public carrier at
>>> home, I may want the pbx ringing tone to be different from the ringing
>>> tone at home. 
>>
>> I don't really have any objection to a bunch of "names" for extra 
>> arbitrary ring types. The real issue is deciding when to use one or 
>> another. Is the assumption here that these will typically be inserted 
>> by some agent acting on behalf of the phone acting on them, rather 
>> than being inserted by "the other" party? IOW, *my* pbx decides that 
>> an incoming call from you is "internal" or "external" and inserts a 
>> suitable value by its policy, rather than *your* phone deciding the 
>> call is internal or external and inserting the value that I then render.
> 
> I wonder if some of the confusion we are having is because we are not 
> cleanly separating ring tones from ringback tones.  I'm going to think a 
> bit how we would structure this if we had separate registries or trees 
> for these.  For instance, the country specification would only make 
> sense for ringback tones.  Also, things like priority would only apply 
> to ring tones, etc.

My original thinking was that there need not be any difference.
But maybe there is, at least in common practice.

While it isn't common practice, I can imagine it might be interesting to 
have the ring tone match the country of origin of the caller. But of 
course many would not like that, and would insist that their choice of 
ring tone(s) be honored.

I agree its worth considering splitting up the namespace into separate 
sections for ring alerts and ringback alerts.

	Thanks,
	Paul

From gonzalo.camarillo@ericsson.com  Mon Nov  2 10:22:56 2009
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Date: Mon, 02 Nov 2009 20:23:12 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [dispatch] Liaison from OMA on SIP Push
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Folks,

here you have an LS we have received from OMA:

https://datatracker.ietf.org/documents/LIAISON/file726.doc

It relates to the following draft:

http://tools.ietf.org/html/draft-mdolly-dispatch-oma-push-00

Cheers,

Gonzalo
DISPATCH co-chair


From R.Jesske@telekom.de  Tue Nov  3 00:51:02 2009
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Subject: [dispatch] draft-jesske-dispatch-reason-in-responses-00, provisional Responses 18x
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Dear all,
Due to some comments I have extended the =
draft-jesske-dispatch-reason-in-responses-00 for the next planned =
version.
It was asked also to allow the inclusion of the Reason header within 18x =
Responses.
So I plan to incorporate the following text within the next 01 Version =
as follows:


4.  Overall Applicability

   The SIP procedures specified in this document are foreseen for
   networks providing simulation services and/or interworking to the
   PSTN/ISDN.

   The document is describing the use of the Reason header in SIP
   responses.  These procedures are only valuable if the reason
   contained in the element "protocol" is "Q.850".  A inclusion of a SIP
   reason (protocol=3D"SIP") is not helpful due to the fact that the
   response already provides the SIP reason.  The Release Causes are
   described within [Q.850].  (Note: The ETSI specifications can be
   downloaded under http://pda.etsi.org/pda/queryform.asp free of
   charge.)

   The appearance of the Reason header is applicable to final responses
   4xx, 5xx and 6xx and in addition for 199 Responses as defined within
   [I-D.ietf-sipcore-199].  Also the Reason header is applicable to
   provisional responses 18x which are generated by an interworking
   gateway from ISUP to SIP.

Comments?=20


Mit freundlichen Gr=FC=DFen; With Best Regards=20
Roland Jesske


Deutsche Telekom Netzproduktion GmbH=20
Zentrum Technik Einf=FChrung=20
Roland Jesske
Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt=20
+49 6151628-2766 (Tel.)=20
+49 521 9210-3753 (Fax)=20
+49 171 8618-445 (Mobil)=20
http://www.telekom.com=20

Erleben, was verbindet.

Deutsche Telekom Netzproduktion GmbH=20
Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)=20
Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert =
Matheis, Klaus Peren=20
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USt-IdNr.: DE 814645262




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<BODY>
<!-- Converted from text/rtf format -->

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

<BR><FONT SIZE=3D2 FACE=3D"Arial">Due to some comments I have extended =
the draft-jesske-dispatch-reason-in-responses-00 for the next planned =
version.</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Arial">It was asked also to allow the =
inclusion of the Reason header within 18x Responses.</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Arial">So I plan to incorporate the following =
text within the next 01 Version as follows:</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">4.&nbsp; Overall =
Applicability</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; The SIP procedures =
specified in this document are foreseen for</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; networks providing =
simulation services and/or interworking to the</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; PSTN/ISDN.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; The document is =
describing the use of the Reason header in SIP</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; responses.&nbsp; =
These procedures are only valuable if the reason</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; contained in the =
element &quot;protocol&quot; is &quot;Q.850&quot;.&nbsp; A inclusion of =
a SIP</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; reason =
(protocol=3D&quot;SIP&quot;) is not helpful due to the fact that =
the</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; response already =
provides the SIP reason.&nbsp; The Release Causes are</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; described within =
[Q.850].&nbsp; (Note: The ETSI specifications can be</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; downloaded under =
</FONT><A HREF=3D"http://pda.etsi.org/pda/queryform.asp"><U><FONT =
COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Courier =
New">http://pda.etsi.org/pda/queryform.asp</FONT></U></A><FONT SIZE=3D2 =
FACE=3D"Courier New"> free of</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; charge.)</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; The appearance of =
the Reason header is applicable to final responses</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; 4xx, 5xx and 6xx =
and in addition for 199 Responses as defined within</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; =
[I-D.ietf-sipcore-199].&nbsp; Also the Reason header is applicable =
to</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; provisional =
responses 18x which are generated by an interworking</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; gateway from ISUP =
to SIP.</FONT>
</P>

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

<P><FONT SIZE=3D2 FACE=3D"Arial">Mit freundlichen Gr=FC=DFen; With Best =
Regards<BR>
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<BR>
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<FONT COLOR=3D"#808080" SIZE=3D1 FACE=3D"Arial">Deutsche Telekom =
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Zentrum Technik Einf=FChrung<BR>
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From alan@sipstation.com  Tue Nov  3 13:08:53 2009
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Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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All,

Here is some updated charter text for CCUS.  Comments are most welcome.

Thanks,
Alan

- - - - -

The Call Control UUI for SIP  (CCUS) working group is chartered to 
define a SIP mechanism for transporting call control related 
user-to-user (UUI) information between User Agents.

The mechanism developed in this working group is generally appropriate 
in the following situations:
1. The information is generated and consumed by an application using SIP 
during session setup but the application is 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. Multi-vendor interoperability is important.

This mechanism is not appropriate in the following situations:
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.

User data of the mechanism will be clearly marked with the application, 
encoding, semantics, and contents of the data, allowing policy to be 
applied by UAs.  The working group will define the information which 
each application must specify in a standards-track document to utilize 
the mechanism.

One important application of this mechanism is interworking with the 
ISDN User to User Information Service.  This application defined by 
ITU-T Q.931, is extensively deployed in the PSTN today, supporting such 
applications as contact centers, call centers, and automatic call 
distributors (ACDs).  A major barrier to the movement of these 
applications to SIP is the lack of a standard mechanism to transport 
this information in SIP.   For interworking with ISDN, minimal 
information about the content of the UUI is available.  In this case 
only, the content will just indicate ISDN UUI Service 1 interworking 
rather than the actual content.

Call control UUI is user information conveyed between user agents during 
call control operations.  As a result, the information must be conveyed 
with the INVITE transaction, and must survive proxy retargeting, 
redirection, and transfers.  The mechanism must utilize a minimum of SIP 
extensions as it will need to be supported by many simple SIP user 
agents such as PSTN gateways.  The mechanism must interwork with the 
existing ISDN service but should also be extensible for use by other 
applications and non-ISDN protocols.

While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control UUI 
can be generated by a number of protocols that are not ISUP, and as such 
it is architecturally unreasonable to interwork these protocols at the 
SIP / ISUP gateway. That is, existing SIP-T approaches defined in 
RFC3372 do not meet the requirements.

Mechanisms based on the SIP INFO method, RFC2976, will not be considered 
by the working group since the UUI contents carry information that must 
be conveyed during session setup at the user agent - the information 
must be conveyed with the INVITE transaction.  The information must be 
passed with the session setup request (INVITE), responses to that 
INVITE, or session termination requests.  As a result, it is not 
possible to use INFO in these cases.

The group will produce:

- A problem statement and requirements document for implementing a SIP 
call control UUI mechanism

- A specification of the SIP extension to best meet those requirements.

Goals and Milestones
====================

Mar 10 - Problem statement and requirements document to IESG 
(Informational)
Aug 10 - SIP call control UUI specification to IESG (PS)


From Leon.Portman@nice.com  Tue Nov  3 23:58:41 2009
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From: Leon Portman <Leon.Portman@nice.com>
To: Alan Johnston <alan@sipstation.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 4 Nov 2009 09:58:56 +0200
Thread-Topic: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
Thread-Index: Acpcyf9cQc7Gv/u7RAyfnE5VWPs2QQAWmnAQ
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Only one comment, does it also supposed to support re-invite (for SRTP re-k=
eying or media updates) scenarios as well?

Thanks

Leon

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Alan Johnston
Sent: Tuesday, November 03, 2009 11:09 PM
To: dispatch@ietf.org
Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP

All,

Here is some updated charter text for CCUS.  Comments are most welcome.

Thanks,
Alan

- - - - -

The Call Control UUI for SIP  (CCUS) working group is chartered to=20
define a SIP mechanism for transporting call control related=20
user-to-user (UUI) information between User Agents.

The mechanism developed in this working group is generally appropriate=20
in the following situations:
1. The information is generated and consumed by an application using SIP=20
during session setup but the application is 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=20
retargeting.
5. SIP elements may need to apply policy about passing and screening the=20
information.
6. Multi-vendor interoperability is important.

This mechanism is not appropriate in the following situations:
1. SIP behavior is changed.
2. The information is generated and consumed at the SIP layer by SIP=20
elements.
3. SIP elements besides the UAC and UAS might be interested and consume=20
the information.
4. There are specific privacy or cross-RAI issues involved with the=20
information being transported.

User data of the mechanism will be clearly marked with the application,=20
encoding, semantics, and contents of the data, allowing policy to be=20
applied by UAs.  The working group will define the information which=20
each application must specify in a standards-track document to utilize=20
the mechanism.

One important application of this mechanism is interworking with the=20
ISDN User to User Information Service.  This application defined by=20
ITU-T Q.931, is extensively deployed in the PSTN today, supporting such=20
applications as contact centers, call centers, and automatic call=20
distributors (ACDs).  A major barrier to the movement of these=20
applications to SIP is the lack of a standard mechanism to transport=20
this information in SIP.   For interworking with ISDN, minimal=20
information about the content of the UUI is available.  In this case=20
only, the content will just indicate ISDN UUI Service 1 interworking=20
rather than the actual content.

Call control UUI is user information conveyed between user agents during=20
call control operations.  As a result, the information must be conveyed=20
with the INVITE transaction, and must survive proxy retargeting,=20
redirection, and transfers.  The mechanism must utilize a minimum of SIP=20
extensions as it will need to be supported by many simple SIP user=20
agents such as PSTN gateways.  The mechanism must interwork with the=20
existing ISDN service but should also be extensible for use by other=20
applications and non-ISDN protocols.

While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control UUI=20
can be generated by a number of protocols that are not ISUP, and as such=20
it is architecturally unreasonable to interwork these protocols at the=20
SIP / ISUP gateway. That is, existing SIP-T approaches defined in=20
RFC3372 do not meet the requirements.

Mechanisms based on the SIP INFO method, RFC2976, will not be considered=20
by the working group since the UUI contents carry information that must=20
be conveyed during session setup at the user agent - the information=20
must be conveyed with the INVITE transaction.  The information must be=20
passed with the session setup request (INVITE), responses to that=20
INVITE, or session termination requests.  As a result, it is not=20
possible to use INFO in these cases.

The group will produce:

- A problem statement and requirements document for implementing a SIP=20
call control UUI mechanism

- A specification of the SIP extension to best meet those requirements.

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

Mar 10 - Problem statement and requirements document to IESG=20
(Informational)
Aug 10 - SIP call control UUI specification to IESG (PS)

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

From sanjsinh@cisco.com  Wed Nov  4 01:18:14 2009
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From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: "Leon Portman" <Leon.Portman@nice.com>, "Alan Johnston" <alan@sipstation.com>, <dispatch@ietf.org>
X-OriginalArrivalTime: 04 Nov 2009 09:18:33.0269 (UTC) FILETIME=[C7EF8650:01CA5D2F]
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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I think UUI is carried during session setup and teardown, and hence
re-Invite scenario may not be applicable here.

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Leon Portman
Sent: Wednesday, November 04, 2009 1:29 PM
To: Alan Johnston; dispatch@ietf.org
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for
SIP

Only one comment, does it also supposed to support re-invite (for SRTP
re-keying or media updates) scenarios as well?

Thanks

Leon

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Alan Johnston
Sent: Tuesday, November 03, 2009 11:09 PM
To: dispatch@ietf.org
Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP

All,

Here is some updated charter text for CCUS.  Comments are most welcome.

Thanks,
Alan

- - - - -

The Call Control UUI for SIP  (CCUS) working group is chartered to=20
define a SIP mechanism for transporting call control related=20
user-to-user (UUI) information between User Agents.

The mechanism developed in this working group is generally appropriate=20
in the following situations:
1. The information is generated and consumed by an application using SIP

during session setup but the application is 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=20
retargeting.
5. SIP elements may need to apply policy about passing and screening the

information.
6. Multi-vendor interoperability is important.

This mechanism is not appropriate in the following situations:
1. SIP behavior is changed.
2. The information is generated and consumed at the SIP layer by SIP=20
elements.
3. SIP elements besides the UAC and UAS might be interested and consume=20
the information.
4. There are specific privacy or cross-RAI issues involved with the=20
information being transported.

User data of the mechanism will be clearly marked with the application,=20
encoding, semantics, and contents of the data, allowing policy to be=20
applied by UAs.  The working group will define the information which=20
each application must specify in a standards-track document to utilize=20
the mechanism.

One important application of this mechanism is interworking with the=20
ISDN User to User Information Service.  This application defined by=20
ITU-T Q.931, is extensively deployed in the PSTN today, supporting such=20
applications as contact centers, call centers, and automatic call=20
distributors (ACDs).  A major barrier to the movement of these=20
applications to SIP is the lack of a standard mechanism to transport=20
this information in SIP.   For interworking with ISDN, minimal=20
information about the content of the UUI is available.  In this case=20
only, the content will just indicate ISDN UUI Service 1 interworking=20
rather than the actual content.

Call control UUI is user information conveyed between user agents during

call control operations.  As a result, the information must be conveyed=20
with the INVITE transaction, and must survive proxy retargeting,=20
redirection, and transfers.  The mechanism must utilize a minimum of SIP

extensions as it will need to be supported by many simple SIP user=20
agents such as PSTN gateways.  The mechanism must interwork with the=20
existing ISDN service but should also be extensible for use by other=20
applications and non-ISDN protocols.

While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control UUI

can be generated by a number of protocols that are not ISUP, and as such

it is architecturally unreasonable to interwork these protocols at the=20
SIP / ISUP gateway. That is, existing SIP-T approaches defined in=20
RFC3372 do not meet the requirements.

Mechanisms based on the SIP INFO method, RFC2976, will not be considered

by the working group since the UUI contents carry information that must=20
be conveyed during session setup at the user agent - the information=20
must be conveyed with the INVITE transaction.  The information must be=20
passed with the session setup request (INVITE), responses to that=20
INVITE, or session termination requests.  As a result, it is not=20
possible to use INFO in these cases.

The group will produce:

- A problem statement and requirements document for implementing a SIP=20
call control UUI mechanism

- A specification of the SIP extension to best meet those requirements.

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

Mar 10 - Problem statement and requirements document to IESG=20
(Informational)
Aug 10 - SIP call control UUI specification to IESG (PS)

_______________________________________________
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 Leon.Portman@nice.com  Wed Nov  4 01:31:29 2009
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From: Leon Portman <Leon.Portman@nice.com>
To: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>, Alan Johnston <alan@sipstation.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 4 Nov 2009 11:31:45 +0200
Thread-Topic: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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Actually I was thinking that if UUI was presented during the original INVIT=
E, if it is going to be required to be presented during RE_INVITES as well,=
 especially if it represents some client state information

Leon

-----Original Message-----
From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com]=20
Sent: Wednesday, November 04, 2009 11:18 AM
To: Leon Portman; Alan Johnston; dispatch@ietf.org
Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI for SI=
P

I think UUI is carried during session setup and teardown, and hence
re-Invite scenario may not be applicable here.

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Leon Portman
Sent: Wednesday, November 04, 2009 1:29 PM
To: Alan Johnston; dispatch@ietf.org
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for
SIP

Only one comment, does it also supposed to support re-invite (for SRTP
re-keying or media updates) scenarios as well?

Thanks

Leon

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Alan Johnston
Sent: Tuesday, November 03, 2009 11:09 PM
To: dispatch@ietf.org
Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP

All,

Here is some updated charter text for CCUS.  Comments are most welcome.

Thanks,
Alan

- - - - -

The Call Control UUI for SIP  (CCUS) working group is chartered to=20
define a SIP mechanism for transporting call control related=20
user-to-user (UUI) information between User Agents.

The mechanism developed in this working group is generally appropriate=20
in the following situations:
1. The information is generated and consumed by an application using SIP

during session setup but the application is 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=20
retargeting.
5. SIP elements may need to apply policy about passing and screening the

information.
6. Multi-vendor interoperability is important.

This mechanism is not appropriate in the following situations:
1. SIP behavior is changed.
2. The information is generated and consumed at the SIP layer by SIP=20
elements.
3. SIP elements besides the UAC and UAS might be interested and consume=20
the information.
4. There are specific privacy or cross-RAI issues involved with the=20
information being transported.

User data of the mechanism will be clearly marked with the application,=20
encoding, semantics, and contents of the data, allowing policy to be=20
applied by UAs.  The working group will define the information which=20
each application must specify in a standards-track document to utilize=20
the mechanism.

One important application of this mechanism is interworking with the=20
ISDN User to User Information Service.  This application defined by=20
ITU-T Q.931, is extensively deployed in the PSTN today, supporting such=20
applications as contact centers, call centers, and automatic call=20
distributors (ACDs).  A major barrier to the movement of these=20
applications to SIP is the lack of a standard mechanism to transport=20
this information in SIP.   For interworking with ISDN, minimal=20
information about the content of the UUI is available.  In this case=20
only, the content will just indicate ISDN UUI Service 1 interworking=20
rather than the actual content.

Call control UUI is user information conveyed between user agents during

call control operations.  As a result, the information must be conveyed=20
with the INVITE transaction, and must survive proxy retargeting,=20
redirection, and transfers.  The mechanism must utilize a minimum of SIP

extensions as it will need to be supported by many simple SIP user=20
agents such as PSTN gateways.  The mechanism must interwork with the=20
existing ISDN service but should also be extensible for use by other=20
applications and non-ISDN protocols.

While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control UUI

can be generated by a number of protocols that are not ISUP, and as such

it is architecturally unreasonable to interwork these protocols at the=20
SIP / ISUP gateway. That is, existing SIP-T approaches defined in=20
RFC3372 do not meet the requirements.

Mechanisms based on the SIP INFO method, RFC2976, will not be considered

by the working group since the UUI contents carry information that must=20
be conveyed during session setup at the user agent - the information=20
must be conveyed with the INVITE transaction.  The information must be=20
passed with the session setup request (INVITE), responses to that=20
INVITE, or session termination requests.  As a result, it is not=20
possible to use INFO in these cases.

The group will produce:

- A problem statement and requirements document for implementing a SIP=20
call control UUI mechanism

- A specification of the SIP extension to best meet those requirements.

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

Mar 10 - Problem statement and requirements document to IESG=20
(Informational)
Aug 10 - SIP call control UUI specification to IESG (PS)

_______________________________________________
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 john.elwell@siemens-enterprise.com  Wed Nov  4 03:43:07 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Alan Johnston <alan@sipstation.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 4 Nov 2009 12:43:41 +0100
Thread-Topic: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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Alan,

Do we need to make any statement about applicability of the mechanism to 3P=
CC situations? For example, UA A has exchanged UUI with UA B, but then is 3=
PCC-transferred to UA C - does UA A need to exchange UUI with UA C? From UA=
 A's perspective it would be mid-call.

John


> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Alan Johnston
> Sent: 03 November 2009 21:09
> To: dispatch@ietf.org
> Subject: [dispatch] Updated CCUS Charter Text - Call Control=20
> UUI for SIP
>=20
> All,
>=20
> Here is some updated charter text for CCUS.  Comments are=20
> most welcome.
>=20
> Thanks,
> Alan
>=20
> - - - - -
>=20
> The Call Control UUI for SIP  (CCUS) working group is chartered to=20
> define a SIP mechanism for transporting call control related=20
> user-to-user (UUI) information between User Agents.
>=20
> The mechanism developed in this working group is generally=20
> appropriate=20
> in the following situations:
> 1. The information is generated and consumed by an=20
> application using SIP=20
> during session setup but the application is not necessarily=20
> 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,=20
> redirection, and=20
> retargeting.
> 5. SIP elements may need to apply policy about passing and=20
> screening the=20
> information.
> 6. Multi-vendor interoperability is important.
>=20
> This mechanism is not appropriate in the following situations:
> 1. SIP behavior is changed.
> 2. The information is generated and consumed at the SIP layer by SIP=20
> elements.
> 3. SIP elements besides the UAC and UAS might be interested=20
> and consume=20
> the information.
> 4. There are specific privacy or cross-RAI issues involved with the=20
> information being transported.
>=20
> User data of the mechanism will be clearly marked with the=20
> application,=20
> encoding, semantics, and contents of the data, allowing policy to be=20
> applied by UAs.  The working group will define the information which=20
> each application must specify in a standards-track document=20
> to utilize=20
> the mechanism.
>=20
> One important application of this mechanism is interworking with the=20
> ISDN User to User Information Service.  This application defined by=20
> ITU-T Q.931, is extensively deployed in the PSTN today,=20
> supporting such=20
> applications as contact centers, call centers, and automatic call=20
> distributors (ACDs).  A major barrier to the movement of these=20
> applications to SIP is the lack of a standard mechanism to transport=20
> this information in SIP.   For interworking with ISDN, minimal=20
> information about the content of the UUI is available.  In this case=20
> only, the content will just indicate ISDN UUI Service 1 interworking=20
> rather than the actual content.
>=20
> Call control UUI is user information conveyed between user=20
> agents during=20
> call control operations.  As a result, the information must=20
> be conveyed=20
> with the INVITE transaction, and must survive proxy retargeting,=20
> redirection, and transfers.  The mechanism must utilize a=20
> minimum of SIP=20
> extensions as it will need to be supported by many simple SIP user=20
> agents such as PSTN gateways.  The mechanism must interwork with the=20
> existing ISDN service but should also be extensible for use by other=20
> applications and non-ISDN protocols.
>=20
> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call=20
> control UUI=20
> can be generated by a number of protocols that are not ISUP,=20
> and as such=20
> it is architecturally unreasonable to interwork these=20
> protocols at the=20
> SIP / ISUP gateway. That is, existing SIP-T approaches defined in=20
> RFC3372 do not meet the requirements.
>=20
> Mechanisms based on the SIP INFO method, RFC2976, will not be=20
> considered=20
> by the working group since the UUI contents carry information=20
> that must=20
> be conveyed during session setup at the user agent - the information=20
> must be conveyed with the INVITE transaction.  The=20
> information must be=20
> passed with the session setup request (INVITE), responses to that=20
> INVITE, or session termination requests.  As a result, it is not=20
> possible to use INFO in these cases.
>=20
> The group will produce:
>=20
> - A problem statement and requirements document for=20
> implementing a SIP=20
> call control UUI mechanism
>=20
> - A specification of the SIP extension to best meet those=20
> requirements.
>=20
> Goals and Milestones
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> Mar 10 - Problem statement and requirements document to IESG=20
> (Informational)
> Aug 10 - SIP call control UUI specification to IESG (PS)
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From sanjsinh@cisco.com  Wed Nov  4 03:44:29 2009
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From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: "Leon Portman" <Leon.Portman@nice.com>, "Alan Johnston" <alan@sipstation.com>, <dispatch@ietf.org>
X-OriginalArrivalTime: 04 Nov 2009 11:44:35.0575 (UTC) FILETIME=[2EAD5070:01CA5D44]
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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Do you mean same IE that was present in initial INVITE needs to be
presented in Re-Invite as well?=20

If the information does not change, then I do not see any value in
carrying the same payload in re-Invite. But if there is a new IE, then
yeah it can be presented in re-Invite.
I am not sure if there can be a new UU IE after the call is setup and
before call termination.

Sanjay

-----Original Message-----
From: Leon Portman [mailto:Leon.Portman@nice.com]=20
Sent: Wednesday, November 04, 2009 3:02 PM
To: Sanjay Sinha (sanjsinh); Alan Johnston; dispatch@ietf.org
Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI for
SIP

Actually I was thinking that if UUI was presented during the original
INVITE, if it is going to be required to be presented during RE_INVITES
as well, especially if it represents some client state information

Leon

-----Original Message-----
From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com]=20
Sent: Wednesday, November 04, 2009 11:18 AM
To: Leon Portman; Alan Johnston; dispatch@ietf.org
Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI for
SIP

I think UUI is carried during session setup and teardown, and hence
re-Invite scenario may not be applicable here.

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Leon Portman
Sent: Wednesday, November 04, 2009 1:29 PM
To: Alan Johnston; dispatch@ietf.org
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for
SIP

Only one comment, does it also supposed to support re-invite (for SRTP
re-keying or media updates) scenarios as well?

Thanks

Leon

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Alan Johnston
Sent: Tuesday, November 03, 2009 11:09 PM
To: dispatch@ietf.org
Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP

All,

Here is some updated charter text for CCUS.  Comments are most welcome.

Thanks,
Alan

- - - - -

The Call Control UUI for SIP  (CCUS) working group is chartered to=20
define a SIP mechanism for transporting call control related=20
user-to-user (UUI) information between User Agents.

The mechanism developed in this working group is generally appropriate=20
in the following situations:
1. The information is generated and consumed by an application using SIP

during session setup but the application is 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=20
retargeting.
5. SIP elements may need to apply policy about passing and screening the

information.
6. Multi-vendor interoperability is important.

This mechanism is not appropriate in the following situations:
1. SIP behavior is changed.
2. The information is generated and consumed at the SIP layer by SIP=20
elements.
3. SIP elements besides the UAC and UAS might be interested and consume=20
the information.
4. There are specific privacy or cross-RAI issues involved with the=20
information being transported.

User data of the mechanism will be clearly marked with the application,=20
encoding, semantics, and contents of the data, allowing policy to be=20
applied by UAs.  The working group will define the information which=20
each application must specify in a standards-track document to utilize=20
the mechanism.

One important application of this mechanism is interworking with the=20
ISDN User to User Information Service.  This application defined by=20
ITU-T Q.931, is extensively deployed in the PSTN today, supporting such=20
applications as contact centers, call centers, and automatic call=20
distributors (ACDs).  A major barrier to the movement of these=20
applications to SIP is the lack of a standard mechanism to transport=20
this information in SIP.   For interworking with ISDN, minimal=20
information about the content of the UUI is available.  In this case=20
only, the content will just indicate ISDN UUI Service 1 interworking=20
rather than the actual content.

Call control UUI is user information conveyed between user agents during

call control operations.  As a result, the information must be conveyed=20
with the INVITE transaction, and must survive proxy retargeting,=20
redirection, and transfers.  The mechanism must utilize a minimum of SIP

extensions as it will need to be supported by many simple SIP user=20
agents such as PSTN gateways.  The mechanism must interwork with the=20
existing ISDN service but should also be extensible for use by other=20
applications and non-ISDN protocols.

While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control UUI

can be generated by a number of protocols that are not ISUP, and as such

it is architecturally unreasonable to interwork these protocols at the=20
SIP / ISUP gateway. That is, existing SIP-T approaches defined in=20
RFC3372 do not meet the requirements.

Mechanisms based on the SIP INFO method, RFC2976, will not be considered

by the working group since the UUI contents carry information that must=20
be conveyed during session setup at the user agent - the information=20
must be conveyed with the INVITE transaction.  The information must be=20
passed with the session setup request (INVITE), responses to that=20
INVITE, or session termination requests.  As a result, it is not=20
possible to use INFO in these cases.

The group will produce:

- A problem statement and requirements document for implementing a SIP=20
call control UUI mechanism

- A specification of the SIP extension to best meet those requirements.

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

Mar 10 - Problem statement and requirements document to IESG=20
(Informational)
Aug 10 - SIP call control UUI specification to IESG (PS)

_______________________________________________
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From pkyzivat@cisco.com  Wed Nov  4 05:56:31 2009
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Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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Leon Portman wrote:
> Actually I was thinking that if UUI was presented during the original INVITE, if it is going to be required to be presented during RE_INVITES as well, especially if it represents some client state information

Can you explain more about this?

The charter seems to spend its time explaining that this is governed by 
the needs for session setup. If this is a mechanism that is also 
intended for use after the session is established, then I think more 
explanation is required.

As described, this is just plain tunneling, pure and simple.
And INFO (expecially with the new Info-Package work) is yet another 
mechanism for tunneling. The rationale for not using INFO for UUI is 
that it is needed during session establishment. Clearly that doesn't 
apply after the session is established.

Of course its a mess to have two mechanisms with overlapping purposes. 
It would be silly to use two mechanisms to transport the same 
information, depending on when in the lifetime of the session the info 
was being transported. But I'm having difficulty drawing a clear 
distinction between the applicability of the two mechanisms.

	Thanks,
	Paul

> Leon
> 
> -----Original Message-----
> From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com] 
> Sent: Wednesday, November 04, 2009 11:18 AM
> To: Leon Portman; Alan Johnston; dispatch@ietf.org
> Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
> 
> I think UUI is carried during session setup and teardown, and hence
> re-Invite scenario may not be applicable here.
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Leon Portman
> Sent: Wednesday, November 04, 2009 1:29 PM
> To: Alan Johnston; dispatch@ietf.org
> Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for
> SIP
> 
> Only one comment, does it also supposed to support re-invite (for SRTP
> re-keying or media updates) scenarios as well?
> 
> Thanks
> 
> Leon
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Alan Johnston
> Sent: Tuesday, November 03, 2009 11:09 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
> 
> All,
> 
> Here is some updated charter text for CCUS.  Comments are most welcome.
> 
> Thanks,
> Alan
> 
> - - - - -
> 
> The Call Control UUI for SIP  (CCUS) working group is chartered to 
> define a SIP mechanism for transporting call control related 
> user-to-user (UUI) information between User Agents.
> 
> The mechanism developed in this working group is generally appropriate 
> in the following situations:
> 1. The information is generated and consumed by an application using SIP
> 
> during session setup but the application is 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. Multi-vendor interoperability is important.
> 
> This mechanism is not appropriate in the following situations:
> 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.
> 
> User data of the mechanism will be clearly marked with the application, 
> encoding, semantics, and contents of the data, allowing policy to be 
> applied by UAs.  The working group will define the information which 
> each application must specify in a standards-track document to utilize 
> the mechanism.
> 
> One important application of this mechanism is interworking with the 
> ISDN User to User Information Service.  This application defined by 
> ITU-T Q.931, is extensively deployed in the PSTN today, supporting such 
> applications as contact centers, call centers, and automatic call 
> distributors (ACDs).  A major barrier to the movement of these 
> applications to SIP is the lack of a standard mechanism to transport 
> this information in SIP.   For interworking with ISDN, minimal 
> information about the content of the UUI is available.  In this case 
> only, the content will just indicate ISDN UUI Service 1 interworking 
> rather than the actual content.
> 
> Call control UUI is user information conveyed between user agents during
> 
> call control operations.  As a result, the information must be conveyed 
> with the INVITE transaction, and must survive proxy retargeting, 
> redirection, and transfers.  The mechanism must utilize a minimum of SIP
> 
> extensions as it will need to be supported by many simple SIP user 
> agents such as PSTN gateways.  The mechanism must interwork with the 
> existing ISDN service but should also be extensible for use by other 
> applications and non-ISDN protocols.
> 
> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control UUI
> 
> can be generated by a number of protocols that are not ISUP, and as such
> 
> it is architecturally unreasonable to interwork these protocols at the 
> SIP / ISUP gateway. That is, existing SIP-T approaches defined in 
> RFC3372 do not meet the requirements.
> 
> Mechanisms based on the SIP INFO method, RFC2976, will not be considered
> 
> by the working group since the UUI contents carry information that must 
> be conveyed during session setup at the user agent - the information 
> must be conveyed with the INVITE transaction.  The information must be 
> passed with the session setup request (INVITE), responses to that 
> INVITE, or session termination requests.  As a result, it is not 
> possible to use INFO in these cases.
> 
> The group will produce:
> 
> - A problem statement and requirements document for implementing a SIP 
> call control UUI mechanism
> 
> - A specification of the SIP extension to best meet those requirements.
> 
> Goals and Milestones
> ====================
> 
> Mar 10 - Problem statement and requirements document to IESG 
> (Informational)
> Aug 10 - SIP call control UUI specification to IESG (PS)
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From john.elwell@siemens-enterprise.com  Wed Nov  4 05:57:14 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 4 Nov 2009 14:57:31 +0100
Thread-Topic: Draft charter for domain registration work
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Below is an initial proposal for chartering this work:

John


Domain REGistration for SIP (DREGS)

Description of Working Group

The Domain REGistration for SIP (DREGS) working group is chartered to speci=
fy a means by which an entity authoritative for SIP URIs in a given domain =
can dynamically register reachability information for that domain with anot=
her domain.
The SIP protocol [RFC 3261 and its extensions] supports multiple means of o=
btaining the connection information necessary to deliver out-of-dialog SIP =
requests to their intended targets.  When a SIP Proxy needs to send a reque=
st to a target address-of-record (AoR) within its domain, it can use a loca=
tion service to obtain the registered contact URI, together with any associ=
ated path information [RFC 3327], and build a route set to reach the target=
 UA(s).  The SIP REGISTER method can be used to register contact URIs and p=
ath information. SIP-outbound [RFC 5626] enhances this mechanism to cater f=
or UAs behind NATs and firewalls. When a SIP UA or Proxy needs to send a re=
quest to a target for which it is not authoritative, the UA/Proxy can use R=
FC3263 procedures for using DNS to resolve the next-hop connection informat=
ion.
It is not uncommon, however, to use a dynamic registration mechanism to pro=
vide reachability information from a first domain to a second domain, to en=
able the second domain to deliver out-of-dialog requests targeted at the fi=
rst domain. A typical example is a SIP-PBX in a small/medium business that =
needs to be reachable from a SIP Service Provider (SSP). Even if the SIP-PB=
X has its own domain name, it is expected that in most situations this will=
 be a sub-domain of the service provider's domain name.  However, not every=
 service provider wishes to maintain DNS infrastructure supporting dynamic =
registration, and not all SIP-PBXs support a dynamic DNS client. Furthermor=
e, typical SSP deployments are such that a SIP proxy has to be traversed in=
 order reach the SIP-PBX, and DNS does not provide a means of discovering t=
his.
On the other hand, because a single SSP may support multiple thousands of s=
uch SMB SIP-PBX's, it is impractical and cost-prohibitive to manually provi=
sion their IP addresses in every SIP node along paths to those SIP-PBXs.  F=
urthermore, IP addresses can be dynamically assigned and therefore can pote=
ntially change relatively frequently.

In practice, a dynamic reachability mechanism is used, based on the SIP REG=
ISTER method. Effectively a single REGISTER request registers the domain of=
 the SIP-PBX, so that any out-of-dialog request targeted at a SIP URI in th=
e SIP-PBX's domain can be delivered from the SSP to the SIP-PBX. However, i=
mplementations of this mechanism vary in details, leading to interoperabili=
ty issues between SIP-PBXs and SSPs, and the need for equipment to support =
different variants.

The task of this working group is therefore to standardize a domain registr=
ation mechanism for SIP, based on the REGISTER method. The solution will ut=
ilize existing SIP mechanisms to the extent possible, although it is antici=
pated that small protocol extensions are likely to be required, and hence a=
 standards track (rather than BCP) deliverable is expected. The solution wi=
ll accommodate existing SIP extensions relating to registration (e.g., Path=
, Service-Route [RFC 3608] and SIP-outbound) by ensuring that they are not =
precluded from use in the context of domain registration, if they are deeme=
d relevant. The solution will incorporate a compatibility mechanism to ensu=
re a deterministic outcome when interworking with entities that do not supp=
ort domain registration.

Goals and Milestones
Jan 2010	Domain registration specification WGLC
Feb 2010	Domain registration specification to IESG (PS)



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Cc: Leon Portman <Leon.Portman@nice.com>, dispatch@ietf.org
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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I always use 3pcc as a test case, because it presents problems almost 
every time.

Of course the problem with 3pcc is that one "side" may remain the same, 
while the other "side" is replaced. The replacement typically has no 
knowledge of the history of what went on before it joined. So there 
usually needs to be some mechanism for it to be clued in about that.

I don't know if the same is true here or not. It depends on whether the 
UUI information is viewed as transient, or as state of the session.

	Thanks,
	Paul

Sanjay Sinha (sanjsinh) wrote:
> Do you mean same IE that was present in initial INVITE needs to be
> presented in Re-Invite as well? 
> 
> If the information does not change, then I do not see any value in
> carrying the same payload in re-Invite. But if there is a new IE, then
> yeah it can be presented in re-Invite.
> I am not sure if there can be a new UU IE after the call is setup and
> before call termination.
> 
> Sanjay
> 
> -----Original Message-----
> From: Leon Portman [mailto:Leon.Portman@nice.com] 
> Sent: Wednesday, November 04, 2009 3:02 PM
> To: Sanjay Sinha (sanjsinh); Alan Johnston; dispatch@ietf.org
> Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI for
> SIP
> 
> Actually I was thinking that if UUI was presented during the original
> INVITE, if it is going to be required to be presented during RE_INVITES
> as well, especially if it represents some client state information
> 
> Leon
> 
> -----Original Message-----
> From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com] 
> Sent: Wednesday, November 04, 2009 11:18 AM
> To: Leon Portman; Alan Johnston; dispatch@ietf.org
> Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI for
> SIP
> 
> I think UUI is carried during session setup and teardown, and hence
> re-Invite scenario may not be applicable here.
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Leon Portman
> Sent: Wednesday, November 04, 2009 1:29 PM
> To: Alan Johnston; dispatch@ietf.org
> Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for
> SIP
> 
> Only one comment, does it also supposed to support re-invite (for SRTP
> re-keying or media updates) scenarios as well?
> 
> Thanks
> 
> Leon
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Alan Johnston
> Sent: Tuesday, November 03, 2009 11:09 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
> 
> All,
> 
> Here is some updated charter text for CCUS.  Comments are most welcome.
> 
> Thanks,
> Alan
> 
> - - - - -
> 
> The Call Control UUI for SIP  (CCUS) working group is chartered to 
> define a SIP mechanism for transporting call control related 
> user-to-user (UUI) information between User Agents.
> 
> The mechanism developed in this working group is generally appropriate 
> in the following situations:
> 1. The information is generated and consumed by an application using SIP
> 
> during session setup but the application is 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. Multi-vendor interoperability is important.
> 
> This mechanism is not appropriate in the following situations:
> 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.
> 
> User data of the mechanism will be clearly marked with the application, 
> encoding, semantics, and contents of the data, allowing policy to be 
> applied by UAs.  The working group will define the information which 
> each application must specify in a standards-track document to utilize 
> the mechanism.
> 
> One important application of this mechanism is interworking with the 
> ISDN User to User Information Service.  This application defined by 
> ITU-T Q.931, is extensively deployed in the PSTN today, supporting such 
> applications as contact centers, call centers, and automatic call 
> distributors (ACDs).  A major barrier to the movement of these 
> applications to SIP is the lack of a standard mechanism to transport 
> this information in SIP.   For interworking with ISDN, minimal 
> information about the content of the UUI is available.  In this case 
> only, the content will just indicate ISDN UUI Service 1 interworking 
> rather than the actual content.
> 
> Call control UUI is user information conveyed between user agents during
> 
> call control operations.  As a result, the information must be conveyed 
> with the INVITE transaction, and must survive proxy retargeting, 
> redirection, and transfers.  The mechanism must utilize a minimum of SIP
> 
> extensions as it will need to be supported by many simple SIP user 
> agents such as PSTN gateways.  The mechanism must interwork with the 
> existing ISDN service but should also be extensible for use by other 
> applications and non-ISDN protocols.
> 
> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control UUI
> 
> can be generated by a number of protocols that are not ISUP, and as such
> 
> it is architecturally unreasonable to interwork these protocols at the 
> SIP / ISUP gateway. That is, existing SIP-T approaches defined in 
> RFC3372 do not meet the requirements.
> 
> Mechanisms based on the SIP INFO method, RFC2976, will not be considered
> 
> by the working group since the UUI contents carry information that must 
> be conveyed during session setup at the user agent - the information 
> must be conveyed with the INVITE transaction.  The information must be 
> passed with the session setup request (INVITE), responses to that 
> INVITE, or session termination requests.  As a result, it is not 
> possible to use INFO in these cases.
> 
> The group will produce:
> 
> - A problem statement and requirements document for implementing a SIP 
> call control UUI mechanism
> 
> - A specification of the SIP extension to best meet those requirements.
> 
> Goals and Milestones
> ====================
> 
> Mar 10 - Problem statement and requirements document to IESG 
> (Informational)
> Aug 10 - SIP call control UUI specification to IESG (PS)
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From Leon.Portman@nice.com  Wed Nov  4 06:08:11 2009
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From: Leon Portman <Leon.Portman@nice.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 4 Nov 2009 16:08:28 +0200
Thread-Topic: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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I will try to explain my case.

The use case for example is a customer is calling to an organization via SI=
P trunk with some UUI info from the hosted IVR session (business data). He =
is connected to an agent and then was put on hold, triggering REINVITE with=
 updated SDP. And then transferred to another agent. Under assumption that =
UUI represents some kind business data state of the call (IVR selection, cu=
stomer ID etc), this information has to be preserved during all session cha=
nges. It is not transient state that is delivered via INFO.
Or even more complex, the call is transferred via SIP trunk to completely a=
nother PBX as another SIP session (B2BUA scenario).

Thanks

Leon

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Wednesday, November 04, 2009 3:57 PM
To: Leon Portman
Cc: Sanjay Sinha (sanjsinh); Alan Johnston; dispatch@ietf.org
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SI=
P



Leon Portman wrote:
> Actually I was thinking that if UUI was presented during the original INV=
ITE, if it is going to be required to be presented during RE_INVITES as wel=
l, especially if it represents some client state information

Can you explain more about this?

The charter seems to spend its time explaining that this is governed by=20
the needs for session setup. If this is a mechanism that is also=20
intended for use after the session is established, then I think more=20
explanation is required.

As described, this is just plain tunneling, pure and simple.
And INFO (expecially with the new Info-Package work) is yet another=20
mechanism for tunneling. The rationale for not using INFO for UUI is=20
that it is needed during session establishment. Clearly that doesn't=20
apply after the session is established.

Of course its a mess to have two mechanisms with overlapping purposes.=20
It would be silly to use two mechanisms to transport the same=20
information, depending on when in the lifetime of the session the info=20
was being transported. But I'm having difficulty drawing a clear=20
distinction between the applicability of the two mechanisms.

	Thanks,
	Paul

> Leon
>=20
> -----Original Message-----
> From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com]=20
> Sent: Wednesday, November 04, 2009 11:18 AM
> To: Leon Portman; Alan Johnston; dispatch@ietf.org
> Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI for =
SIP
>=20
> I think UUI is carried during session setup and teardown, and hence
> re-Invite scenario may not be applicable here.
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Leon Portman
> Sent: Wednesday, November 04, 2009 1:29 PM
> To: Alan Johnston; dispatch@ietf.org
> Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for
> SIP
>=20
> Only one comment, does it also supposed to support re-invite (for SRTP
> re-keying or media updates) scenarios as well?
>=20
> Thanks
>=20
> Leon
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Alan Johnston
> Sent: Tuesday, November 03, 2009 11:09 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
>=20
> All,
>=20
> Here is some updated charter text for CCUS.  Comments are most welcome.
>=20
> Thanks,
> Alan
>=20
> - - - - -
>=20
> The Call Control UUI for SIP  (CCUS) working group is chartered to=20
> define a SIP mechanism for transporting call control related=20
> user-to-user (UUI) information between User Agents.
>=20
> The mechanism developed in this working group is generally appropriate=20
> in the following situations:
> 1. The information is generated and consumed by an application using SIP
>=20
> during session setup but the application is 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=20
> retargeting.
> 5. SIP elements may need to apply policy about passing and screening the
>=20
> information.
> 6. Multi-vendor interoperability is important.
>=20
> This mechanism is not appropriate in the following situations:
> 1. SIP behavior is changed.
> 2. The information is generated and consumed at the SIP layer by SIP=20
> elements.
> 3. SIP elements besides the UAC and UAS might be interested and consume=20
> the information.
> 4. There are specific privacy or cross-RAI issues involved with the=20
> information being transported.
>=20
> User data of the mechanism will be clearly marked with the application,=20
> encoding, semantics, and contents of the data, allowing policy to be=20
> applied by UAs.  The working group will define the information which=20
> each application must specify in a standards-track document to utilize=20
> the mechanism.
>=20
> One important application of this mechanism is interworking with the=20
> ISDN User to User Information Service.  This application defined by=20
> ITU-T Q.931, is extensively deployed in the PSTN today, supporting such=20
> applications as contact centers, call centers, and automatic call=20
> distributors (ACDs).  A major barrier to the movement of these=20
> applications to SIP is the lack of a standard mechanism to transport=20
> this information in SIP.   For interworking with ISDN, minimal=20
> information about the content of the UUI is available.  In this case=20
> only, the content will just indicate ISDN UUI Service 1 interworking=20
> rather than the actual content.
>=20
> Call control UUI is user information conveyed between user agents during
>=20
> call control operations.  As a result, the information must be conveyed=20
> with the INVITE transaction, and must survive proxy retargeting,=20
> redirection, and transfers.  The mechanism must utilize a minimum of SIP
>=20
> extensions as it will need to be supported by many simple SIP user=20
> agents such as PSTN gateways.  The mechanism must interwork with the=20
> existing ISDN service but should also be extensible for use by other=20
> applications and non-ISDN protocols.
>=20
> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control UUI
>=20
> can be generated by a number of protocols that are not ISUP, and as such
>=20
> it is architecturally unreasonable to interwork these protocols at the=20
> SIP / ISUP gateway. That is, existing SIP-T approaches defined in=20
> RFC3372 do not meet the requirements.
>=20
> Mechanisms based on the SIP INFO method, RFC2976, will not be considered
>=20
> by the working group since the UUI contents carry information that must=20
> be conveyed during session setup at the user agent - the information=20
> must be conveyed with the INVITE transaction.  The information must be=20
> passed with the session setup request (INVITE), responses to that=20
> INVITE, or session termination requests.  As a result, it is not=20
> possible to use INFO in these cases.
>=20
> The group will produce:
>=20
> - A problem statement and requirements document for implementing a SIP=20
> call control UUI mechanism
>=20
> - A specification of the SIP extension to best meet those requirements.
>=20
> Goals and Milestones
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> Mar 10 - Problem statement and requirements document to IESG=20
> (Informational)
> Aug 10 - SIP call control UUI specification to IESG (PS)
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From alan@sipstation.com  Wed Nov  4 07:47:01 2009
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Cc: Leon Portman <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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This is an interesting question.  On first blush, it seems that due to  
3pcc type applications, the mechanism will need to be used in re- 
INVITEs. However, we will need to be very careful to define the  
semantics for true mid-call uses.

I don't think the charter needs to specify this much detail - the  
requirements document produced by the working group should include this.

Any comments on the charter text? I know I am looking forward to  
getting back to the mechanism discussion...

Thanks
Alan



On Nov 4, 2009, at 8:01 AM, Paul Kyzivat <pkyzivat@cisco.com> wrote:

> I always use 3pcc as a test case, because it presents problems  
> almost every time.
>
> Of course the problem with 3pcc is that one "side" may remain the  
> same, while the other "side" is replaced. The replacement typically  
> has no knowledge of the history of what went on before it joined. So  
> there usually needs to be some mechanism for it to be clued in about  
> that.
>
> I don't know if the same is true here or not. It depends on whether  
> the UUI information is viewed as transient, or as state of the  
> session.
>
>    Thanks,
>    Paul
>
> Sanjay Sinha (sanjsinh) wrote:
>> Do you mean same IE that was present in initial INVITE needs to be
>> presented in Re-Invite as well? If the information does not change,  
>> then I do not see any value in
>> carrying the same payload in re-Invite. But if there is a new IE,  
>> then
>> yeah it can be presented in re-Invite.
>> I am not sure if there can be a new UU IE after the call is setup and
>> before call termination.
>> Sanjay
>> -----Original Message-----
>> From: Leon Portman [mailto:Leon.Portman@nice.com] Sent: Wednesday,  
>> November 04, 2009 3:02 PM
>> To: Sanjay Sinha (sanjsinh); Alan Johnston; dispatch@ietf.org
>> Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control  
>> UUI for
>> SIP
>> Actually I was thinking that if UUI was presented during the original
>> INVITE, if it is going to be required to be presented during  
>> RE_INVITES
>> as well, especially if it represents some client state information
>> Leon
>> -----Original Message-----
>> From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com] Sent:  
>> Wednesday, November 04, 2009 11:18 AM
>> To: Leon Portman; Alan Johnston; dispatch@ietf.org
>> Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control  
>> UUI for
>> SIP
>> I think UUI is carried during session setup and teardown, and hence
>> re-Invite scenario may not be applicable here.
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Leon Portman
>> Sent: Wednesday, November 04, 2009 1:29 PM
>> To: Alan Johnston; dispatch@ietf.org
>> Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control  
>> UUI for
>> SIP
>> Only one comment, does it also supposed to support re-invite (for  
>> SRTP
>> re-keying or media updates) scenarios as well?
>> Thanks
>> Leon
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Alan Johnston
>> Sent: Tuesday, November 03, 2009 11:09 PM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI  
>> for SIP
>> All,
>> Here is some updated charter text for CCUS.  Comments are most  
>> welcome.
>> Thanks,
>> Alan
>> - - - - -
>> The Call Control UUI for SIP  (CCUS) working group is chartered to  
>> define a SIP mechanism for transporting call control related user- 
>> to-user (UUI) information between User Agents.
>> The mechanism developed in this working group is generally  
>> appropriate in the following situations:
>> 1. The information is generated and consumed by an application  
>> using SIP
>> during session setup but the application is 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. Multi-vendor interoperability is important.
>> This mechanism is not appropriate in the following situations:
>> 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.
>> User data of the mechanism will be clearly marked with the  
>> application, encoding, semantics, and contents of the data,  
>> allowing policy to be applied by UAs.  The working group will  
>> define the information which each application must specify in a  
>> standards-track document to utilize the mechanism.
>> One important application of this mechanism is interworking with  
>> the ISDN User to User Information Service.  This application  
>> defined by ITU-T Q.931, is extensively deployed in the PSTN today,  
>> supporting such applications as contact centers, call centers, and  
>> automatic call distributors (ACDs).  A major barrier to the  
>> movement of these applications to SIP is the lack of a standard  
>> mechanism to transport this information in SIP.   For interworking  
>> with ISDN, minimal information about the content of the UUI is  
>> available.  In this case only, the content will just indicate ISDN  
>> UUI Service 1 interworking rather than the actual content.
>> Call control UUI is user information conveyed between user agents  
>> during
>> call control operations.  As a result, the information must be  
>> conveyed with the INVITE transaction, and must survive proxy  
>> retargeting, redirection, and transfers.  The mechanism must  
>> utilize a minimum of SIP
>> extensions as it will need to be supported by many simple SIP user  
>> agents such as PSTN gateways.  The mechanism must interwork with  
>> the existing ISDN service but should also be extensible for use by  
>> other applications and non-ISDN protocols.
>> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call  
>> control UUI
>> can be generated by a number of protocols that are not ISUP, and as  
>> such
>> it is architecturally unreasonable to interwork these protocols at  
>> the SIP / ISUP gateway. That is, existing SIP-T approaches defined  
>> in RFC3372 do not meet the requirements.
>> Mechanisms based on the SIP INFO method, RFC2976, will not be  
>> considered
>> by the working group since the UUI contents carry information that  
>> must be conveyed during session setup at the user agent - the  
>> information must be conveyed with the INVITE transaction.  The  
>> information must be passed with the session setup request (INVITE),  
>> responses to that INVITE, or session termination requests.  As a  
>> result, it is not possible to use INFO in these cases.
>> The group will produce:
>> - A problem statement and requirements document for implementing a  
>> SIP call control UUI mechanism
>> - A specification of the SIP extension to best meet those  
>> requirements.
>> Goals and Milestones
>> ====================
>> Mar 10 - Problem statement and requirements document to IESG  
>> (Informational)
>> Aug 10 - SIP call control UUI specification to IESG (PS)
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch

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Thread-Topic: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: "Paul Kyzivat (pkyzivat)" <pkyzivat@cisco.com>
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Cc: Leon Portman <Leon.Portman@nice.com>, dispatch@ietf.org
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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Yeah this is a good case and for it we need re-Invite to carry it.

If it is session related data, then for transfers should it be carried
in REFER also, esp. for REFER with Replaces?

Sanjay

-----Original Message-----
From: Paul Kyzivat (pkyzivat)=20
Sent: Wednesday, November 04, 2009 7:31 PM
To: Sanjay Sinha (sanjsinh)
Cc: Leon Portman; Alan Johnston; dispatch@ietf.org
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for
SIP

I always use 3pcc as a test case, because it presents problems almost=20
every time.

Of course the problem with 3pcc is that one "side" may remain the same,=20
while the other "side" is replaced. The replacement typically has no=20
knowledge of the history of what went on before it joined. So there=20
usually needs to be some mechanism for it to be clued in about that.

I don't know if the same is true here or not. It depends on whether the=20
UUI information is viewed as transient, or as state of the session.

	Thanks,
	Paul

Sanjay Sinha (sanjsinh) wrote:
> Do you mean same IE that was present in initial INVITE needs to be
> presented in Re-Invite as well?=20
>=20
> If the information does not change, then I do not see any value in
> carrying the same payload in re-Invite. But if there is a new IE, then
> yeah it can be presented in re-Invite.
> I am not sure if there can be a new UU IE after the call is setup and
> before call termination.
>=20
> Sanjay
>=20
> -----Original Message-----
> From: Leon Portman [mailto:Leon.Portman@nice.com]=20
> Sent: Wednesday, November 04, 2009 3:02 PM
> To: Sanjay Sinha (sanjsinh); Alan Johnston; dispatch@ietf.org
> Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI
for
> SIP
>=20
> Actually I was thinking that if UUI was presented during the original
> INVITE, if it is going to be required to be presented during
RE_INVITES
> as well, especially if it represents some client state information
>=20
> Leon
>=20
> -----Original Message-----
> From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com]=20
> Sent: Wednesday, November 04, 2009 11:18 AM
> To: Leon Portman; Alan Johnston; dispatch@ietf.org
> Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI
for
> SIP
>=20
> I think UUI is carried during session setup and teardown, and hence
> re-Invite scenario may not be applicable here.
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Leon Portman
> Sent: Wednesday, November 04, 2009 1:29 PM
> To: Alan Johnston; dispatch@ietf.org
> Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI
for
> SIP
>=20
> Only one comment, does it also supposed to support re-invite (for SRTP
> re-keying or media updates) scenarios as well?
>=20
> Thanks
>=20
> Leon
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Alan Johnston
> Sent: Tuesday, November 03, 2009 11:09 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI for
SIP
>=20
> All,
>=20
> Here is some updated charter text for CCUS.  Comments are most
welcome.
>=20
> Thanks,
> Alan
>=20
> - - - - -
>=20
> The Call Control UUI for SIP  (CCUS) working group is chartered to=20
> define a SIP mechanism for transporting call control related=20
> user-to-user (UUI) information between User Agents.
>=20
> The mechanism developed in this working group is generally appropriate

> in the following situations:
> 1. The information is generated and consumed by an application using
SIP
>=20
> during session setup but the application is 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=20
> retargeting.
> 5. SIP elements may need to apply policy about passing and screening
the
>=20
> information.
> 6. Multi-vendor interoperability is important.
>=20
> This mechanism is not appropriate in the following situations:
> 1. SIP behavior is changed.
> 2. The information is generated and consumed at the SIP layer by SIP=20
> elements.
> 3. SIP elements besides the UAC and UAS might be interested and
consume=20
> the information.
> 4. There are specific privacy or cross-RAI issues involved with the=20
> information being transported.
>=20
> User data of the mechanism will be clearly marked with the
application,=20
> encoding, semantics, and contents of the data, allowing policy to be=20
> applied by UAs.  The working group will define the information which=20
> each application must specify in a standards-track document to utilize

> the mechanism.
>=20
> One important application of this mechanism is interworking with the=20
> ISDN User to User Information Service.  This application defined by=20
> ITU-T Q.931, is extensively deployed in the PSTN today, supporting
such=20
> applications as contact centers, call centers, and automatic call=20
> distributors (ACDs).  A major barrier to the movement of these=20
> applications to SIP is the lack of a standard mechanism to transport=20
> this information in SIP.   For interworking with ISDN, minimal=20
> information about the content of the UUI is available.  In this case=20
> only, the content will just indicate ISDN UUI Service 1 interworking=20
> rather than the actual content.
>=20
> Call control UUI is user information conveyed between user agents
during
>=20
> call control operations.  As a result, the information must be
conveyed=20
> with the INVITE transaction, and must survive proxy retargeting,=20
> redirection, and transfers.  The mechanism must utilize a minimum of
SIP
>=20
> extensions as it will need to be supported by many simple SIP user=20
> agents such as PSTN gateways.  The mechanism must interwork with the=20
> existing ISDN service but should also be extensible for use by other=20
> applications and non-ISDN protocols.
>=20
> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control
UUI
>=20
> can be generated by a number of protocols that are not ISUP, and as
such
>=20
> it is architecturally unreasonable to interwork these protocols at the

> SIP / ISUP gateway. That is, existing SIP-T approaches defined in=20
> RFC3372 do not meet the requirements.
>=20
> Mechanisms based on the SIP INFO method, RFC2976, will not be
considered
>=20
> by the working group since the UUI contents carry information that
must=20
> be conveyed during session setup at the user agent - the information=20
> must be conveyed with the INVITE transaction.  The information must be

> passed with the session setup request (INVITE), responses to that=20
> INVITE, or session termination requests.  As a result, it is not=20
> possible to use INFO in these cases.
>=20
> The group will produce:
>=20
> - A problem statement and requirements document for implementing a SIP

> call control UUI mechanism
>=20
> - A specification of the SIP extension to best meet those
requirements.
>=20
> Goals and Milestones
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> Mar 10 - Problem statement and requirements document to IESG=20
> (Informational)
> Aug 10 - SIP call control UUI specification to IESG (PS)
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From pkyzivat@cisco.com  Wed Nov  4 16:10:36 2009
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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Leon Portman wrote:
> I will try to explain my case.
> 
> The use case for example is a customer is calling to an organization via SIP trunk with some UUI info from the hosted IVR session (business data). He is connected to an agent and then was put on hold, triggering REINVITE with updated SDP. And then transferred to another agent. Under assumption that UUI represents some kind business data state of the call (IVR selection, customer ID etc), this information has to be preserved during all session changes. It is not transient state that is delivered via INFO.
> Or even more complex, the call is transferred via SIP trunk to completely another PBX as another SIP session (B2BUA scenario).

So you are making the case that this is *state* that must remain 
synchronized over time, not just transient information for call 
establishment.

But that supports my argument that this is generalized 
application-application tunneling over sip.
As such it overlaps greatly with the charter for INFO.
We then end up with two mechanisms, one using headers and the other 
using bodies.
- INFO can build on the rich structure of MIME definitions
   but it can't be used until at least an early dialog is established

- The proposed UUI mechanism has to be embedded in headers,
   and has a much more limited representation mechanism built up.
   (But could you embed arbitrarily complex MIME bodies into a
   UUI header value?) It can be used earlier, and in contexts
   where INFO can't.

As best I can tell, in practice, the UUI stuff will only be used where 
the data formats are already established for use in legacy PSTN 
equipment. As that stuff gets cut over to SIP infrastructure we will be 
left with applications having to use encodings that are no longer at all 
natural. (E.g. need to find something that can encode/decode ASN.1 stuff 
in an application that uses it for nothing else.)

And this is all in spite of long resistance to using SIP for tunneling 
in general.

	Just grumbling,
	Paul

> Thanks
> 
> Leon
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Wednesday, November 04, 2009 3:57 PM
> To: Leon Portman
> Cc: Sanjay Sinha (sanjsinh); Alan Johnston; dispatch@ietf.org
> Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
> 
> 
> 
> Leon Portman wrote:
>> Actually I was thinking that if UUI was presented during the original INVITE, if it is going to be required to be presented during RE_INVITES as well, especially if it represents some client state information
> 
> Can you explain more about this?
> 
> The charter seems to spend its time explaining that this is governed by 
> the needs for session setup. If this is a mechanism that is also 
> intended for use after the session is established, then I think more 
> explanation is required.
> 
> As described, this is just plain tunneling, pure and simple.
> And INFO (expecially with the new Info-Package work) is yet another 
> mechanism for tunneling. The rationale for not using INFO for UUI is 
> that it is needed during session establishment. Clearly that doesn't 
> apply after the session is established.
> 
> Of course its a mess to have two mechanisms with overlapping purposes. 
> It would be silly to use two mechanisms to transport the same 
> information, depending on when in the lifetime of the session the info 
> was being transported. But I'm having difficulty drawing a clear 
> distinction between the applicability of the two mechanisms.
> 
> 	Thanks,
> 	Paul
> 
>> Leon
>>
>> -----Original Message-----
>> From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com] 
>> Sent: Wednesday, November 04, 2009 11:18 AM
>> To: Leon Portman; Alan Johnston; dispatch@ietf.org
>> Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
>>
>> I think UUI is carried during session setup and teardown, and hence
>> re-Invite scenario may not be applicable here.
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Leon Portman
>> Sent: Wednesday, November 04, 2009 1:29 PM
>> To: Alan Johnston; dispatch@ietf.org
>> Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for
>> SIP
>>
>> Only one comment, does it also supposed to support re-invite (for SRTP
>> re-keying or media updates) scenarios as well?
>>
>> Thanks
>>
>> Leon
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Alan Johnston
>> Sent: Tuesday, November 03, 2009 11:09 PM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
>>
>> All,
>>
>> Here is some updated charter text for CCUS.  Comments are most welcome.
>>
>> Thanks,
>> Alan
>>
>> - - - - -
>>
>> The Call Control UUI for SIP  (CCUS) working group is chartered to 
>> define a SIP mechanism for transporting call control related 
>> user-to-user (UUI) information between User Agents.
>>
>> The mechanism developed in this working group is generally appropriate 
>> in the following situations:
>> 1. The information is generated and consumed by an application using SIP
>>
>> during session setup but the application is 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. Multi-vendor interoperability is important.
>>
>> This mechanism is not appropriate in the following situations:
>> 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.
>>
>> User data of the mechanism will be clearly marked with the application, 
>> encoding, semantics, and contents of the data, allowing policy to be 
>> applied by UAs.  The working group will define the information which 
>> each application must specify in a standards-track document to utilize 
>> the mechanism.
>>
>> One important application of this mechanism is interworking with the 
>> ISDN User to User Information Service.  This application defined by 
>> ITU-T Q.931, is extensively deployed in the PSTN today, supporting such 
>> applications as contact centers, call centers, and automatic call 
>> distributors (ACDs).  A major barrier to the movement of these 
>> applications to SIP is the lack of a standard mechanism to transport 
>> this information in SIP.   For interworking with ISDN, minimal 
>> information about the content of the UUI is available.  In this case 
>> only, the content will just indicate ISDN UUI Service 1 interworking 
>> rather than the actual content.
>>
>> Call control UUI is user information conveyed between user agents during
>>
>> call control operations.  As a result, the information must be conveyed 
>> with the INVITE transaction, and must survive proxy retargeting, 
>> redirection, and transfers.  The mechanism must utilize a minimum of SIP
>>
>> extensions as it will need to be supported by many simple SIP user 
>> agents such as PSTN gateways.  The mechanism must interwork with the 
>> existing ISDN service but should also be extensible for use by other 
>> applications and non-ISDN protocols.
>>
>> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control UUI
>>
>> can be generated by a number of protocols that are not ISUP, and as such
>>
>> it is architecturally unreasonable to interwork these protocols at the 
>> SIP / ISUP gateway. That is, existing SIP-T approaches defined in 
>> RFC3372 do not meet the requirements.
>>
>> Mechanisms based on the SIP INFO method, RFC2976, will not be considered
>>
>> by the working group since the UUI contents carry information that must 
>> be conveyed during session setup at the user agent - the information 
>> must be conveyed with the INVITE transaction.  The information must be 
>> passed with the session setup request (INVITE), responses to that 
>> INVITE, or session termination requests.  As a result, it is not 
>> possible to use INFO in these cases.
>>
>> The group will produce:
>>
>> - A problem statement and requirements document for implementing a SIP 
>> call control UUI mechanism
>>
>> - A specification of the SIP extension to best meet those requirements.
>>
>> Goals and Milestones
>> ====================
>>
>> Mar 10 - Problem statement and requirements document to IESG 
>> (Informational)
>> Aug 10 - SIP call control UUI specification to IESG (PS)
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> 

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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Alan Johnston <alan@sipstation.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 5 Nov 2009 14:16:09 +0100
Thread-Topic: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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1)	"4. The information is expected to survive retargeting, redirection, and=
 retargeting."=20

"retargeting" appears twice.

2)	"For interworking with ISDN, minimal=20
information about the content of the UUI is available.  In this case only, =
the content will just indicate ISDN UUI Service 1 interworking rather than =
the actual content."

This could imply that the existing protocol discriminator, which does carry=
 information about the content, is lost. I hope that is not true.

3)	I am somewhat concerned that the focus now appears to be on SIP UA to SI=
P UA, rather than interworking with other signalling systems that also carr=
y such information. Where we have a SIP UA talking to a SIP UA, For at leas=
t some of the usages where a SIP UA is talking to a SIP UA, we may well alr=
eady have SIP mechanisms in place to support this. In general, we should be=
 focussing on those existing usages transported in other protocols, and any=
 new pure SIP usage should go through the normal SIP extension consideratio=
ns. With the focus as it currently appears, we could end up spending years =
defining a fully proper solution, whereas I believe we need something quick=
er and potentially less than perfect for the SIP UA to SIP UA scenarios. No=
te that by SIP UA I am talking about a real SIP endpoint here, not the UA t=
hat may form part of an interworking gateway.

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Alan Johnston
> Sent: Tuesday, November 03, 2009 9:09 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Updated CCUS Charter Text - Call Control=20
> UUI for SIP
>=20
> All,
>=20
> Here is some updated charter text for CCUS.  Comments are=20
> most welcome.
>=20
> Thanks,
> Alan
>=20
> - - - - -
>=20
> The Call Control UUI for SIP  (CCUS) working group is=20
> chartered to define a SIP mechanism for transporting call=20
> control related user-to-user (UUI) information between User Agents.
>=20
> The mechanism developed in this working group is generally=20
> appropriate in the following situations:
> 1. The information is generated and consumed by an=20
> application using SIP during session setup but the=20
> application is 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,=20
> redirection, and retargeting.
> 5. SIP elements may need to apply policy about passing and=20
> screening the information.
> 6. Multi-vendor interoperability is important.
>=20
> This mechanism is not appropriate in the following situations:
> 1. SIP behavior is changed.
> 2. The information is generated and consumed at the SIP layer=20
> by SIP elements.
> 3. SIP elements besides the UAC and UAS might be interested=20
> and consume the information.
> 4. There are specific privacy or cross-RAI issues involved=20
> with the information being transported.
>=20
> User data of the mechanism will be clearly marked with the=20
> application, encoding, semantics, and contents of the data,=20
> allowing policy to be applied by UAs.  The working group will=20
> define the information which each application must specify in=20
> a standards-track document to utilize the mechanism.
>=20
> One important application of this mechanism is interworking=20
> with the ISDN User to User Information Service.  This=20
> application defined by ITU-T Q.931, is extensively deployed=20
> in the PSTN today, supporting such applications as contact=20
> centers, call centers, and automatic call distributors=20
> (ACDs).  A major barrier to the movement of these=20
> applications to SIP is the lack of a standard mechanism to transport=20
> this information in SIP.   For interworking with ISDN, minimal=20
> information about the content of the UUI is available.  In=20
> this case only, the content will just indicate ISDN UUI=20
> Service 1 interworking rather than the actual content.
>=20
> Call control UUI is user information conveyed between user=20
> agents during call control operations.  As a result, the=20
> information must be conveyed with the INVITE transaction, and=20
> must survive proxy retargeting, redirection, and transfers. =20
> The mechanism must utilize a minimum of SIP extensions as it=20
> will need to be supported by many simple SIP user agents such=20
> as PSTN gateways.  The mechanism must interwork with the=20
> existing ISDN service but should also be extensible for use=20
> by other applications and non-ISDN protocols.
>=20
> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call=20
> control UUI can be generated by a number of protocols that=20
> are not ISUP, and as such it is architecturally unreasonable=20
> to interwork these protocols at the SIP / ISUP gateway. That=20
> is, existing SIP-T approaches defined in
> RFC3372 do not meet the requirements.
>=20
> Mechanisms based on the SIP INFO method, RFC2976, will not be=20
> considered by the working group since the UUI contents carry=20
> information that must be conveyed during session setup at the=20
> user agent - the information must be conveyed with the INVITE=20
> transaction.  The information must be passed with the session=20
> setup request (INVITE), responses to that INVITE, or session=20
> termination requests.  As a result, it is not possible to use=20
> INFO in these cases.
>=20
> The group will produce:
>=20
> - A problem statement and requirements document for=20
> implementing a SIP call control UUI mechanism
>=20
> - A specification of the SIP extension to best meet those=20
> requirements.
>=20
> Goals and Milestones
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> Mar 10 - Problem statement and requirements document to IESG
> (Informational)
> Aug 10 - SIP call control UUI specification to IESG (PS)
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From drage@alcatel-lucent.com  Thu Nov  5 05:17:18 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
Date: Thu, 5 Nov 2009 14:17:29 +0100
Thread-Topic: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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References: <4AF09BF6.9010902@sipstation.com> <07465C1D981ABC41A344374066AE1A2C37D6AF5828@TLVMBX01.nice.com> <00FC4AA684E90E4DA2FF71021CD5A6CA0C005E@XMB-RCD-101.cisco.com> <07465C1D981ABC41A344374066AE1A2C37D6AF594B@TLVMBX01.nice.com> <00FC4AA684E90E4DA2FF71021CD5A6CA0C00A1@XMB-RCD-101.cisco.com> <4AF1892C.4060504@cisco.com>
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Cc: Leon Portman <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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I would argue that the 3pcc case is covered by the statement that the infor=
mation must survive transfer.

Talking more generally about the 3pcc case in the charter means that we cou=
ld end up discussing scenarios of information transfer during the call.=20

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Wednesday, November 04, 2009 2:01 PM
> To: Sanjay Sinha (sanjsinh)
> Cc: Leon Portman; dispatch@ietf.org
> Subject: Re: [dispatch] Updated CCUS Charter Text - Call=20
> Control UUI for SIP
>=20
> I always use 3pcc as a test case, because it presents=20
> problems almost every time.
>=20
> Of course the problem with 3pcc is that one "side" may remain=20
> the same, while the other "side" is replaced. The replacement=20
> typically has no knowledge of the history of what went on=20
> before it joined. So there usually needs to be some mechanism=20
> for it to be clued in about that.
>=20
> I don't know if the same is true here or not. It depends on=20
> whether the UUI information is viewed as transient, or as=20
> state of the session.
>=20
> 	Thanks,
> 	Paul
>=20
> Sanjay Sinha (sanjsinh) wrote:
> > Do you mean same IE that was present in initial INVITE needs to be=20
> > presented in Re-Invite as well?
> >=20
> > If the information does not change, then I do not see any value in=20
> > carrying the same payload in re-Invite. But if there is a=20
> new IE, then=20
> > yeah it can be presented in re-Invite.
> > I am not sure if there can be a new UU IE after the call is=20
> setup and=20
> > before call termination.
> >=20
> > Sanjay
> >=20
> > -----Original Message-----
> > From: Leon Portman [mailto:Leon.Portman@nice.com]
> > Sent: Wednesday, November 04, 2009 3:02 PM
> > To: Sanjay Sinha (sanjsinh); Alan Johnston; dispatch@ietf.org
> > Subject: RE: [dispatch] Updated CCUS Charter Text - Call=20
> Control UUI=20
> > for SIP
> >=20
> > Actually I was thinking that if UUI was presented during=20
> the original=20
> > INVITE, if it is going to be required to be presented during=20
> > RE_INVITES as well, especially if it represents some client state=20
> > information
> >=20
> > Leon
> >=20
> > -----Original Message-----
> > From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com]
> > Sent: Wednesday, November 04, 2009 11:18 AM
> > To: Leon Portman; Alan Johnston; dispatch@ietf.org
> > Subject: RE: [dispatch] Updated CCUS Charter Text - Call=20
> Control UUI=20
> > for SIP
> >=20
> > I think UUI is carried during session setup and teardown, and hence=20
> > re-Invite scenario may not be applicable here.
> >=20
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On=20
> > Behalf Of Leon Portman
> > Sent: Wednesday, November 04, 2009 1:29 PM
> > To: Alan Johnston; dispatch@ietf.org
> > Subject: Re: [dispatch] Updated CCUS Charter Text - Call=20
> Control UUI=20
> > for SIP
> >=20
> > Only one comment, does it also supposed to support=20
> re-invite (for SRTP=20
> > re-keying or media updates) scenarios as well?
> >=20
> > Thanks
> >=20
> > Leon
> >=20
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On=20
> > Behalf Of Alan Johnston
> > Sent: Tuesday, November 03, 2009 11:09 PM
> > To: dispatch@ietf.org
> > Subject: [dispatch] Updated CCUS Charter Text - Call=20
> Control UUI for=20
> > SIP
> >=20
> > All,
> >=20
> > Here is some updated charter text for CCUS.  Comments are=20
> most welcome.
> >=20
> > Thanks,
> > Alan
> >=20
> > - - - - -
> >=20
> > The Call Control UUI for SIP  (CCUS) working group is chartered to=20
> > define a SIP mechanism for transporting call control related=20
> > user-to-user (UUI) information between User Agents.
> >=20
> > The mechanism developed in this working group is generally=20
> appropriate=20
> > in the following situations:
> > 1. The information is generated and consumed by an=20
> application using=20
> > SIP
> >=20
> > during session setup but the application is not necessarily=20
> even SIP=20
> > 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,=20
> > and retargeting.
> > 5. SIP elements may need to apply policy about passing and=20
> screening=20
> > the
> >=20
> > information.
> > 6. Multi-vendor interoperability is important.
> >=20
> > This mechanism is not appropriate in the following situations:
> > 1. SIP behavior is changed.
> > 2. The information is generated and consumed at the SIP=20
> layer by SIP=20
> > elements.
> > 3. SIP elements besides the UAC and UAS might be interested and=20
> > consume the information.
> > 4. There are specific privacy or cross-RAI issues involved with the=20
> > information being transported.
> >=20
> > User data of the mechanism will be clearly marked with the=20
> > application, encoding, semantics, and contents of the data,=20
> allowing=20
> > policy to be applied by UAs.  The working group will define the=20
> > information which each application must specify in a=20
> standards-track=20
> > document to utilize the mechanism.
> >=20
> > One important application of this mechanism is interworking=20
> with the=20
> > ISDN User to User Information Service.  This application defined by=20
> > ITU-T Q.931, is extensively deployed in the PSTN today, supporting=20
> > such applications as contact centers, call centers, and=20
> automatic call=20
> > distributors (ACDs).  A major barrier to the movement of these=20
> > applications to SIP is the lack of a standard mechanism to transport
> > this information in SIP.   For interworking with ISDN, minimal=20
> > information about the content of the UUI is available.  In=20
> this case=20
> > only, the content will just indicate ISDN UUI Service 1=20
> interworking=20
> > rather than the actual content.
> >=20
> > Call control UUI is user information conveyed between user agents=20
> > during
> >=20
> > call control operations.  As a result, the information must be=20
> > conveyed with the INVITE transaction, and must survive proxy=20
> > retargeting, redirection, and transfers.  The mechanism=20
> must utilize a=20
> > minimum of SIP
> >=20
> > extensions as it will need to be supported by many simple SIP user=20
> > agents such as PSTN gateways.  The mechanism must interwork=20
> with the=20
> > existing ISDN service but should also be extensible for use=20
> by other=20
> > applications and non-ISDN protocols.
> >=20
> > While PSTN UUI is carried in ISUP (ISDN User Part), SIP=20
> call control=20
> > UUI
> >=20
> > can be generated by a number of protocols that are not ISUP, and as=20
> > such
> >=20
> > it is architecturally unreasonable to interwork these=20
> protocols at the=20
> > SIP / ISUP gateway. That is, existing SIP-T approaches defined in
> > RFC3372 do not meet the requirements.
> >=20
> > Mechanisms based on the SIP INFO method, RFC2976, will not be=20
> > considered
> >=20
> > by the working group since the UUI contents carry information that=20
> > must be conveyed during session setup at the user agent - the=20
> > information must be conveyed with the INVITE transaction.  The=20
> > information must be passed with the session setup request (INVITE),=20
> > responses to that INVITE, or session termination requests.  As a=20
> > result, it is not possible to use INFO in these cases.
> >=20
> > The group will produce:
> >=20
> > - A problem statement and requirements document for=20
> implementing a SIP=20
> > call control UUI mechanism
> >=20
> > - A specification of the SIP extension to best meet those=20
> requirements.
> >=20
> > Goals and Milestones
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >=20
> > Mar 10 - Problem statement and requirements document to IESG
> > (Informational)
> > Aug 10 - SIP call control UUI specification to IESG (PS)
> >=20
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Leon Portman <Leon.Portman@nice.com>
Date: Thu, 5 Nov 2009 14:17:58 +0100
Thread-Topic: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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References: <4AF09BF6.9010902@sipstation.com> <07465C1D981ABC41A344374066AE1A2C37D6AF5828@TLVMBX01.nice.com> <00FC4AA684E90E4DA2FF71021CD5A6CA0C005E@XMB-RCD-101.cisco.com> <07465C1D981ABC41A344374066AE1A2C37D6AF594B@TLVMBX01.nice.com> <4AF18822.4070603@cisco.com>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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I concur with Paul here.

Keith=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Wednesday, November 04, 2009 1:57 PM
> To: Leon Portman
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Updated CCUS Charter Text - Call=20
> Control UUI for SIP
>=20
>=20
>=20
> Leon Portman wrote:
> > Actually I was thinking that if UUI was presented during=20
> the original=20
> > INVITE, if it is going to be required to be presented during=20
> > RE_INVITES as well, especially if it represents some client state=20
> > information
>=20
> Can you explain more about this?
>=20
> The charter seems to spend its time explaining that this is=20
> governed by the needs for session setup. If this is a=20
> mechanism that is also intended for use after the session is=20
> established, then I think more explanation is required.
>=20
> As described, this is just plain tunneling, pure and simple.
> And INFO (expecially with the new Info-Package work) is yet=20
> another mechanism for tunneling. The rationale for not using=20
> INFO for UUI is that it is needed during session=20
> establishment. Clearly that doesn't apply after the session=20
> is established.
>=20
> Of course its a mess to have two mechanisms with overlapping=20
> purposes.=20
> It would be silly to use two mechanisms to transport the same=20
> information, depending on when in the lifetime of the session=20
> the info was being transported. But I'm having difficulty=20
> drawing a clear distinction between the applicability of the=20
> two mechanisms.
>=20
> 	Thanks,
> 	Paul
>=20
> > Leon
> >=20
> > -----Original Message-----
> > From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com]
> > Sent: Wednesday, November 04, 2009 11:18 AM
> > To: Leon Portman; Alan Johnston; dispatch@ietf.org
> > Subject: RE: [dispatch] Updated CCUS Charter Text - Call=20
> Control UUI=20
> > for SIP
> >=20
> > I think UUI is carried during session setup and teardown, and hence=20
> > re-Invite scenario may not be applicable here.
> >=20
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On=20
> > Behalf Of Leon Portman
> > Sent: Wednesday, November 04, 2009 1:29 PM
> > To: Alan Johnston; dispatch@ietf.org
> > Subject: Re: [dispatch] Updated CCUS Charter Text - Call=20
> Control UUI=20
> > for SIP
> >=20
> > Only one comment, does it also supposed to support=20
> re-invite (for SRTP=20
> > re-keying or media updates) scenarios as well?
> >=20
> > Thanks
> >=20
> > Leon
> >=20
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On=20
> > Behalf Of Alan Johnston
> > Sent: Tuesday, November 03, 2009 11:09 PM
> > To: dispatch@ietf.org
> > Subject: [dispatch] Updated CCUS Charter Text - Call=20
> Control UUI for=20
> > SIP
> >=20
> > All,
> >=20
> > Here is some updated charter text for CCUS.  Comments are=20
> most welcome.
> >=20
> > Thanks,
> > Alan
> >=20
> > - - - - -
> >=20
> > The Call Control UUI for SIP  (CCUS) working group is chartered to=20
> > define a SIP mechanism for transporting call control related=20
> > user-to-user (UUI) information between User Agents.
> >=20
> > The mechanism developed in this working group is generally=20
> appropriate=20
> > in the following situations:
> > 1. The information is generated and consumed by an=20
> application using=20
> > SIP
> >=20
> > during session setup but the application is not necessarily=20
> even SIP=20
> > 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,=20
> > and retargeting.
> > 5. SIP elements may need to apply policy about passing and=20
> screening=20
> > the
> >=20
> > information.
> > 6. Multi-vendor interoperability is important.
> >=20
> > This mechanism is not appropriate in the following situations:
> > 1. SIP behavior is changed.
> > 2. The information is generated and consumed at the SIP=20
> layer by SIP=20
> > elements.
> > 3. SIP elements besides the UAC and UAS might be interested and=20
> > consume the information.
> > 4. There are specific privacy or cross-RAI issues involved with the=20
> > information being transported.
> >=20
> > User data of the mechanism will be clearly marked with the=20
> > application, encoding, semantics, and contents of the data,=20
> allowing=20
> > policy to be applied by UAs.  The working group will define the=20
> > information which each application must specify in a=20
> standards-track=20
> > document to utilize the mechanism.
> >=20
> > One important application of this mechanism is interworking=20
> with the=20
> > ISDN User to User Information Service.  This application defined by=20
> > ITU-T Q.931, is extensively deployed in the PSTN today, supporting=20
> > such applications as contact centers, call centers, and=20
> automatic call=20
> > distributors (ACDs).  A major barrier to the movement of these=20
> > applications to SIP is the lack of a standard mechanism to transport
> > this information in SIP.   For interworking with ISDN, minimal=20
> > information about the content of the UUI is available.  In=20
> this case=20
> > only, the content will just indicate ISDN UUI Service 1=20
> interworking=20
> > rather than the actual content.
> >=20
> > Call control UUI is user information conveyed between user agents=20
> > during
> >=20
> > call control operations.  As a result, the information must be=20
> > conveyed with the INVITE transaction, and must survive proxy=20
> > retargeting, redirection, and transfers.  The mechanism=20
> must utilize a=20
> > minimum of SIP
> >=20
> > extensions as it will need to be supported by many simple SIP user=20
> > agents such as PSTN gateways.  The mechanism must interwork=20
> with the=20
> > existing ISDN service but should also be extensible for use=20
> by other=20
> > applications and non-ISDN protocols.
> >=20
> > While PSTN UUI is carried in ISUP (ISDN User Part), SIP=20
> call control=20
> > UUI
> >=20
> > can be generated by a number of protocols that are not ISUP, and as=20
> > such
> >=20
> > it is architecturally unreasonable to interwork these=20
> protocols at the=20
> > SIP / ISUP gateway. That is, existing SIP-T approaches defined in
> > RFC3372 do not meet the requirements.
> >=20
> > Mechanisms based on the SIP INFO method, RFC2976, will not be=20
> > considered
> >=20
> > by the working group since the UUI contents carry information that=20
> > must be conveyed during session setup at the user agent - the=20
> > information must be conveyed with the INVITE transaction.  The=20
> > information must be passed with the session setup request (INVITE),=20
> > responses to that INVITE, or session termination requests.  As a=20
> > result, it is not possible to use INFO in these cases.
> >=20
> > The group will produce:
> >=20
> > - A problem statement and requirements document for=20
> implementing a SIP=20
> > call control UUI mechanism
> >=20
> > - A specification of the SIP extension to best meet those=20
> requirements.
> >=20
> > Goals and Milestones
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >=20
> > Mar 10 - Problem statement and requirements document to IESG
> > (Informational)
> > Aug 10 - SIP call control UUI specification to IESG (PS)
> >=20
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> > _______________________________________________
> > 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 drage@alcatel-lucent.com  Thu Nov  5 05:20:16 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Leon Portman <Leon.Portman@nice.com>, Alan Johnston <alan@sipstation.com>,  "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 5 Nov 2009 14:20:35 +0100
Thread-Topic: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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Given that the prime purpose of SIP is to support a session description of =
the media, which presumably you want to do the SRTP rekeying for, I would a=
rgue that the CCU is not the appropriate mechanism to support this. I suspe=
ct your SRTP rekeying scenario would also expect to be terminated by some i=
nterworking gateways, whereas the CCU information is meant to pass through =
such gateways.

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Leon Portman
> Sent: Wednesday, November 04, 2009 7:59 AM
> To: Alan Johnston; dispatch@ietf.org
> Subject: Re: [dispatch] Updated CCUS Charter Text - Call=20
> Control UUI for SIP
>=20
> Only one comment, does it also supposed to support re-invite=20
> (for SRTP re-keying or media updates) scenarios as well?
>=20
> Thanks
>=20
> Leon
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Alan Johnston
> Sent: Tuesday, November 03, 2009 11:09 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Updated CCUS Charter Text - Call Control=20
> UUI for SIP
>=20
> All,
>=20
> Here is some updated charter text for CCUS.  Comments are=20
> most welcome.
>=20
> Thanks,
> Alan
>=20
> - - - - -
>=20
> The Call Control UUI for SIP  (CCUS) working group is=20
> chartered to define a SIP mechanism for transporting call=20
> control related user-to-user (UUI) information between User Agents.
>=20
> The mechanism developed in this working group is generally=20
> appropriate in the following situations:
> 1. The information is generated and consumed by an=20
> application using SIP during session setup but the=20
> application is 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,=20
> redirection, and retargeting.
> 5. SIP elements may need to apply policy about passing and=20
> screening the information.
> 6. Multi-vendor interoperability is important.
>=20
> This mechanism is not appropriate in the following situations:
> 1. SIP behavior is changed.
> 2. The information is generated and consumed at the SIP layer=20
> by SIP elements.
> 3. SIP elements besides the UAC and UAS might be interested=20
> and consume the information.
> 4. There are specific privacy or cross-RAI issues involved=20
> with the information being transported.
>=20
> User data of the mechanism will be clearly marked with the=20
> application, encoding, semantics, and contents of the data,=20
> allowing policy to be applied by UAs.  The working group will=20
> define the information which each application must specify in=20
> a standards-track document to utilize the mechanism.
>=20
> One important application of this mechanism is interworking=20
> with the ISDN User to User Information Service.  This=20
> application defined by ITU-T Q.931, is extensively deployed=20
> in the PSTN today, supporting such applications as contact=20
> centers, call centers, and automatic call distributors=20
> (ACDs).  A major barrier to the movement of these=20
> applications to SIP is the lack of a standard mechanism to transport=20
> this information in SIP.   For interworking with ISDN, minimal=20
> information about the content of the UUI is available.  In=20
> this case only, the content will just indicate ISDN UUI=20
> Service 1 interworking rather than the actual content.
>=20
> Call control UUI is user information conveyed between user=20
> agents during call control operations.  As a result, the=20
> information must be conveyed with the INVITE transaction, and=20
> must survive proxy retargeting, redirection, and transfers. =20
> The mechanism must utilize a minimum of SIP extensions as it=20
> will need to be supported by many simple SIP user agents such=20
> as PSTN gateways.  The mechanism must interwork with the=20
> existing ISDN service but should also be extensible for use=20
> by other applications and non-ISDN protocols.
>=20
> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call=20
> control UUI can be generated by a number of protocols that=20
> are not ISUP, and as such it is architecturally unreasonable=20
> to interwork these protocols at the SIP / ISUP gateway. That=20
> is, existing SIP-T approaches defined in
> RFC3372 do not meet the requirements.
>=20
> Mechanisms based on the SIP INFO method, RFC2976, will not be=20
> considered by the working group since the UUI contents carry=20
> information that must be conveyed during session setup at the=20
> user agent - the information must be conveyed with the INVITE=20
> transaction.  The information must be passed with the session=20
> setup request (INVITE), responses to that INVITE, or session=20
> termination requests.  As a result, it is not possible to use=20
> INFO in these cases.
>=20
> The group will produce:
>=20
> - A problem statement and requirements document for=20
> implementing a SIP call control UUI mechanism
>=20
> - A specification of the SIP extension to best meet those=20
> requirements.
>=20
> Goals and Milestones
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> Mar 10 - Problem statement and requirements document to IESG
> (Informational)
> Aug 10 - SIP call control UUI specification to IESG (PS)
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From salvatore.loreto@ericsson.com  Thu Nov  5 07:04:59 2009
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Hi Dale,

sorry to be late answering this.

see my comments in line.

cheer
/Sal

Dale Worley wrote:
> On Mon, 2009-10-12 at 16:49 +0300, Salvatore Loreto wrote:
>   
>> Abstract
>>
>>    SIP Referred-By header field conveys the referrer identity of a
>>    request.  This header field may be used when exploding a SIP MESSAGE
>>    or a SIP INVITE request to a pre-defined group URI and the
>>    P-Asserted-Identity header field or From header field in the exploded
>>    SIP requests needs to carry another value (e.g. the URI of a pre-
>>    defined group, or a conference focus URI).  In those cases, the
>>    Referred-By header field in the resulting exploded requests is set to
>>    the P-Asserted-Identity or to the From header field value of the
>>    original SIP request received before exploding, so as to convey to
>>    the receiver the identity of the original inviting sender.
>>     
>
> What do you do if the original INVITE already has a Referred-By header?
>   
[Sal] this seems to me a general problem of any exploder procedure.
My guess is that you would replace it with the new value!
> It seems to me that you want a new header to carry this information, as
> it has distinct semantics from all existing headers.
>   
[Sal] It is not what we want!
> I am also curious why the From header of the exploded messages would
> need to be different from the From header of the original message.
>   
[Sal]
Yes it is true that RFC5365 says that the exploder MUST include the same
 From header field value as the incoming request but RFC5365 does not
take into consideration exploding when using a pre-defined group. There
needs to be a way for the message recipients to know the name of the 
pre-defined group.

For SIP INVITE the From header contains the URI for the conference (see 
RFC5366),
so the original sender goes in the Referred-By.

In order to keep things a similar as possible between SIP INVITE and SIP
MESSAGE, it makes sense to put the original sender in the Referred-By
when SIP MESSAGE From has to carry a pre-defined group instead of the 
original sender.
> Dale
>
>
>   


From lei.zhu@huawei.com  Fri Nov  6 02:14:10 2009
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Date: Fri, 06 Nov 2009 18:14:17 +0800
From: Zhu Lei <lei.zhu@huawei.com>
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Subject: [dispatch] Individual draft for sigcomp
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Hi all,
 
I just updated an individual draft for signalling compression. This draft
was submitted to ROHC WG of TSV area, and does not receive opposite
comments. 
 
http://tools.ietf.org/html/draft-lei-rohc-sigcomp-static-dictionary-01
 
I think this draft would finally used to compress sip and sdp strings, and
needed.
 
Would you like to review this draft? In addition, we may discuss it during
IETF 76 event.
 
Best regards,
Lei Zhu

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<DIV><SPAN class=489390710-06112009><FONT size=2>Hi all,</FONT></SPAN></DIV>
<DIV><SPAN class=489390710-06112009><FONT size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=489390710-06112009><FONT size=2>I just updated an individual 
draft for signalling compression. This draft was submitted to ROHC WG of TSV 
area, and does not receive opposite comments. </FONT></SPAN></DIV>
<DIV><SPAN class=489390710-06112009><FONT size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=489390710-06112009><FONT size=2><A 
href="http://tools.ietf.org/html/draft-lei-rohc-sigcomp-static-dictionary-01">http://tools.ietf.org/html/draft-lei-rohc-sigcomp-static-dictionary-01</A></FONT></SPAN></DIV>
<DIV><SPAN class=489390710-06112009><FONT size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=489390710-06112009><FONT size=2>I think this draft would 
finally used to compress sip and sdp strings, and needed.</FONT></SPAN></DIV>
<DIV><SPAN class=489390710-06112009><FONT size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=489390710-06112009><FONT size=2>Would you like to review this 
draft? In addition, we may discuss it during IETF 76 event.</FONT></SPAN></DIV>
<DIV><SPAN class=489390710-06112009><FONT size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=489390710-06112009><FONT size=2>Best 
regards,</FONT></SPAN></DIV>
<DIV><SPAN class=489390710-06112009><FONT size=2>Lei 
Zhu</FONT></SPAN></DIV></BODY></HTML>

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From adam@nostrum.com  Fri Nov  6 06:15:23 2009
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Subject: Re: [dispatch] New version for the Alert-Info URNs	draft:	draft-liess-dispatch-alert-info-urns-00.txt
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On 10/28/09 6:22 PM, L.Liess@telekom.de wrote:
> I will take this as an open issue for the dispatch meeting slides. I
> hope Adam will be there because it was his proposal to use combinations
> of discrete items rather than hierarchy.

Sorry for not following this more closely.

At this point, I think we have two different things to tease apart: (1) 
The semantics for combining different dimensions of ringback, and (2) 
the syntax for combining these dimensions.

We really need to understand the first one before we start quibbling 
over the second one. So, while I do have some thoughts about the syntax, 
I'm not going to go into them -- even a little bit -- until we have a 
better grip on the semantic problem.

Here's an important point that Paul raises:
> It seems that what you propose is a multidimensional space, where 
> every point in that space is potentially rendered uniquely. But in 
> practice values will often be supplied for some but not all of the 
> dimensions. In that case we end up specifying some "slice" of the 
> array. I guess in this case you could look to see what you would 
> render for each point in the slice. If those are all identical, then 
> you are in good shape. But if they are not, then what? Do you pick one 
> of the points? That amounts to choosing a default value for each 
> unspecified dimension. Is that the right answer? 

I agree. We're effectively defining an n-dimensional space, and assuming 
that a point can be selected with fewer than n ordinals. And, yes, I 
think the correct answer is simply defining a default value for every 
new dimension defined -- preferably within the specification of the 
dimension itself (although some things, like country indication, would 
probably be better handled as local policy). Then, you can specify a 
value for as few as zero of the dimensions, and still have the answer be 
a point.

The other side of that coin is that clients receiving this information 
will not necessarily understand (or care to support) all of the 
dimensions being presented. I believe that this is handled on the 
recipient's part by ignoring any dimensions it does not understand.

However, I think that Paul's characterization of the difference between 
the previous approach and the current approach isn't correct:
> It seems like what you must be doing is replacing a hierarchy with an 
> N-dimensional matrix which is sparsely populated. That seems harder to 
> deal with. 

I don't agree that it is sparsely populated. Many of the points might 
result in the same answer, but that doesn't mean it's sparsely 
populated. I think you could take the current draft and (modulo the 
issues with 7.3.3) pick any value from each of the three dimensions at 
random, and have a semantically sensible ringtone. Here; I'll roll the 
dice a couple of times: Internal short Brazilian ringback. External 
priority Pakistani ringback. You can play along at home -- find me an 
empty space in the matrix, and I'll concede the point.

So, quite to the contrary, I think the two approaches are *semantically* 
identical, and the only difference between what was proposed before and 
what is proposed now is a *superficial* and *syntactic* difference. If 
you need proof, I'll write a quick perl script that converts between the 
two syntactic representations. :)

/a

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On 10/19/09 9:38 PM, L.Liess@telekom.de wrote:
>     - Alert-Info URNs Indications for country-specific tones
>    

While I think this is a good dimension to have, I question the wisdom of 
replicating the entire ISO 3166-1 registry in an RFC and into IANA, for 
a variety of reasons:

   1. The initial registered list -- the one in the RFC -- will rapidly
      become out of date.

   2. The IANA registry will require constant updating to match the
      changes in ISO 3166-1. This either requires the IANA to keep up
      with changes in that registry (which is a task well beyond what we
      require of them at the moment) or it requires someone to send
      notes to IANA to update the registry whenever a change is made. We
      don't really have the structure necessary to ensure that someone
      will do that.

   3. If IANA and ISO 3166-1 do become out of sync -- which would almost
      necessarily happen from time to time -- you'll end up with
      implementations selecting at random between them (which is not a
      recipe for interoperability).

   4. The selection of country codes is highly political and rife with
      land mines -- some are forseeable (e.g. tw, fk), while others
      aren't necessarily. You can try to side-step this -- like ICANN
      tries -- by claiming that we don't take codes outside of ISO
      3166-1, and that we must accept all codes in ISO 3166-1; however,
      that hasn't kept them out of occasional hot water.


I would strongly suggest simply citing ISO 3166-1, not putting any 
values in the IANA registry, and simply giving a couple of 
representative (and politically uncontroversial) examples.

/a

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On 10/19/09 9:38 PM, <a class="moz-txt-link-abbreviated" href="mailto:L.Liess@telekom.de">L.Liess@telekom.de</a> wrote:
<blockquote
 cite="mid:40FB0FFB97588246A1BEFB05759DC8A0038854EC@S4DE9JSAANI.ost.t-com.de"
 type="cite">
  <pre wrap="">   - Alert-Info URNs Indications for country-specific tones
  </pre>
</blockquote>
<br>
While I think this is a good dimension to have, I question the wisdom
of replicating the entire ISO 3166-1 registry in an RFC and into IANA,
for a variety of reasons:<br>
<br>
<ol>
  <li>The initial registered list -- the one in the RFC -- will rapidly
become out of date.<br>
    <br>
  </li>
  <li>The IANA registry will require constant updating to match the
changes in ISO 3166-1. This either requires the IANA to keep up with
changes in that registry (which is a task well beyond what we require
of them at the moment) or it requires someone to send notes to IANA to
update the registry whenever a change is made. We don't really have the
structure necessary to ensure that someone will do that.<br>
    <br>
  </li>
  <li>If IANA and ISO 3166-1 do become out of sync -- which would
almost necessarily happen from time to time -- you'll end up with
implementations selecting at random between them (which is not a recipe
for interoperability).<br>
    <br>
  </li>
  <li>The selection of country codes is highly political and rife with
land mines -- some are forseeable (e.g. tw, fk), while others aren't
necessarily. You can try to side-step this -- like ICANN tries -- by
claiming that we don't take codes outside of ISO 3166-1, and that we
must accept all codes in ISO 3166-1; however, that hasn't kept them out
of occasional hot water.</li>
</ol>
<br>
I would strongly suggest simply citing ISO 3166-1, not putting any
values in the IANA registry, and simply giving a couple of
representative (and politically uncontroversial) examples.<br>
<br>
/a<br>
</body>
</html>

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From adam@nostrum.com  Fri Nov  6 06:40:21 2009
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
> Hi Christer and Francois,
> I have added a sentence under section Overall Applicability:
>
>
> The appearance of the Reason header is applicable to final responses 4xx, 5xx and 6xx and in addition for 199 Responses.
>
> Is this proper enough? Or do you have more in mind?
>    

Given that RFC 3398 specifies the generation of a 301 in response to a 
cause code of 22 under certain circumstances, it seems pretty 
appropriate to allow the inclusion of a Q.850 cause code in 300-class 
responses also. Or, at least, as appropriate as it would be to include 
them in 4xx, 5xx, and 6xx responses.

/a

From pkyzivat@cisco.com  Fri Nov  6 06:59:48 2009
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: anwars@avaya.com, dispatch@ietf.org, R.Jesske@telekom.de, L.Liess@telekom.de
Subject: Re: [dispatch] New version for the Alert-Info URNs	draft:	draft-liess-dispatch-alert-info-urns-00.txt
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I agree. It would be even better is there were already an URN 
designating country code. Then it could simply be referenced.
But there doesn't seem to be one right now.

Perhaps this doc could define one, and cite ISO 3166-1 for the 
definition of the values it can contain. It could then be used anywhere 
one wants to reference a country by URN, not just in Alert-Info.

	Thanks,
	Paul

Adam Roach wrote:
> On 10/19/09 9:38 PM, L.Liess@telekom.de wrote:
>>    - Alert-Info URNs Indications for country-specific tones
>>   
> 
> While I think this is a good dimension to have, I question the wisdom of 
> replicating the entire ISO 3166-1 registry in an RFC and into IANA, for 
> a variety of reasons:
> 
>    1. The initial registered list -- the one in the RFC -- will rapidly
>       become out of date.
> 
>    2. The IANA registry will require constant updating to match the
>       changes in ISO 3166-1. This either requires the IANA to keep up
>       with changes in that registry (which is a task well beyond what we
>       require of them at the moment) or it requires someone to send
>       notes to IANA to update the registry whenever a change is made. We
>       don't really have the structure necessary to ensure that someone
>       will do that.
> 
>    3. If IANA and ISO 3166-1 do become out of sync -- which would almost
>       necessarily happen from time to time -- you'll end up with
>       implementations selecting at random between them (which is not a
>       recipe for interoperability).
> 
>    4. The selection of country codes is highly political and rife with
>       land mines -- some are forseeable (e.g. tw, fk), while others
>       aren't necessarily. You can try to side-step this -- like ICANN
>       tries -- by claiming that we don't take codes outside of ISO
>       3166-1, and that we must accept all codes in ISO 3166-1; however,
>       that hasn't kept them out of occasional hot water.
> 
> 
> I would strongly suggest simply citing ISO 3166-1, not putting any 
> values in the IANA registry, and simply giving a couple of 
> representative (and politically uncontroversial) examples.
> 
> /a
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From L.Liess@telekom.de  Fri Nov  6 07:59:21 2009
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From: <L.Liess@telekom.de>
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It's a good proposal. I fully agree.=20
=20
Laura




________________________________

	From: dispatch-bounces@ietf.org
[mailto:dispatch-bounces@ietf.org] On Behalf Of Adam Roach
	Sent: Friday, November 06, 2009 1:31 AM
	To: Liess, Laura
	Cc: anwars@avaya.com; dispatch@ietf.org; Jesske, Roland
	Subject: Re: [dispatch] New version for the Alert-Info
URNsdraft: draft-liess-dispatch-alert-info-urns-00.txt
=09
=09
	On 10/19/09 9:38 PM, L.Liess@telekom.de wrote:=20

		   - Alert-Info URNs Indications for country-specific
tones
		 =20


	While I think this is a good dimension to have, I question the
wisdom of replicating the entire ISO 3166-1 registry in an RFC and into
IANA, for a variety of reasons:
=09
=09

	1.	The initial registered list -- the one in the RFC --
will rapidly become out of date.
	=09
	=09
	2.	The IANA registry will require constant updating to
match the changes in ISO 3166-1. This either requires the IANA to keep
up with changes in that registry (which is a task well beyond what we
require of them at the moment) or it requires someone to send notes to
IANA to update the registry whenever a change is made. We don't really
have the structure necessary to ensure that someone will do that.
	=09
	=09
	3.	If IANA and ISO 3166-1 do become out of sync -- which
would almost necessarily happen from time to time -- you'll end up with
implementations selecting at random between them (which is not a recipe
for interoperability).
	=09
	=09
	4.	The selection of country codes is highly political and
rife with land mines -- some are forseeable (e.g. tw, fk), while others
aren't necessarily. You can try to side-step this -- like ICANN tries --
by claiming that we don't take codes outside of ISO 3166-1, and that we
must accept all codes in ISO 3166-1; however, that hasn't kept them out
of occasional hot water.=20


	I would strongly suggest simply citing ISO 3166-1, not putting
any values in the IANA registry, and simply giving a couple of
representative (and politically uncontroversial) examples.
=09
	/a
=09


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<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3603" name=3DGENERATOR></HEAD>
<BODY text=3D#000000 bgColor=3D#ffffff>
<DIV dir=3Dltr align=3Dleft>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D085045615-06112009><SPAN class=3D902385815-06112009>It's =
</SPAN>a good=20
proposal. </SPAN>I&nbsp;<SPAN class=3D085045615-06112009>fully=20
agree</SPAN>.&nbsp;</FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D085045615-06112009><FONT face=3DArial color=3D#0000ff =

size=3D2>Laura</FONT></SPAN></DIV>
<DIV><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT><BR></DIV></DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Dde 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>Adam=20
  Roach<BR><B>Sent:</B> Friday, November 06, 2009 1:31 AM<BR><B>To:</B> =
Liess,=20
  Laura<BR><B>Cc:</B> anwars@avaya.com; dispatch@ietf.org; Jesske,=20
  Roland<BR><B>Subject:</B> Re: [dispatch] New version for the =
Alert-Info=20
  URNsdraft: =
draft-liess-dispatch-alert-info-urns-00.txt<BR></FONT><BR></DIV>
  <DIV></DIV>On 10/19/09 9:38 PM, <A class=3Dmoz-txt-link-abbreviated=20
  href=3D"mailto:L.Liess@telekom.de">L.Liess@telekom.de</A> wrote:=20
  <BLOCKQUOTE=20
  =
cite=3Dmid:40FB0FFB97588246A1BEFB05759DC8A0038854EC@S4DE9JSAANI.ost.t-com=
.de=20
  type=3D"cite"><PRE wrap=3D"">   - Alert-Info URNs Indications for =
country-specific tones
  </PRE></BLOCKQUOTE><BR>While I think this is a good dimension to have, =
I=20
  question the wisdom of replicating the entire ISO 3166-1 registry in =
an RFC=20
  and into IANA, for a variety of reasons:<BR><BR>
  <OL>
    <LI>The initial registered list -- the one in the RFC -- will =
rapidly become=20
    out of date.<BR><BR>
    <LI>The IANA registry will require constant updating to match the =
changes in=20
    ISO 3166-1. This either requires the IANA to keep up with changes in =
that=20
    registry (which is a task well beyond what we require of them at the =
moment)=20
    or it requires someone to send notes to IANA to update the registry =
whenever=20
    a change is made. We don't really have the structure necessary to =
ensure=20
    that someone will do that.<BR><BR>
    <LI>If IANA and ISO 3166-1 do become out of sync -- which would =
almost=20
    necessarily happen from time to time -- you'll end up with =
implementations=20
    selecting at random between them (which is not a recipe for=20
    interoperability).<BR><BR>
    <LI>The selection of country codes is highly political and rife with =
land=20
    mines -- some are forseeable (e.g. tw, fk), while others aren't =
necessarily.=20
    You can try to side-step this -- like ICANN tries -- by claiming =
that we=20
    don't take codes outside of ISO 3166-1, and that we must accept all =
codes in=20
    ISO 3166-1; however, that hasn't kept them out of occasional hot =
water.=20
  </LI></OL><BR>I would strongly suggest simply citing ISO 3166-1, not =
putting=20
  any values in the IANA registry, and simply giving a couple of =
representative=20
  (and politically uncontroversial)=20
examples.<BR><BR>/a<BR></BLOCKQUOTE></BODY></HTML>

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From spencer@wonderhamster.org  Fri Nov  6 12:22:46 2009
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Subject: Re: [dispatch] New version for the Alert-InfoURNs	draft:	draft-liess-dispatch-alert-info-urns-00.txt
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I agree with Adam's point here (matrix isn't sparse, but a lot of the 
n-dimensional points are really boring because they have the same value as 
other n-dimensional points).

I was seeing this the way Paul was seeing it previously, so this was a 
helpful insight.

Thanks,

Spencer

>> I will take this as an open issue for the dispatch meeting slides. I
>> hope Adam will be there because it was his proposal to use combinations
>> of discrete items rather than hierarchy.
>
> Sorry for not following this more closely.
>
> At this point, I think we have two different things to tease apart: (1) 
> The semantics for combining different dimensions of ringback, and (2) the 
> syntax for combining these dimensions.
>
> We really need to understand the first one before we start quibbling over 
> the second one. So, while I do have some thoughts about the syntax, I'm 
> not going to go into them -- even a little bit -- until we have a better 
> grip on the semantic problem.
>
> Here's an important point that Paul raises:
>> It seems that what you propose is a multidimensional space, where every 
>> point in that space is potentially rendered uniquely. But in practice 
>> values will often be supplied for some but not all of the dimensions. In 
>> that case we end up specifying some "slice" of the array. I guess in this 
>> case you could look to see what you would render for each point in the 
>> slice. If those are all identical, then you are in good shape. But if 
>> they are not, then what? Do you pick one of the points? That amounts to 
>> choosing a default value for each unspecified dimension. Is that the 
>> right answer?
>
> I agree. We're effectively defining an n-dimensional space, and assuming 
> that a point can be selected with fewer than n ordinals. And, yes, I think 
> the correct answer is simply defining a default value for every new 
> dimension defined -- preferably within the specification of the dimension 
> itself (although some things, like country indication, would probably be 
> better handled as local policy). Then, you can specify a value for as few 
> as zero of the dimensions, and still have the answer be a point.
>
> The other side of that coin is that clients receiving this information 
> will not necessarily understand (or care to support) all of the dimensions 
> being presented. I believe that this is handled on the recipient's part by 
> ignoring any dimensions it does not understand.
>
> However, I think that Paul's characterization of the difference between 
> the previous approach and the current approach isn't correct:
>> It seems like what you must be doing is replacing a hierarchy with an 
>> N-dimensional matrix which is sparsely populated. That seems harder to 
>> deal with.
>
> I don't agree that it is sparsely populated. Many of the points might 
> result in the same answer, but that doesn't mean it's sparsely populated. 
> I think you could take the current draft and (modulo the issues with 
> 7.3.3) pick any value from each of the three dimensions at random, and 
> have a semantically sensible ringtone. Here; I'll roll the dice a couple 
> of times: Internal short Brazilian ringback. External priority Pakistani 
> ringback. You can play along at home -- find me an empty space in the 
> matrix, and I'll concede the point.
>
> So, quite to the contrary, I think the two approaches are *semantically* 
> identical, and the only difference between what was proposed before and 
> what is proposed now is a *superficial* and *syntactic* difference. If you 
> need proof, I'll write a quick perl script that converts between the two 
> syntactic representations. :)
>
> /a 


From R.Jesske@telekom.de  Sun Nov  8 00:16:47 2009
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From: <R.Jesske@telekom.de>
To: <adam@nostrum.com>
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Cc: audet@nortel.com, dispatch@ietf.org, gonzalo.camarillo@ericsson.com
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Hi Adam,
Thank you for your mail. So I propose the following text then:

The appearance of the Reason header is applicable to final responses
      3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as defined =
within
      ietf-sipcore-199.txt. Also the Reason header is
      applicable to provisional responses18x which are generated by an
      interworking gateway from ISUP to SIP.</t>

Best Beagards

Roland=20

-----Urspr=FCngliche Nachricht-----
Von: Adam Roach [mailto:adam@nostrum.com]=20
Gesendet: Freitag, 6. November 2009 01:43
An: Jesske, Roland
Cc: audet@nortel.com; christer.holmberg@ericsson.com; =
gonzalo.camarillo@ericsson.com; dispatch@ietf.org
Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
> Hi Christer and Francois,
> I have added a sentence under section Overall Applicability:
>
>
> The appearance of the Reason header is applicable to final responses =
4xx, 5xx and 6xx and in addition for 199 Responses.
>
> Is this proper enough? Or do you have more in mind?
>   =20

Given that RFC 3398 specifies the generation of a 301 in response to a=20
cause code of 22 under certain circumstances, it seems pretty=20
appropriate to allow the inclusion of a Q.850 cause code in 300-class=20
responses also. Or, at least, as appropriate as it would be to include=20
them in 4xx, 5xx, and 6xx responses.

/a

From mary.barnes@nortel.com  Sun Nov  8 03:56:28 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: <dispatch@ietf.org>
Subject: [dispatch] Reminder: Identity Adhoc - Monday, Nov 9th, 12:00-13:00, Acacia East room
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Hi all,

Per the agenda, the first of 3 DISPATCH adhocs is scheduled for =
tomorrow:
http://www.ietf.org/proceedings/09nov/agenda/dispatch.html

We shifted the start time to allow folks to grab some lunch before the =
session.=20

The charts for the session entitled "Identity Adhoc"  have been uploaded =
to the meeting materials webpage for the DISPATCH WG:
https://datatracker.ietf.org/meeting/76/materials.html

Regards,
Mary.=20

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<P><FONT SIZE=3D2>Hi all,<BR>
<BR>
Per the agenda, the first of 3 DISPATCH adhocs is scheduled for =
tomorrow:<BR>
<A =
HREF=3D"http://www.ietf.org/proceedings/09nov/agenda/dispatch.html">http:=
//www.ietf.org/proceedings/09nov/agenda/dispatch.html</A><BR>
<BR>
We shifted the start time to allow folks to grab some lunch before the =
session.<BR>
<BR>
The charts for the session entitled &quot;Identity Adhoc&quot;&nbsp; =
have been uploaded to the meeting materials webpage for the DISPATCH =
WG:<BR>
<A =
HREF=3D"https://datatracker.ietf.org/meeting/76/materials.html">https://d=
atatracker.ietf.org/meeting/76/materials.html</A><BR>
<BR>
Regards,<BR>
Mary. </FONT>
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From gonzalo.camarillo@ericsson.com  Sun Nov  8 06:52:55 2009
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Hi,

the hotel has announced a quick lunch service for IETF folks. That may 
be an option people want to consider.

Cheers,

Gonzalo

Mary Barnes wrote:
> Hi all,
> 
> Per the agenda, the first of 3 DISPATCH adhocs is scheduled for tomorrow:
> http://www.ietf.org/proceedings/09nov/agenda/dispatch.html
> 
> We shifted the start time to allow folks to grab some lunch before the 
> session.
> 
> The charts for the session entitled "Identity Adhoc"  have been uploaded 
> to the meeting materials webpage for the DISPATCH WG:
> https://datatracker.ietf.org/meeting/76/materials.html
> 
> Regards,
> Mary.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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R.Jesske@telekom.de wrote:
> Hi Adam,
> Thank you for your mail. So I propose the following text then:
> 
> The appearance of the Reason header is applicable to final responses
>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as defined within
>       ietf-sipcore-199.txt. Also the Reason header is
>       applicable to provisional responses18x which are generated by an
>       interworking gateway from ISUP to SIP.</t>

I think this is ok *except* if you are going to allow one kind of UAS to 
use Reason in a 18x response, then it should be allowed for *any* UAS to 
do so.

For instance, suppose the gateway is replaced by a B2BUA to another sip 
environment, and the request eventually arrives as a gateway. Then a 18x 
response with a Reason header may arrive at the B2BUA and need to be 
replicated on the other side.

	Thanks,
	Paul

> Best Beagards
> 
> Roland 
> 
> -----Ursprüngliche Nachricht-----
> Von: Adam Roach [mailto:adam@nostrum.com] 
> Gesendet: Freitag, 6. November 2009 01:43
> An: Jesske, Roland
> Cc: audet@nortel.com; christer.holmberg@ericsson.com; gonzalo.camarillo@ericsson.com; dispatch@ietf.org
> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> 
> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
>> Hi Christer and Francois,
>> I have added a sentence under section Overall Applicability:
>>
>>
>> The appearance of the Reason header is applicable to final responses 4xx, 5xx and 6xx and in addition for 199 Responses.
>>
>> Is this proper enough? Or do you have more in mind?
>>    
> 
> Given that RFC 3398 specifies the generation of a 301 in response to a 
> cause code of 22 under certain circumstances, it seems pretty 
> appropriate to allow the inclusion of a Q.850 cause code in 300-class 
> responses also. Or, at least, as appropriate as it would be to include 
> them in 4xx, 5xx, and 6xx responses.
> 
> /a
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From dworley@nortel.com  Sun Nov  8 21:11:42 2009
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Subject: Re: [dispatch] New Version Notification for draft-worley-instrument-identification-00
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I've written up an internet-draft describing how the sipXecs open-source
PBX identifies which telephone makes each registration, which allows us
to obtain information about specific telephones rather than about AORs.
For example, "SUBSCRIBE sip:~~in~0000DEADBEEF@example.net" will get the
complete dialog status for the telephone with MAC address 0000DEADBEEF,
even if the AORs that appear on it appear on other telephones.

The mechanism works well in practice, but it's a hack, since it involves
inserting the phone's MAC address into the authorization user-name
field.  What we want to do is use the phone's "SIP instance" URI to
identify itself, but there are a number of problems involved in doing
that.  It doesn't look like they are particularly difficult, but they
would require consensus and getting the phone vendors to bring their
devices into line.  See section 5 of the draft, which I have copied into
the bottom of this message.

The announcement of the I-D follows, and then section 5, which I want
comments on.

Dale


On Sun, 2009-10-18 at 09:50 -0700, IETF I-D Submission Tool wrote:
> A new version of I-D, draft-worley-instrument-identification-00.txt has been successfuly submitted by Dale Worley and posted to the IETF repository.
> 
> Filename:	 draft-worley-instrument-identification
> Revision:	 00
> Title:		 Identifying Individual Telephone Instruments in SIP
> Creation_date:	 2009-10-18
> WG ID:		 Independent Submission
> Number_of_pages: 18
> 
> http://www.ietf.org/id/draft-worley-instrument-identification-00.txt
>
> Abstract:
> When requesting and reporting dialog status in SIP, users often want
> to know the status of a particular telephone instrument, rather than
> the status of an Address of Record (AOR).  The architecture of SIP
> makes it difficult to obtain the status of a telephone instrument in
> a way that is convenient for use in common situations, in particular,
> in an organization's PBX.  This document describes a method for
> identifying which telephone instrument is making each registration
> request that is convenient to deploy in PBXs.  This information can
> be used to obtain the status of individual telephone instruments.
> 
> Unfortunately, although the method works well in practice, it
> violates separation of concerns by carrying instrument identification
> information in an authentication-related field.  This draft includes
> a preliminary discussion of better ways to identify instruments.


5.  Future Improvements

   As well as this method works, it is a hack that violates separation
   of concerns because instrument identification information is carried
   in a field which is intended for authentication information.  This
   requires all elements that need to authenticate the source of a
   request to understand the syntax of the authentication user name.
   This burden is smaller for centralized systems, but in decentralized
   systems, there may be a large number of elements that need to
   authenticate requests, requiring a broad distribution of information
   that in principle should be encapsulated in the "user name and
   password" database.

   The philosophically ideal way to identify instruments is via the SIP
   "instance id".  The instance id is used to support the "SIP outbound"
   mechanism[outbound], the GRUU mechanism[gruu], and the SIP Forum's UA
   configuration system[ua-config].

   There are two distinct ways to use the SIP instance id to support
   instrument identification:

   1.  The instance id is carried in the "+sip.instance" field parameter
       of each Contact in each REGISTER request.  The proxy/registrar
       can extract the instance id and use it to label each registration
       database record.  Once the registrations are labelled with
       instrument identifications, requests can be directed to special
       URIs that route to the registered contacts of a specific
       instrument.

   2.  Each instrument can register a single special contact which is
       intended to identify the instrument as a whole rather than any
       line appearance on the instrument.  This contact can be
       recognized by containing the instance id as either a component of
       the contact URI or as a "+sip.instance" field parameter.  With
       this method, the functions described in this document are be
       implemented by sending SIP requests to this special contact.
       E.g., the overall dialog status of the instrument would be
       obtained by a dialog subscription to the special contact.

   A number of problems need to be overcome before the instance id can
   be used to provide instrument identification:

   1.  While the authentication user name is a baseline part of SIP and
       is always configurable by a PBX's provisioning system, the SIP
       instance id is not part of baseline SIP, and user agents need to
       be upgraded to support the SIP instance id.  In practice, only
       some makes of instrument support SIP instance ids.

   2.  Ideally, all line appearances on an instrument would use the same
       instance id.  Whether this is required by the RFCs is not
       entirely clear (probably because there has never been any reason
       to debate doing so).  In practice, some instruments do use
       different instance ids for each line appearance that they
       register.  However, since the preferred format of instance ids is
       a URN derived from a UUID, which is in turn based on the MAC
       address of the instrument, it is straightforward to extract the
       instrument's MAC address from each instance id, and the extracted
       MAC addresses of the various line appearances will be the same.

   3.  The PBX's provisioning system needs to be able to map instance
       ids into its internal instrument identities, which likely
       requires having the user enter the instrument identifiers into
       the provisioning system.  It would be inconvenient to use full
       instance id URNs for this reason: UUID-based URNs are quite long
       and tedious to type.  However, using the base MAC addresses of
       the URNs as the instrument identifiers would be convenient, as
       they are short enough to be convenient to enter and most SIP
       phones are clearly labeled with their MAC address

   4.  The reverse problem can happen with software-based instruments:
       The provisioning system needs to output an instrument identifier
       which can be input into the instrument, as the network interface
       of the hardware platform is not "owned" by the instrument, and
       the instrument may not be the only one on the platform.  In the
       method of this document, doing so is relatively simple, since the
       instrument's interface allows the authentication user name to be
       entered directly by the user.  To utilize the instance id, the
       instrument needs to allow its instance id to be configured, or
       else it needs to output its instance id so that it can be entered
       into the provisioning system.



From alan@sipstation.com  Sun Nov  8 22:40:33 2009
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Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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Keith,

Thank you for your careful read and comments on the charter text.

See my comments below.  Your last point below is very important and it 
would be great to get some opinions from others.

Thanks,
Alan

DRAGE, Keith (Keith) wrote:
> 1)	"4. The information is expected to survive retargeting, redirection, and retargeting." 
>
> "retargeting" appears twice.
>   

Yes, and the information must survive if the retargeting happens twice, 
too ;-)

> 2)	"For interworking with ISDN, minimal 
> information about the content of the UUI is available.  In this case only, the content will just indicate ISDN UUI Service 1 interworking rather than the actual content."
>
> This could imply that the existing protocol discriminator, which does carry information about the content, is lost. I hope that is not true.
>   

No, we will not loose the protocol discriminator.  I didn't think it 
provided that much information about the content, but whatever 
information it provides will be available.

> 3)	I am somewhat concerned that the focus now appears to be on SIP UA to SIP UA, rather than interworking with other signalling systems that also carry such information. Where we have a SIP UA talking to a SIP UA, For at least some of the usages where a SIP UA is talking to a SIP UA, we may well already have SIP mechanisms in place to support this. 

The previous charter text had an ISDN interworking focus, but the 
consensus in the DISPATCH meeting was that there was strong interest in 
UUI outside ISDN interworking.

> In general, we should be focussing on those existing usages transported in other protocols, and any new pure SIP usage should go through the normal SIP extension considerations. With the focus as it currently appears, we could end up spending years defining a fully proper solution, whereas I believe we need something quicker and potentially less than perfect for the SIP UA to SIP UA scenarios. Note that by SIP UA I am talking about a real SIP endpoint here, not the UA that may form part of an interworking gateway.
>   
This is an important scope issue for the charter.

I understand your concern.  I am hopeful that we will not end up with a 
complicated solution and that the mechanism will work for both ISDN and 
non-ISDN approaches.

I think that enough people care strongly about the ISDN interworking 
case that if the mechanism discussions bog down, that the WG could 
recharter to deliver two mechanisms - one for ISDN interworking and 
another more  general mechanism.  However, I'd prefer not to start with 
that approach until we have proven that a single mechanism is too complex.

Do others have an opinion on this?

> regards
>
> Keith
>
>   
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Alan Johnston
>> Sent: Tuesday, November 03, 2009 9:09 PM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Updated CCUS Charter Text - Call Control 
>> UUI for SIP
>>
>> All,
>>
>> Here is some updated charter text for CCUS.  Comments are 
>> most welcome.
>>
>> Thanks,
>> Alan
>>
>> - - - - -
>>
>> The Call Control UUI for SIP  (CCUS) working group is 
>> chartered to define a SIP mechanism for transporting call 
>> control related user-to-user (UUI) information between User Agents.
>>
>> The mechanism developed in this working group is generally 
>> appropriate in the following situations:
>> 1. The information is generated and consumed by an 
>> application using SIP during session setup but the 
>> application is 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. Multi-vendor interoperability is important.
>>
>> This mechanism is not appropriate in the following situations:
>> 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.
>>
>> User data of the mechanism will be clearly marked with the 
>> application, encoding, semantics, and contents of the data, 
>> allowing policy to be applied by UAs.  The working group will 
>> define the information which each application must specify in 
>> a standards-track document to utilize the mechanism.
>>
>> One important application of this mechanism is interworking 
>> with the ISDN User to User Information Service.  This 
>> application defined by ITU-T Q.931, is extensively deployed 
>> in the PSTN today, supporting such applications as contact 
>> centers, call centers, and automatic call distributors 
>> (ACDs).  A major barrier to the movement of these 
>> applications to SIP is the lack of a standard mechanism to transport 
>> this information in SIP.   For interworking with ISDN, minimal 
>> information about the content of the UUI is available.  In 
>> this case only, the content will just indicate ISDN UUI 
>> Service 1 interworking rather than the actual content.
>>
>> Call control UUI is user information conveyed between user 
>> agents during call control operations.  As a result, the 
>> information must be conveyed with the INVITE transaction, and 
>> must survive proxy retargeting, redirection, and transfers.  
>> The mechanism must utilize a minimum of SIP extensions as it 
>> will need to be supported by many simple SIP user agents such 
>> as PSTN gateways.  The mechanism must interwork with the 
>> existing ISDN service but should also be extensible for use 
>> by other applications and non-ISDN protocols.
>>
>> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call 
>> control UUI can be generated by a number of protocols that 
>> are not ISUP, and as such it is architecturally unreasonable 
>> to interwork these protocols at the SIP / ISUP gateway. That 
>> is, existing SIP-T approaches defined in
>> RFC3372 do not meet the requirements.
>>
>> Mechanisms based on the SIP INFO method, RFC2976, will not be 
>> considered by the working group since the UUI contents carry 
>> information that must be conveyed during session setup at the 
>> user agent - the information must be conveyed with the INVITE 
>> transaction.  The information must be passed with the session 
>> setup request (INVITE), responses to that INVITE, or session 
>> termination requests.  As a result, it is not possible to use 
>> INFO in these cases.
>>
>> The group will produce:
>>
>> - A problem statement and requirements document for 
>> implementing a SIP call control UUI mechanism
>>
>> - A specification of the SIP extension to best meet those 
>> requirements.
>>
>> Goals and Milestones
>> ====================
>>
>> Mar 10 - Problem statement and requirements document to IESG
>> (Informational)
>> Aug 10 - SIP call control UUI specification to IESG (PS)
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>>     


From jdrosen@jdrosen.net  Sun Nov  8 23:43:12 2009
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Subject: [dispatch] Introducing ViPR: a new federation technology
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Folks,

As you are all well aware, SIP has been very successful in the 
marketplace as a tool for voice and video within a single domain. Its 
success for inter-domain federation has been more limited, and it has 
seen almost no deployment over the public Internet for any-to-any 
federation, even though this was the model originally envisaged for SIP, 
and a model I still believe in.

This is something that needed to be fixed, and I have fixed it.

Today, Cullen and I submitted documents describing a new technology 
called ViPR. ViPR is a new federation technique which will finally 
enable the any-to-any SIP federation that we have always wanted for SIP. 
It solves several hard problems that have prevented widespread 
federation. In particular, it solves the phone number routing problem, 
providing a non-centralized technique that securely maps phone numbers 
to domains. It also provides a built-in solution for the VoIP anti-spam 
problem - a new one that is specific to VoIP. As such, it enables 
worldwide, scalable, any-to-any federation over the public Internet, 
securely, for phone numbers, and in a way that supports incremental 
deployability.

This technology is not just documents - it is working code, and was 
announced by Cisco today as part of their products shipping early next year.

I'm extremely excited about ViPR, and I think others will be too.

The documents are:


http://www.ietf.org/id/draft-rosenberg-dispatch-vipr-overview-01.txt
http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-vap-00.txt
http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-pvp-00.txt
http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-sip-antispam-00.txt
http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-reload-usage-00.txt 


Thanks,
Jonathan Rosenberg
jdrosen@jdrosen.net

From scott.lawrence@nortel.com  Mon Nov  9 07:00:49 2009
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From: "Scott Lawrence" <scott.lawrence@nortel.com>
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Subject: Re: [dispatch] Charter proposal: Session Recording Protocol
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On Mon, 2009-09-21 at 08:33 -0500, Vijay K. Gurbani 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.

I'd like to register my support for this work.  I think that the draft
does a good job laying out the requirements, and that interoperable
solutions to these problems would be of benefit to everyone.

I would like to have architectures considered that don't rely on a
middlebox of any kind for this, with dataflows like:


 +-------------+                   SIP                       +-------------+
 |             |<------------------------------------------->|             |
 |             |                                             |             |
 |     UA-B    |<---------+                          +------>|     UA-C    |
 |             |          | RTP                  RTP |       |             |
 +-------------+          |    +--------------+      |       +-------------+ 
    |                     |    |   Recorder   |      |
    | Recording Control   +--->|     (RS)     |<-----+         
    | Call Metadata events     |              | 
    +------------------------->+--------------+ 
           (SIP)             

I won't be able to participate even remotely during the DISPATCH session
on this, but would like to point out that in many ways this fits the
description of BLISS work items: it is fundamentally a calling feature
dealing with switching call legs appropriately among participants.
Whether to establish a dedicated working group or not is up to the ADs;
whether they decide to send it to BLISS or to charter a new group, I
look forward to participating.



From pkyzivat@cisco.com  Mon Nov  9 08:19:04 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] Updated CCUS Charter Text - Call Control UUI for SIP
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Comment at end

Alan Johnston wrote:

>> 3)    I am somewhat concerned that the focus now appears to be on SIP 
>> UA to SIP UA, rather than interworking with other signalling systems 
>> that also carry such information. Where we have a SIP UA talking to a 
>> SIP UA, For at least some of the usages where a SIP UA is talking to a 
>> SIP UA, we may well already have SIP mechanisms in place to support this. 
> 
> The previous charter text had an ISDN interworking focus, but the 
> consensus in the DISPATCH meeting was that there was strong interest in 
> UUI outside ISDN interworking.
> 
>> In general, we should be focussing on those existing usages 
>> transported in other protocols, and any new pure SIP usage should go 
>> through the normal SIP extension considerations. With the focus as it 
>> currently appears, we could end up spending years defining a fully 
>> proper solution, whereas I believe we need something quicker and 
>> potentially less than perfect for the SIP UA to SIP UA scenarios. Note 
>> that by SIP UA I am talking about a real SIP endpoint here, not the UA 
>> that may form part of an interworking gateway.
>>   
> This is an important scope issue for the charter.
> 
> I understand your concern.  I am hopeful that we will not end up with a 
> complicated solution and that the mechanism will work for both ISDN and 
> non-ISDN approaches.
> 
> I think that enough people care strongly about the ISDN interworking 
> case that if the mechanism discussions bog down, that the WG could 
> recharter to deliver two mechanisms - one for ISDN interworking and 
> another more  general mechanism.  However, I'd prefer not to start with 
> that approach until we have proven that a single mechanism is too complex.
> 
> Do others have an opinion on this?

My concern all along has been that if one focuses on ISDN interworking,
then what happens when the user that used to have an ISDN device decides 
to upgrade to a new SIP device, and we then have e2e sip connectivity?

The likely scenario is that *nothing* will change. The new device will 
be expected to emulate the behavior of the ISDN device including using 
the ISDN-style UUI mechanism, even though it is not "natural" in an 
all-sip environment.

OTOH, if you start with a UUI mechanism that is "sip-friendly" but can 
also be translated to the ISDN mechanisms when necessary, then the 
migration is much smoother. And you might avoid having ISDN-isms 
lingering after ISDN itself is gone.

The argument I hear against this is that the ISDN encoding itself is 
opaque - pure tunneling of bits. But the protocol discriminator is 
supposed to deal with that isn't it? The values that the discriminator 
can take on, and the encoding of the remainder for each discriminator 
value but be standardized somewhere.

I think a "natural" encoding for sip would have to consider each 
discriminator value, and what an appropriate encoding would be for that 
case.

	Thanks,
	Paul

From richard@shockey.us  Mon Nov  9 09:05:29 2009
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Subject: Re: [dispatch] Introducing ViPR: a new federation technology
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OK Asterisk DUNDi on steroids. 

Useful for enterprise to enterprise supply chain federations, since TRIP was
never going to work there, but to a SIP service provider it provides no real
value.

Did I miss something?

>  -----Original Message-----
>  From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>  Behalf Of Jonathan Rosenberg
>  Sent: Monday, November 09, 2009 2:44 AM
>  To: dispatch@ietf.org
>  Subject: [dispatch] Introducing ViPR: a new federation technology
>  
>  Folks,
>  
>  As you are all well aware, SIP has been very successful in the
>  marketplace as a tool for voice and video within a single domain. Its
>  success for inter-domain federation has been more limited, and it has
>  seen almost no deployment over the public Internet for any-to-any
>  federation, even though this was the model originally envisaged for
>  SIP,
>  and a model I still believe in.
>  
>  This is something that needed to be fixed, and I have fixed it.
>  
>  Today, Cullen and I submitted documents describing a new technology
>  called ViPR. ViPR is a new federation technique which will finally
>  enable the any-to-any SIP federation that we have always wanted for
>  SIP.
>  It solves several hard problems that have prevented widespread
>  federation. In particular, it solves the phone number routing problem,
>  providing a non-centralized technique that securely maps phone numbers
>  to domains. It also provides a built-in solution for the VoIP anti-
>  spam
>  problem - a new one that is specific to VoIP. As such, it enables
>  worldwide, scalable, any-to-any federation over the public Internet,
>  securely, for phone numbers, and in a way that supports incremental
>  deployability.
>  
>  This technology is not just documents - it is working code, and was
>  announced by Cisco today as part of their products shipping early next
>  year.
>  
>  I'm extremely excited about ViPR, and I think others will be too.
>  
>  The documents are:
>  
>  
>  http://www.ietf.org/id/draft-rosenberg-dispatch-vipr-overview-01.txt
>  http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-vap-
>  00.txt
>  http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-pvp-
>  00.txt
>  http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-sip-
>  antispam-00.txt
>  http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-
>  reload-usage-00.txt
>  
>  
>  Thanks,
>  Jonathan Rosenberg
>  jdrosen@jdrosen.net
>  _______________________________________________
>  dispatch mailing list
>  dispatch@ietf.org
>  https://www.ietf.org/mailman/listinfo/dispatch


From kpfleming@digium.com  Mon Nov  9 14:04:50 2009
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Subject: Re: [dispatch] Introducing ViPR: a new federation technology
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Richard Shockey wrote:
> OK Asterisk DUNDi on steroids. 

Sort of, yeah :-)

> Useful for enterprise to enterprise supply chain federations, since TRIP was
> never going to work there, but to a SIP service provider it provides no real
> value.
> 
> Did I miss something?

I came away with the same conclusion you did; it's really only
applicable to closely-aligned parties that want to federate, not for any
 providers that may provide SIP connectivity in between the parties that
want to federate. This would make it somewhat complex to deploy in a
hosted PBX scenario, unless the ViPR server was also virtualized to
provide a separate set of data for each hosted PBX that chose to use ViPR.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

From mary.barnes@nortel.com  Mon Nov  9 14:21:18 2009
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Subject: [dispatch] Identity Adhoc - Nov 9th: Notes available
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Hi all,

I have uploaded Jim McEachern's notes from the Identity Adhoc yesterday:
http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/Identity-AdHoc-N=
otes-jmce.doc

Regards,
Mary.=20

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<P><FONT SIZE=3D2>Hi all,<BR>
<BR>
I have uploaded Jim McEachern's notes from the Identity Adhoc =
yesterday:<BR>
<A =
HREF=3D"http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/Identity=
-AdHoc-Notes-jmce.doc">http://www.softarmor.com/dispatch/IETF76/Meeting%2=
0notes/Identity-AdHoc-Notes-jmce.doc</A><BR>
<BR>
Regards,<BR>
Mary. </FONT>
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From R.Jesske@telekom.de  Mon Nov  9 15:57:24 2009
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Thread-Topic: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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References: <9886E5FCA6D76549A3011068483A4BD40498CFB8@S4DE8PSAAQB.mitte.t-com.de>	<9886E5FCA6D76549A3011068483A4BD40498D2CA@S4DE8PSAAQB.mitte.t-com.de>	<1247255492.3757.40.camel@victoria-pingtel-com.us.nortel.com>	<9886E5FCA6D76549A3011068483A4BD404A14E83@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F050471@zrc2hxm0.corp.nortel.com>	<1247764118.4085.24.camel@victoria-pingtel-com.us.nortel.com><1ECE0EB50388174790F9694F77522CCF1F05050C@zrc2hxm0.corp.nortel.com><4A643B95.3060800@ericsson.com><9886E5FCA6D76549A3011068483A4BD404A9C2B7@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F155AC5@zrc2hxm0.corp.nortel.com>	<CA9998CD4A020D418654FCDEF4E707DF0B1683CC@esealmw113.eemea.ericsson.se>	<1ECE0EB50388174790F9694F77522CCF1F556A65@zrc2hxm0.corp.nortel.com>	<9886E5FCA6D76549A3011068483A4BD404BFFC37@S4DE8PSAAQB.mitte.t-com.de>	<4AF37113.8030908@nostrum.com> <9886E5FCA6D76549A3011068483A4BD405319E68@S4DE8PSAAQB.mitte.t-com.de> <4AF7934B.7080902@cisco.com>
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To: <pkyzivat@cisco.com>
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Hello Paul,
Thanks for that hint. I will change this too.=20
Like:

 The appearance of the Reason header is applicable to final responses
       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as =
defined within
       ietf-sipcore-199.txt. Also the Reason header is
       applicable to provisional responses18x.

Best Regards

Roland

-----Urspr=FCngliche Nachricht-----
Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Gesendet: Montag, 9. November 2009 12:58
An: Jesske, Roland
Cc: adam@nostrum.com; audet@nortel.com; dispatch@ietf.org; =
gonzalo.camarillo@ericsson.com
Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04



R.Jesske@telekom.de wrote:
> Hi Adam,
> Thank you for your mail. So I propose the following text then:
>=20
> The appearance of the Reason header is applicable to final responses
>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as =
defined within
>       ietf-sipcore-199.txt. Also the Reason header is
>       applicable to provisional responses18x which are generated by an
>       interworking gateway from ISUP to SIP.</t>

I think this is ok *except* if you are going to allow one kind of UAS to =

use Reason in a 18x response, then it should be allowed for *any* UAS to =

do so.

For instance, suppose the gateway is replaced by a B2BUA to another sip=20
environment, and the request eventually arrives as a gateway. Then a 18x =

response with a Reason header may arrive at the B2BUA and need to be=20
replicated on the other side.

	Thanks,
	Paul

> Best Beagards
>=20
> Roland=20
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: Adam Roach [mailto:adam@nostrum.com]=20
> Gesendet: Freitag, 6. November 2009 01:43
> An: Jesske, Roland
> Cc: audet@nortel.com; christer.holmberg@ericsson.com; =
gonzalo.camarillo@ericsson.com; dispatch@ietf.org
> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
>> Hi Christer and Francois,
>> I have added a sentence under section Overall Applicability:
>>
>>
>> The appearance of the Reason header is applicable to final responses =
4xx, 5xx and 6xx and in addition for 199 Responses.
>>
>> Is this proper enough? Or do you have more in mind?
>>   =20
>=20
> Given that RFC 3398 specifies the generation of a 301 in response to a =

> cause code of 22 under certain circumstances, it seems pretty=20
> appropriate to allow the inclusion of a Q.850 cause code in 300-class=20
> responses also. Or, at least, as appropriate as it would be to include =

> them in 4xx, 5xx, and 6xx responses.
>=20
> /a
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From pkyzivat@cisco.com  Mon Nov  9 16:05:59 2009
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Roland,

Thanks! That addresses my concern.

	Thanks,
	Paul

R.Jesske@telekom.de wrote:
> Hello Paul,
> Thanks for that hint. I will change this too. 
> Like:
> 
>  The appearance of the Reason header is applicable to final responses
>        3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as defined within
>        ietf-sipcore-199.txt. Also the Reason header is
>        applicable to provisional responses18x.
> 
> Best Regards
> 
> Roland
> 
> -----Ursprüngliche Nachricht-----
> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Gesendet: Montag, 9. November 2009 12:58
> An: Jesske, Roland
> Cc: adam@nostrum.com; audet@nortel.com; dispatch@ietf.org; gonzalo.camarillo@ericsson.com
> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> 
> 
> 
> R.Jesske@telekom.de wrote:
>> Hi Adam,
>> Thank you for your mail. So I propose the following text then:
>>
>> The appearance of the Reason header is applicable to final responses
>>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as defined within
>>       ietf-sipcore-199.txt. Also the Reason header is
>>       applicable to provisional responses18x which are generated by an
>>       interworking gateway from ISUP to SIP.</t>
> 
> I think this is ok *except* if you are going to allow one kind of UAS to 
> use Reason in a 18x response, then it should be allowed for *any* UAS to 
> do so.
> 
> For instance, suppose the gateway is replaced by a B2BUA to another sip 
> environment, and the request eventually arrives as a gateway. Then a 18x 
> response with a Reason header may arrive at the B2BUA and need to be 
> replicated on the other side.
> 
> 	Thanks,
> 	Paul
> 
>> Best Beagards
>>
>> Roland 
>>
>> -----Ursprüngliche Nachricht-----
>> Von: Adam Roach [mailto:adam@nostrum.com] 
>> Gesendet: Freitag, 6. November 2009 01:43
>> An: Jesske, Roland
>> Cc: audet@nortel.com; christer.holmberg@ericsson.com; gonzalo.camarillo@ericsson.com; dispatch@ietf.org
>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
>>> Hi Christer and Francois,
>>> I have added a sentence under section Overall Applicability:
>>>
>>>
>>> The appearance of the Reason header is applicable to final responses 4xx, 5xx and 6xx and in addition for 199 Responses.
>>>
>>> Is this proper enough? Or do you have more in mind?
>>>    
>> Given that RFC 3398 specifies the generation of a 301 in response to a 
>> cause code of 22 under certain circumstances, it seems pretty 
>> appropriate to allow the inclusion of a Q.850 cause code in 300-class 
>> responses also. Or, at least, as appropriate as it would be to include 
>> them in 4xx, 5xx, and 6xx responses.
>>
>> /a
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> 

From christer.holmberg@ericsson.com  Mon Nov  9 16:11:34 2009
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Thread-Topic: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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References: <9886E5FCA6D76549A3011068483A4BD40498CFB8@S4DE8PSAAQB.mitte.t-com.de>	<1247255492.3757.40.camel@victoria-pingtel-com.us.nortel.com>	<9886E5FCA6D76549A3011068483A4BD404A14E83@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F050471@zrc2hxm0.corp.nortel.com>	<1247764118.4085.24.camel@victoria-pingtel-com.us.nortel.com><1ECE0EB50388174790F9694F77522CCF1F05050C@zrc2hxm0.corp.nortel.com><4A643B95.3060800@ericsson.com><9886E5FCA6D76549A3011068483A4BD404A9C2B7@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F155AC5@zrc2hxm0.corp.nortel.com>	<CA9998CD4A020D418654FCDEF4E707DF0B1683CC@esealmw113.eemea.ericsson.se>	<1ECE0EB50388174790F9694F77522CCF1F556A65@zrc2hxm0.corp.nortel.com>	<9886E5FCA6D76549A3011068483A4BD404BFFC37@S4DE8PSAAQB.mitte.t-com.de>	<4AF37113.8030908@nostrum.com><9886E5FCA6D76549A3011068483A4BD405319E68@S4DE8PSAAQB.mitte.t-com.de><4AF7934B.7080902@cisco.com><9886E5FCA6D76549A3011068483A4BD40537238D@S4DE8PSAAQB.mitte.t-com.de> <4AF 8AE73.40 50405@cisco.com>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, <R.Jesske@telekom.de>
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Cc: audet@nortel.com, dispatch@ietf.org, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Hi,

A small editorial change proposal:

"The appearance of the Reason header is applicable to 3xx, 4xx, 5xx and =
6xx final responses
and 18x and 199 <reference> provisional responsess."

Regards,

Christer



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Paul Kyzivat
Sent: 10. marraskuuta 2009 3:06
To: R.Jesske@telekom.de
Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

Roland,

Thanks! That addresses my concern.

	Thanks,
	Paul

R.Jesske@telekom.de wrote:
> Hello Paul,
> Thanks for that hint. I will change this too.=20
> Like:
>=20
>  The appearance of the Reason header is applicable to final responses
>        3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as =
defined within
>        ietf-sipcore-199.txt. Also the Reason header is
>        applicable to provisional responses18x.
>=20
> Best Regards
>=20
> Roland
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Gesendet: Montag, 9. November 2009 12:58
> An: Jesske, Roland
> Cc: adam@nostrum.com; audet@nortel.com; dispatch@ietf.org;=20
> gonzalo.camarillo@ericsson.com
> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
>=20
>=20
> R.Jesske@telekom.de wrote:
>> Hi Adam,
>> Thank you for your mail. So I propose the following text then:
>>
>> The appearance of the Reason header is applicable to final responses
>>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as =
defined within
>>       ietf-sipcore-199.txt. Also the Reason header is
>>       applicable to provisional responses18x which are generated by =
an
>>       interworking gateway from ISUP to SIP.</t>
>=20
> I think this is ok *except* if you are going to allow one kind of UAS=20
> to use Reason in a 18x response, then it should be allowed for *any*=20
> UAS to do so.
>=20
> For instance, suppose the gateway is replaced by a B2BUA to another=20
> sip environment, and the request eventually arrives as a gateway. Then =

> a 18x response with a Reason header may arrive at the B2BUA and need=20
> to be replicated on the other side.
>=20
> 	Thanks,
> 	Paul
>=20
>> Best Beagards
>>
>> Roland
>>
>> -----Urspr=FCngliche Nachricht-----
>> Von: Adam Roach [mailto:adam@nostrum.com]
>> Gesendet: Freitag, 6. November 2009 01:43
>> An: Jesske, Roland
>> Cc: audet@nortel.com; christer.holmberg@ericsson.com;=20
>> gonzalo.camarillo@ericsson.com; dispatch@ietf.org
>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
>>> Hi Christer and Francois,
>>> I have added a sentence under section Overall Applicability:
>>>
>>>
>>> The appearance of the Reason header is applicable to final responses =
4xx, 5xx and 6xx and in addition for 199 Responses.
>>>
>>> Is this proper enough? Or do you have more in mind?
>>>   =20
>> Given that RFC 3398 specifies the generation of a 301 in response to=20
>> a cause code of 22 under certain circumstances, it seems pretty=20
>> appropriate to allow the inclusion of a Q.850 cause code in 300-class =

>> responses also. Or, at least, as appropriate as it would be to=20
>> include them in 4xx, 5xx, and 6xx responses.
>>
>> /a
>> _______________________________________________
>> 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 andrew.hutton@siemens-enterprise.com  Mon Nov  9 17:42:40 2009
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Scott Lawrence <scott.lawrence@nortel.com>, "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
Date: Tue, 10 Nov 2009 02:42:51 +0100
Thread-Topic: [dispatch] Charter proposal: Session Recording Protocol
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Subject: Re: [dispatch] Charter proposal: Session Recording Protocol
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Hi Scott,

I agree that the architecture you described with no middle boxes should be =
considered and included in the architecture draft.

I think the result of today's Dispatch meeting is that a new WG will be for=
med which I think is probably the best way forward because session recordin=
g is much more than a call control feature which is what I believe BLISS is=
 chartered for.

Regards
Andy



>-----Original Message-----
>From: dispatch-bounces@ietf.org
>[mailto:dispatch-bounces@ietf.org] On Behalf Of Scott Lawrence
>Sent: 09 November 2009 15:01
>To: Vijay K. Gurbani
>Cc: dispatch@ietf.org; rai-ads
>Subject: Re: [dispatch] Charter proposal: Session Recording Protocol
>
>On Mon, 2009-09-21 at 08:33 -0500, Vijay K. Gurbani 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.
>
>I'd like to register my support for this work.  I think that the draft
>does a good job laying out the requirements, and that interoperable
>solutions to these problems would be of benefit to everyone.
>
>I would like to have architectures considered that don't rely on a
>middlebox of any kind for this, with dataflows like:
>
>
> +-------------+                   SIP
>+-------------+
> |             |<------------------------------------------->|
>            |
> |             |                                             |
>            |
> |     UA-B    |<---------+                          +------>|
>    UA-C    |
> |             |          | RTP                  RTP |       |
>            |
> +-------------+          |    +--------------+      |
>+-------------+
>    |                     |    |   Recorder   |      |
>    | Recording Control   +--->|     (RS)     |<-----+
>    | Call Metadata events     |              |
>    +------------------------->+--------------+
>           (SIP)
>
>I won't be able to participate even remotely during the
>DISPATCH session
>on this, but would like to point out that in many ways this fits the
>description of BLISS work items: it is fundamentally a calling feature
>dealing with switching call legs appropriately among participants.
>Whether to establish a dedicated working group or not is up to the ADs;
>whether they decide to send it to BLISS or to charter a new group, I
>look forward to participating.
>
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch
>

From adam@nostrum.com  Mon Nov  9 18:03:44 2009
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Cc: dispatch@ietf.org, audet@nortel.com, R.Jesske@telekom.de, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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To clarify the point I was trying to make in the meeting today -- if 
we've decided to go down this path, I'll save you the trouble of 
creating the new ABNF for the new INVITE header fields. Here's what they 
look like:

Nature-Of-Connection-Indicators          = "Nature-Of-Connection-Indicators"
                                            HCOLON quoted-string

Forward-Call-Indicators                  = "Forward-Call-Indicators"
                                            HCOLON quoted-string

Calling-Party's-Category                 = "Calling-Party's-Category"
                                            HCOLON quoted-string

Transmission-Medium-Requirement          = "Transmission-Medium-Requirement"
                                            HCOLON quoted-string

Called-Party-Number                      = "Called-Party-Number"
                                            HCOLON quoted-string

Transit-Network-Selection                = "Transit-Network-Selection"
                                            HCOLON quoted-string

Call-Reference                           = "Call-Reference"
                                            HCOLON quoted-string

Calling-Party-Number                     = "Calling-Party-Number"
                                            HCOLON quoted-string

Optional-Forward-Call-Indicators         = 
"Optional-Forward-Call-Indicators"
                                            HCOLON quoted-string

Redirecting-Number                       = "Redirecting-Number"
                                            HCOLON quoted-string

Redirection-Information                  = "Redirection-Information"
                                            HCOLON quoted-string

Closed-User-Group-Interlock-Code         = 
"Closed-User-Group-Interlock-Code"
                                            HCOLON quoted-string

Connection-Request                       = "Connection-Request"
                                            HCOLON quoted-string

Original-Called-Number                   = "Original-Called-Number"
                                            HCOLON quoted-string

User-to-user-Information                 = "User-to-user-Information"
                                            HCOLON quoted-string

Access-Transport                         = "Access-Transport"
                                            HCOLON quoted-string

User-Service-Information                 = "User-Service-Information"
                                            HCOLON quoted-string

User-to-user-Indicators                  = "User-to-user-Indicators"
                                            HCOLON quoted-string

Generic-Number                           = "Generic-Number"
                                            HCOLON quoted-string

Propagation-Delay-Counter                = "Propagation-Delay-Counter"
                                            HCOLON quoted-string

User-Service-Information-Prime           = "User-Service-Information-Prime"
                                            HCOLON quoted-string

Network-Specific-Facility                = "Network-Specific-Facility"
                                            HCOLON quoted-string

Generic-Digits                           = "Generic-Digits"
                                            HCOLON quoted-string

Origination-ISC-Point-Code               = "Origination-ISC-Point-Code"
                                            HCOLON quoted-string

User-Teleservice-Information             = "User-Teleservice-Information"
                                            HCOLON quoted-string

Remote-Operations                        = "Remote-Operations"
                                            HCOLON quoted-string

Parameter-Compatibility-Information      = 
"Parameter-Compatibility-Information"
                                            HCOLON quoted-string

Generic-Notification-Indicator           = "Generic-Notification-Indicator"
                                            HCOLON quoted-string

Service-Activation                       = "Service-Activation"
                                            HCOLON quoted-string

Generic-Reference                        = "Generic-Reference"
                                            HCOLON quoted-string

MLPP-Precedence                          = "MLPP-Precedence"
                                            HCOLON quoted-string

Transmission-Medium-Requirement-Prime    = 
"Transmission-Medium-Requirement-Prime"
                                            HCOLON quoted-string

Location-Number                          = "Location-Number"
                                            HCOLON quoted-string

Forward-GVNS                             = "Forward-GVNS"
                                            HCOLON quoted-string

CCSS                                     = "CCSS"
                                            HCOLON quoted-string

Network-Management-Controls              = "Network-Management-Controls"
                                            HCOLON quoted-string

Circuit-Assignment-Map                   = "Circuit-Assignment-Map"
                                            HCOLON quoted-string

Correlation-Id                           = "Correlation-Id"
                                            HCOLON quoted-string

Call-Diversion-Treatment-Indicators      = 
"Call-Diversion-Treatment-Indicators"
                                            HCOLON quoted-string

Called-IN-Number                         = "Called-IN-Number"
                                            HCOLON quoted-string

Call-Offering-Treatment-Indicators       = 
"Call-Offering-Treatment-Indicators"
                                            HCOLON quoted-string

Conference-Treatment-Indicators          = "Conference-Treatment-Indicators"
                                            HCOLON quoted-string

SCF-Id                                   = "SCF-Id"
                                            HCOLON quoted-string

UID-Capability-Indicators                = "UID-Capability-Indicators"
                                            HCOLON quoted-string

Echo-Control-Information                 = "Echo-Control-Information"
                                            HCOLON quoted-string

Hop-Counter                              = "Hop-Counter"
                                            HCOLON quoted-string

Collect-Call-Request                     = "Collect-Call-Request"
                                            HCOLON quoted-string

Application-Transport-Parameter          = "Application-Transport-Parameter"
                                            HCOLON quoted-string

Pivot-Capability                         = "Pivot-Capability"
                                            HCOLON quoted-string

Called-Directory-Number                  = "Called-Directory-Number"
                                            HCOLON quoted-string

Original-Called-IN-Number                = "Original-Called-IN-Number"
                                            HCOLON quoted-string

Calling-Geodetic-Location                = "Calling-Geodetic-Location"
                                            HCOLON quoted-string

Network-Routing-Number                   = "Network-Routing-Number"
                                            HCOLON quoted-string

QoR-Capability                           = "QoR-Capability"
                                            HCOLON quoted-string

Pivot-Counter                            = "Pivot-Counter"
                                            HCOLON quoted-string

Pivot-Routing-Forward-Information        = 
"Pivot-Routing-Forward-Information"
                                            HCOLON quoted-string

Redirect-Capability                      = "Redirect-Capability"
                                            HCOLON quoted-string

Redirect-Counter                         = "Redirect-Counter"
                                            HCOLON quoted-string

Redirect-Status                          = "Redirect-Status"
                                            HCOLON quoted-string

Redirect-Forward-Information             = "Redirect-Forward-Information"
                                            HCOLON quoted-string

Number-Portability-Forward-Information   =
                                        
"Number-Portability-Forward-Information"
                                            HCOLON quoted-string

Perhaps we can refine some the the quoted strings with tokens, but I 
think this is a good start.

/a

From Simo.Veikkolainen@nokia.com  Mon Nov  9 18:14:04 2009
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From: <Simo.Veikkolainen@nokia.com>
To: <dispatch@ietf.org>
Date: Tue, 10 Nov 2009 03:14:14 +0100
Thread-Topic: Agenda for SIP/XMPP co-existence ad-hoc meeting on Friday Nov 13
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Subject: [dispatch] Agenda for SIP/XMPP co-existence ad-hoc meeting on Friday Nov 13
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Hello,

Here is the proposed agenda for the DISPATCH WG ad-hoc sessionon SIP/XMPP c=
o-existence scheduled for Friday Nov 13, 2009, at 13:00-15:00 at room Cattl=
eya West. The ad-hoc session will be chaired by Gonzalo Camarillo.

1. Motivation and objectives of SIP/XMPP co-existence (10 min)

2. Overview of solution space (15 min)
- baseline proposal http://tools.ietf.org/html/draft-veikkolainen-sip-voip-=
xmpp-im-01

3. Technical issues discussed so far (15 min)
- http://www.ietf.org/mail-archive/web/dispatch/current/msg00845.html

4. Discussion on charter proposal (30 min)
- http://www.ietf.org/mail-archive/web/dispatch/current/msg00560.html

5. Conclusion and next steps=20

Regards,
Simo

From bernard_aboba@hotmail.com  Mon Nov  9 19:30:48 2009
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From: "Bernard Aboba" <bernard_aboba@hotmail.com>
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Subject: Re: [dispatch] Introducing ViPR: a new federation technology
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Kevin Fleming said:

" I came away with the same conclusion you did; it's really only applicable
to closely-aligned parties that want to federate, not for any  providers
that may provide SIP connectivity in between the parties that want to
federate. This would make it somewhat complex to deploy in a hosted PBX
scenario, unless the ViPR server was also virtualized to provide a separate
set of data for each hosted PBX that chose to use ViPR."

I'm not sure that it's only applicable to closely-aligned parties.  As
noted, the DHT approach does scale logarithmically.  Although one might
quibble with some of the dependencies (e.g. SRP/TLS is encumbered by
multiple IPR declarations, as is RELOAD), the overall approach does not
require pre-existing trust between the parties. 

There are existing approaches for moderate numbers of closely-aligned
parties (e.g. DUNDI works at that scale, and even the existing RFC 3261
model isn't that broken in for that case, since I'm aware of customers who
are federating with dozens of suppliers/customers using TLS with mutual
authentication). 

The bigger issue as I see it is the tying of SIP federation in non-VOIP
scenarios (e.g. Video, Web conferencing, IM&P) to PSTN and phone number
identifiers.  In those scenarios, the proposed introduction and verification
mechanisms aren't much better than existing "buddy list" techniques for
trust establishment.  Those techniques have been in use for a long time
(e.g. in XMPP federation) and have proven quite effective.  

Also, if one buys the arguments, then one could conclude that all that is
necessary for VOIP bypass is a PRI and an Internet connection.  Why bother
with SIP trunking? However, that argument assumes that as the decline in the
PSTN accelerates that SIP Trunking providers will utilize infrastructure
ENUM to their advantage to route calls directly over the Internet without
passing on the savings to customers. 

However, the evolution of Internet peering would indicate this not the most
likely outcome.  Back in the mid-1990s, peering was a significant issue for
second Tier ISPs and access cost much more than it does today.  Today recent
measurements show that 50 percent of Internet traffic is originating in only
150 ASes, and that kind of consolidation also tends to go along with major
declines in consumer costs and an lessening of importance of peering
concerns.  

So my own guess is that as SIP trunking becomes a commodity that the
industry will grow more and more competitive and that in a decade
infrastructure ENUM will apply to a very large fraction of all calls, with
the benefits being passed on to customers. 

In that kind of scenario, ViPR would be unnecessary. 


From hsinnrei@adobe.com  Mon Nov  9 20:52:34 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Mon, 9 Nov 2009 20:52:35 -0800
Thread-Topic: [dispatch] Introducing ViPR: a new federation technology
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--_000_C71F20A310EC7hsinnreiadobecom_
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>In that kind of scenario, ViPR would be unnecessary.

100% agree for the reasons given below and several other that already well =
known and need not be repeated here.
I also don't remember reading anywhere in the Internet architecture about f=
ederations and cartels.

My two personal cents,

Henry


On 11/10/09 12:17 PM, "Bernard Aboba" <bernard_aboba@hotmail.com> wrote:

Kevin Fleming said:

" I came away with the same conclusion you did; it's really only applicable
to closely-aligned parties that want to federate, not for any  providers
that may provide SIP connectivity in between the parties that want to
federate. This would make it somewhat complex to deploy in a hosted PBX
scenario, unless the ViPR server was also virtualized to provide a separate
set of data for each hosted PBX that chose to use ViPR."

I'm not sure that it's only applicable to closely-aligned parties.  As
noted, the DHT approach does scale logarithmically.  Although one might
quibble with some of the dependencies (e.g. SRP/TLS is encumbered by
multiple IPR declarations, as is RELOAD), the overall approach does not
require pre-existing trust between the parties.

There are existing approaches for moderate numbers of closely-aligned
parties (e.g. DUNDI works at that scale, and even the existing RFC 3261
model isn't that broken in for that case, since I'm aware of customers who
are federating with dozens of suppliers/customers using TLS with mutual
authentication).

The bigger issue as I see it is the tying of SIP federation in non-VOIP
scenarios (e.g. Video, Web conferencing, IM&P) to PSTN and phone number
identifiers.  In those scenarios, the proposed introduction and verificatio=
n
mechanisms aren't much better than existing "buddy list" techniques for
trust establishment.  Those techniques have been in use for a long time
(e.g. in XMPP federation) and have proven quite effective.

Also, if one buys the arguments, then one could conclude that all that is
necessary for VOIP bypass is a PRI and an Internet connection.  Why bother
with SIP trunking? However, that argument assumes that as the decline in th=
e
PSTN accelerates that SIP Trunking providers will utilize infrastructure
ENUM to their advantage to route calls directly over the Internet without
passing on the savings to customers.

However, the evolution of Internet peering would indicate this not the most
likely outcome.  Back in the mid-1990s, peering was a significant issue for
second Tier ISPs and access cost much more than it does today.  Today recen=
t
measurements show that 50 percent of Internet traffic is originating in onl=
y
150 ASes, and that kind of consolidation also tends to go along with major
declines in consumer costs and an lessening of importance of peering
concerns.

So my own guess is that as SIP trunking becomes a commodity that the
industry will grow more and more competitive and that in a decade
infrastructure ENUM will apply to a very large fraction of all calls, with
the benefits being passed on to customers.

In that kind of scenario, ViPR would be unnecessary.

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


--_000_C71F20A310EC7hsinnreiadobecom_
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<HTML>
<HEAD>
<TITLE>Re: [dispatch] Introducing ViPR: a new federation technology</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
13pt'>&gt;In that kind of scenario, ViPR would be unnecessary.<BR>
<BR>
100% agree for the reasons given below and several other that already well =
known and need not be repeated here.<BR>
I also don&#8217;t remember reading anywhere in the Internet architecture a=
bout federations and cartels.<BR>
<BR>
My two personal cents,<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 11/10/09 12:17 PM, &quot;Bernard Aboba&quot; &lt;<a href=3D"bernard_abob=
a@hotmail.com">bernard_aboba@hotmail.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:13pt'>Kevin Fleming said:<BR>
<BR>
&quot; I came away with the same conclusion you did; it's really only appli=
cable<BR>
to closely-aligned parties that want to federate, not for any &nbsp;provide=
rs<BR>
that may provide SIP connectivity in between the parties that want to<BR>
federate. This would make it somewhat complex to deploy in a hosted PBX<BR>
scenario, unless the ViPR server was also virtualized to provide a separate=
<BR>
set of data for each hosted PBX that chose to use ViPR.&quot;<BR>
<BR>
I'm not sure that it's only applicable to closely-aligned parties. &nbsp;As=
<BR>
noted, the DHT approach does scale logarithmically. &nbsp;Although one migh=
t<BR>
quibble with some of the dependencies (e.g. SRP/TLS is encumbered by<BR>
multiple IPR declarations, as is RELOAD), the overall approach does not<BR>
require pre-existing trust between the parties.<BR>
<BR>
There are existing approaches for moderate numbers of closely-aligned<BR>
parties (e.g. DUNDI works at that scale, and even the existing RFC 3261<BR>
model isn't that broken in for that case, since I'm aware of customers who<=
BR>
are federating with dozens of suppliers/customers using TLS with mutual<BR>
authentication).<BR>
<BR>
The bigger issue as I see it is the tying of SIP federation in non-VOIP<BR>
scenarios (e.g. Video, Web conferencing, IM&amp;P) to PSTN and phone number=
<BR>
identifiers. &nbsp;In those scenarios, the proposed introduction and verifi=
cation<BR>
mechanisms aren't much better than existing &quot;buddy list&quot; techniqu=
es for<BR>
trust establishment. &nbsp;Those techniques have been in use for a long tim=
e<BR>
(e.g. in XMPP federation) and have proven quite effective. <BR>
<BR>
Also, if one buys the arguments, then one could conclude that all that is<B=
R>
necessary for VOIP bypass is a PRI and an Internet connection. &nbsp;Why bo=
ther<BR>
with SIP trunking? However, that argument assumes that as the decline in th=
e<BR>
PSTN accelerates that SIP Trunking providers will utilize infrastructure<BR=
>
ENUM to their advantage to route calls directly over the Internet without<B=
R>
passing on the savings to customers.<BR>
<BR>
However, the evolution of Internet peering would indicate this not the most=
<BR>
likely outcome. &nbsp;Back in the mid-1990s, peering was a significant issu=
e for<BR>
second Tier ISPs and access cost much more than it does today. &nbsp;Today =
recent<BR>
measurements show that 50 percent of Internet traffic is originating in onl=
y<BR>
150 ASes, and that kind of consolidation also tends to go along with major<=
BR>
declines in consumer costs and an lessening of importance of peering<BR>
concerns. <BR>
<BR>
So my own guess is that as SIP trunking becomes a commodity that the<BR>
industry will grow more and more competitive and that in a decade<BR>
infrastructure ENUM will apply to a very large fraction of all calls, with<=
BR>
the benefits being passed on to customers.<BR>
<BR>
In that kind of scenario, ViPR would be unnecessary.<BR>
<BR>
_______________________________________________<BR>
dispatch mailing 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>
</BODY>
</HTML>


--_000_C71F20A310EC7hsinnreiadobecom_--

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From: <BeckW@telekom.de>
To: <adam@nostrum.com>, <christer.holmberg@ericsson.com>
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Cc: audet@nortel.com, dispatch@ietf.org, gonzalo.camarillo@ericsson.com, R.Jesske@telekom.de
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04; why encapsulation is problematic
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=20
These are my issues with SIP-T:

1) ISUP messages end up at end users. This violates regulatory =
requirements in countries where ISUP must only be sent to licensed =
carriers. For example, called parties will see the calling party =
numbers, even if the calling party required privacy. Apart from the MIME =
issues, this prevents us from using encapsulation.=20

2) Encrypting SIP-T message bodies would solve 1), but looking at SIPit =
results, S/MIME for SIP is a dead technology.

3) Will application servers acting as B2BUAs deal properly with SIP-T =
multipart MIME bodies? People seem to be worried that they are not even =
able to forward a Reason header..

4) Many client don't implement Multipart MIME, as it can get quite =
complex (what to do with multiple SDP bodies, alternative SDP bodies =
etc). I'd have to ask my vendors to add complexity to the software =
without real benefit ("You'll have to implement multipart MIME, so we =
can send binary blobs to you the device has to throw away"). In the year =
2009 some SIP clients don't even get simple SDP offer/answer right, how =
could we expect them to handle Multipart MIME in a sensible manner?=20

5) User agents might have to switch TCP, just to receive oversized SIP =
messages with SIP-T bodies that will by thrown away.=20



Wolfgang




Deutsche Telekom Netzproduktion GmbH=20
Zentrum Technik Einf=FChrung=20
Wolfgang Beck=20
TE 122-14=20
Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt=20
+49 6151 6282832 (Tel.)=20
E-Mail: beckw@telekom.de=20
http://www.telekom.com=20

Erleben, was verbindet.

Deutsche Telekom Netzproduktion GmbH=20
Aufsichtsrat: Timotheus H=F6ttges (Vorsitzender)=20
Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert =
Matheis, Klaus Peren=20
Handelsregister: Amtsgericht Bonn HRB 14190=20
Sitz der Gesellschaft: Bonn=20
USt-IdNr.: DE 814645262



-----Urspr=FCngliche Nachricht-----
Von: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] Im =
Auftrag von Adam Roach
Gesendet: Dienstag, 10. November 2009 03:04
An: Christer Holmberg
Cc: dispatch@ietf.org; audet@nortel.com; Jesske, Roland; Gonzalo =
Camarillo
Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

To clarify the point I was trying to make in the meeting today -- if=20
we've decided to go down this path, I'll save you the trouble of=20
creating the new ABNF for the new INVITE header fields. Here's what they =

look like:

Nature-Of-Connection-Indicators          =3D =
"Nature-Of-Connection-Indicators"
                                            HCOLON quoted-string

Forward-Call-Indicators                  =3D "Forward-Call-Indicators"
                                            HCOLON quoted-string

Calling-Party's-Category                 =3D "Calling-Party's-Category"
                                            HCOLON quoted-string

Transmission-Medium-Requirement          =3D =
"Transmission-Medium-Requirement"
                                            HCOLON quoted-string

Called-Party-Number                      =3D "Called-Party-Number"
                                            HCOLON quoted-string

Transit-Network-Selection                =3D "Transit-Network-Selection"
                                            HCOLON quoted-string

Call-Reference                           =3D "Call-Reference"
                                            HCOLON quoted-string

Calling-Party-Number                     =3D "Calling-Party-Number"
                                            HCOLON quoted-string

Optional-Forward-Call-Indicators         =3D=20
"Optional-Forward-Call-Indicators"
                                            HCOLON quoted-string

Redirecting-Number                       =3D "Redirecting-Number"
                                            HCOLON quoted-string

Redirection-Information                  =3D "Redirection-Information"
                                            HCOLON quoted-string

Closed-User-Group-Interlock-Code         =3D=20
"Closed-User-Group-Interlock-Code"
                                            HCOLON quoted-string

Connection-Request                       =3D "Connection-Request"
                                            HCOLON quoted-string

Original-Called-Number                   =3D "Original-Called-Number"
                                            HCOLON quoted-string

User-to-user-Information                 =3D "User-to-user-Information"
                                            HCOLON quoted-string

Access-Transport                         =3D "Access-Transport"
                                            HCOLON quoted-string

User-Service-Information                 =3D "User-Service-Information"
                                            HCOLON quoted-string

User-to-user-Indicators                  =3D "User-to-user-Indicators"
                                            HCOLON quoted-string

Generic-Number                           =3D "Generic-Number"
                                            HCOLON quoted-string

Propagation-Delay-Counter                =3D "Propagation-Delay-Counter"
                                            HCOLON quoted-string

User-Service-Information-Prime           =3D =
"User-Service-Information-Prime"
                                            HCOLON quoted-string

Network-Specific-Facility                =3D "Network-Specific-Facility"
                                            HCOLON quoted-string

Generic-Digits                           =3D "Generic-Digits"
                                            HCOLON quoted-string

Origination-ISC-Point-Code               =3D =
"Origination-ISC-Point-Code"
                                            HCOLON quoted-string

User-Teleservice-Information             =3D =
"User-Teleservice-Information"
                                            HCOLON quoted-string

Remote-Operations                        =3D "Remote-Operations"
                                            HCOLON quoted-string

Parameter-Compatibility-Information      =3D=20
"Parameter-Compatibility-Information"
                                            HCOLON quoted-string

Generic-Notification-Indicator           =3D =
"Generic-Notification-Indicator"
                                            HCOLON quoted-string

Service-Activation                       =3D "Service-Activation"
                                            HCOLON quoted-string

Generic-Reference                        =3D "Generic-Reference"
                                            HCOLON quoted-string

MLPP-Precedence                          =3D "MLPP-Precedence"
                                            HCOLON quoted-string

Transmission-Medium-Requirement-Prime    =3D=20
"Transmission-Medium-Requirement-Prime"
                                            HCOLON quoted-string

Location-Number                          =3D "Location-Number"
                                            HCOLON quoted-string

Forward-GVNS                             =3D "Forward-GVNS"
                                            HCOLON quoted-string

CCSS                                     =3D "CCSS"
                                            HCOLON quoted-string

Network-Management-Controls              =3D =
"Network-Management-Controls"
                                            HCOLON quoted-string

Circuit-Assignment-Map                   =3D "Circuit-Assignment-Map"
                                            HCOLON quoted-string

Correlation-Id                           =3D "Correlation-Id"
                                            HCOLON quoted-string

Call-Diversion-Treatment-Indicators      =3D=20
"Call-Diversion-Treatment-Indicators"
                                            HCOLON quoted-string

Called-IN-Number                         =3D "Called-IN-Number"
                                            HCOLON quoted-string

Call-Offering-Treatment-Indicators       =3D=20
"Call-Offering-Treatment-Indicators"
                                            HCOLON quoted-string

Conference-Treatment-Indicators          =3D =
"Conference-Treatment-Indicators"
                                            HCOLON quoted-string

SCF-Id                                   =3D "SCF-Id"
                                            HCOLON quoted-string

UID-Capability-Indicators                =3D "UID-Capability-Indicators"
                                            HCOLON quoted-string

Echo-Control-Information                 =3D "Echo-Control-Information"
                                            HCOLON quoted-string

Hop-Counter                              =3D "Hop-Counter"
                                            HCOLON quoted-string

Collect-Call-Request                     =3D "Collect-Call-Request"
                                            HCOLON quoted-string

Application-Transport-Parameter          =3D =
"Application-Transport-Parameter"
                                            HCOLON quoted-string

Pivot-Capability                         =3D "Pivot-Capability"
                                            HCOLON quoted-string

Called-Directory-Number                  =3D "Called-Directory-Number"
                                            HCOLON quoted-string

Original-Called-IN-Number                =3D "Original-Called-IN-Number"
                                            HCOLON quoted-string

Calling-Geodetic-Location                =3D "Calling-Geodetic-Location"
                                            HCOLON quoted-string

Network-Routing-Number                   =3D "Network-Routing-Number"
                                            HCOLON quoted-string

QoR-Capability                           =3D "QoR-Capability"
                                            HCOLON quoted-string

Pivot-Counter                            =3D "Pivot-Counter"
                                            HCOLON quoted-string

Pivot-Routing-Forward-Information        =3D=20
"Pivot-Routing-Forward-Information"
                                            HCOLON quoted-string

Redirect-Capability                      =3D "Redirect-Capability"
                                            HCOLON quoted-string

Redirect-Counter                         =3D "Redirect-Counter"
                                            HCOLON quoted-string

Redirect-Status                          =3D "Redirect-Status"
                                            HCOLON quoted-string

Redirect-Forward-Information             =3D =
"Redirect-Forward-Information"
                                            HCOLON quoted-string

Number-Portability-Forward-Information   =3D
                                       =20
"Number-Portability-Forward-Information"
                                            HCOLON quoted-string

Perhaps we can refine some the the quoted strings with tokens, but I=20
think this is a good start.

/a
_______________________________________________
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From adam@nostrum.com  Mon Nov  9 21:41:37 2009
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Cc: audet@nortel.com, dispatch@ietf.org, gonzalo.camarillo@ericsson.com, R.Jesske@telekom.de
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04; why encapsulation is problematic
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On 11/10/09 2:01 PM, BeckW@telekom.de wrote:
>
> These are my issues with SIP-T:
>
> 1) ISUP messages end up at end users. This violates regulatory requirements in countries where ISUP must only be sent to licensed carriers. For example, called parties will see the calling party numbers, even if the calling party required privacy. Apart from the MIME issues, this prevents us from using encapsulation.
>
> 2) Encrypting SIP-T message bodies would solve 1), but looking at SIPit results, S/MIME for SIP is a dead technology.
>    

S/MIME for a general purpose network suffers from the classical key 
distribution problem. It's not used for the same reason encrypted email 
is not used (except among the geek community).

But you're not talking about general purpose networks.

You're talking about tightly constrained networks with PSTN gateways 
that are either all within a single administrative domain, or that 
communicate with each other through bilateral agreements. In these 
circumstances, key distribution is no more difficult a problem than IP 
address assignment. It's a simple matter of configuration.

> 3) Will application servers acting as B2BUAs deal properly with SIP-T multipart MIME bodies? People seem to be worried that they are not even able to forward a Reason header..
>    

B2BUAs will break whatever they break. Until someone sits down and 
documents their behavior, we can't really do too much to work around 
their behavior. It's the same problem we're having all over the place 
(identity, MSRP ACM, etc). There's no point in talking about something 
as nebulous as B2BUA behavior.

> 4) Many client don't implement Multipart MIME, as it can get quite complex (what to do with multiple SDP bodies, alternative SDP bodies etc). I'd have to ask my vendors to add complexity to the software without real benefit ("You'll have to implement multipart MIME, so we can send binary blobs to you the device has to throw away"). In the year 2009 some SIP clients don't even get simple SDP offer/answer right, how could we expect them to handle Multipart MIME in a sensible manner?

While many UAs won't be able to ignore the multitype body (aside: this 
is sad -- email clients all can), you'd be hard pressed to find one that 
can't understand multipart that also won't respond to a multipart body 
in a request with a 415. This ends up looking just like SIP 
authentication, since the UAC will detect the failure to support 
multipart and re-attempt with a single body:

  UAC           UAS
   |---INVITE--->|
   |<----415-----|
   |---INVITE--->|
   |<----180-----|
         ...
   |<----200-----|
   |-----ACK---->|



> 5) User agents might have to switch TCP, just to receive oversized SIP messages with SIP-T bodies that will by thrown away.
>    

At one point, you said "IMS." I don't think you'll find an IMS client 
that has any hope of receiving a UDP SIP message -- show me an INVITE on 
the terminating side of an IMS network that is somehow smaller than 1500 
bytes, and I'll be shocked into silence. TCP is a foregone conclusion.

/a

From adam@nostrum.com  Mon Nov  9 21:50:45 2009
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Cc: dispatch@ietf.org, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>, R.Jesske@telekom.de
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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It was pointed out to me that this email was somewhat nonsensical to 
people who were not in attendance at the face-to-face meeting.

My point is that increasing the amount of ISUP information that we 
include in SIP message headers is going down a slippery slope, the 
ultimate conclusion of which is including SIP header fields for every 
ISUP parameter.

We have a means for sending ISUP fields transparently through a SIP 
network (RFC 3398). I think it's folly to revisit that set of decisions 
simply because someone wants a less-general point solution.

/a

On 11/10/09 11:04 AM, Adam Roach wrote:
> To clarify the point I was trying to make in the meeting today -- if 
> we've decided to go down this path, I'll save you the trouble of 
> creating the new ABNF for the new INVITE header fields. Here's what 
> they look like:
>
> Nature-Of-Connection-Indicators          = 
> "Nature-Of-Connection-Indicators"
>                                            HCOLON quoted-string
>
> Forward-Call-Indicators                  = "Forward-Call-Indicators"
>                                            HCOLON quoted-string
>
> Calling-Party's-Category                 = "Calling-Party's-Category"
>                                            HCOLON quoted-string
>
> Transmission-Medium-Requirement          = 
> "Transmission-Medium-Requirement"
>                                            HCOLON quoted-string
>
> Called-Party-Number                      = "Called-Party-Number"
>                                            HCOLON quoted-string
>
> Transit-Network-Selection                = "Transit-Network-Selection"
>                                            HCOLON quoted-string
>
> Call-Reference                           = "Call-Reference"
>                                            HCOLON quoted-string
>
> Calling-Party-Number                     = "Calling-Party-Number"
>                                            HCOLON quoted-string
>
> Optional-Forward-Call-Indicators         = 
> "Optional-Forward-Call-Indicators"
>                                            HCOLON quoted-string
>
> Redirecting-Number                       = "Redirecting-Number"
>                                            HCOLON quoted-string
>
> Redirection-Information                  = "Redirection-Information"
>                                            HCOLON quoted-string
>
> Closed-User-Group-Interlock-Code         = 
> "Closed-User-Group-Interlock-Code"
>                                            HCOLON quoted-string
>
> Connection-Request                       = "Connection-Request"
>                                            HCOLON quoted-string
>
> Original-Called-Number                   = "Original-Called-Number"
>                                            HCOLON quoted-string
>
> User-to-user-Information                 = "User-to-user-Information"
>                                            HCOLON quoted-string
>
> Access-Transport                         = "Access-Transport"
>                                            HCOLON quoted-string
>
> User-Service-Information                 = "User-Service-Information"
>                                            HCOLON quoted-string
>
> User-to-user-Indicators                  = "User-to-user-Indicators"
>                                            HCOLON quoted-string
>
> Generic-Number                           = "Generic-Number"
>                                            HCOLON quoted-string
>
> Propagation-Delay-Counter                = "Propagation-Delay-Counter"
>                                            HCOLON quoted-string
>
> User-Service-Information-Prime           = 
> "User-Service-Information-Prime"
>                                            HCOLON quoted-string
>
> Network-Specific-Facility                = "Network-Specific-Facility"
>                                            HCOLON quoted-string
>
> Generic-Digits                           = "Generic-Digits"
>                                            HCOLON quoted-string
>
> Origination-ISC-Point-Code               = "Origination-ISC-Point-Code"
>                                            HCOLON quoted-string
>
> User-Teleservice-Information             = "User-Teleservice-Information"
>                                            HCOLON quoted-string
>
> Remote-Operations                        = "Remote-Operations"
>                                            HCOLON quoted-string
>
> Parameter-Compatibility-Information      = 
> "Parameter-Compatibility-Information"
>                                            HCOLON quoted-string
>
> Generic-Notification-Indicator           = 
> "Generic-Notification-Indicator"
>                                            HCOLON quoted-string
>
> Service-Activation                       = "Service-Activation"
>                                            HCOLON quoted-string
>
> Generic-Reference                        = "Generic-Reference"
>                                            HCOLON quoted-string
>
> MLPP-Precedence                          = "MLPP-Precedence"
>                                            HCOLON quoted-string
>
> Transmission-Medium-Requirement-Prime    = 
> "Transmission-Medium-Requirement-Prime"
>                                            HCOLON quoted-string
>
> Location-Number                          = "Location-Number"
>                                            HCOLON quoted-string
>
> Forward-GVNS                             = "Forward-GVNS"
>                                            HCOLON quoted-string
>
> CCSS                                     = "CCSS"
>                                            HCOLON quoted-string
>
> Network-Management-Controls              = "Network-Management-Controls"
>                                            HCOLON quoted-string
>
> Circuit-Assignment-Map                   = "Circuit-Assignment-Map"
>                                            HCOLON quoted-string
>
> Correlation-Id                           = "Correlation-Id"
>                                            HCOLON quoted-string
>
> Call-Diversion-Treatment-Indicators      = 
> "Call-Diversion-Treatment-Indicators"
>                                            HCOLON quoted-string
>
> Called-IN-Number                         = "Called-IN-Number"
>                                            HCOLON quoted-string
>
> Call-Offering-Treatment-Indicators       = 
> "Call-Offering-Treatment-Indicators"
>                                            HCOLON quoted-string
>
> Conference-Treatment-Indicators          = 
> "Conference-Treatment-Indicators"
>                                            HCOLON quoted-string
>
> SCF-Id                                   = "SCF-Id"
>                                            HCOLON quoted-string
>
> UID-Capability-Indicators                = "UID-Capability-Indicators"
>                                            HCOLON quoted-string
>
> Echo-Control-Information                 = "Echo-Control-Information"
>                                            HCOLON quoted-string
>
> Hop-Counter                              = "Hop-Counter"
>                                            HCOLON quoted-string
>
> Collect-Call-Request                     = "Collect-Call-Request"
>                                            HCOLON quoted-string
>
> Application-Transport-Parameter          = 
> "Application-Transport-Parameter"
>                                            HCOLON quoted-string
>
> Pivot-Capability                         = "Pivot-Capability"
>                                            HCOLON quoted-string
>
> Called-Directory-Number                  = "Called-Directory-Number"
>                                            HCOLON quoted-string
>
> Original-Called-IN-Number                = "Original-Called-IN-Number"
>                                            HCOLON quoted-string
>
> Calling-Geodetic-Location                = "Calling-Geodetic-Location"
>                                            HCOLON quoted-string
>
> Network-Routing-Number                   = "Network-Routing-Number"
>                                            HCOLON quoted-string
>
> QoR-Capability                           = "QoR-Capability"
>                                            HCOLON quoted-string
>
> Pivot-Counter                            = "Pivot-Counter"
>                                            HCOLON quoted-string
>
> Pivot-Routing-Forward-Information        = 
> "Pivot-Routing-Forward-Information"
>                                            HCOLON quoted-string
>
> Redirect-Capability                      = "Redirect-Capability"
>                                            HCOLON quoted-string
>
> Redirect-Counter                         = "Redirect-Counter"
>                                            HCOLON quoted-string
>
> Redirect-Status                          = "Redirect-Status"
>                                            HCOLON quoted-string
>
> Redirect-Forward-Information             = "Redirect-Forward-Information"
>                                            HCOLON quoted-string
>
> Number-Portability-Forward-Information   =
>                                        
> "Number-Portability-Forward-Information"
>                                            HCOLON quoted-string
>
> Perhaps we can refine some the the quoted strings with tokens, but I 
> think this is a good start.
>
> /a
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From jdrosen@jdrosen.net  Mon Nov  9 21:59:04 2009
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Subject: Re: [dispatch] Introducing ViPR: a new federation technology
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inline:

Bernard Aboba wrote:
> Kevin Fleming said:
> 
> " I came away with the same conclusion you did; it's really only applicable
> to closely-aligned parties that want to federate, not for any  providers
> that may provide SIP connectivity in between the parties that want to
> federate. This would make it somewhat complex to deploy in a hosted PBX
> scenario, unless the ViPR server was also virtualized to provide a separate
> set of data for each hosted PBX that chose to use ViPR."
> 
> I'm not sure that it's only applicable to closely-aligned parties.  As
> noted, the DHT approach does scale logarithmically.  Although one might
> quibble with some of the dependencies (e.g. SRP/TLS is encumbered by
> multiple IPR declarations, as is RELOAD), the overall approach does not
> require pre-existing trust between the parties. 

Exactly. The whole idea is that no close alignment is needed; its 
any-to-any in the traditional Internet model.

ViPR applies to any domain that has endpoints that have been assigned 
numbers. That certainly includes enterprises, but is not at all limited 
to that. It has applicability to all kinds of service providers - 
wireless operators (lots of endpoints), application providers (Vonage, 
Skype), cable operators, traditional LEC, CLEC, etc.

> 
> There are existing approaches for moderate numbers of closely-aligned
> parties (e.g. DUNDI works at that scale, and even the existing RFC 3261
> model isn't that broken in for that case, since I'm aware of customers who
> are federating with dozens of suppliers/customers using TLS with mutual
> authentication). 

As the drafts point out, private federation is in use today and it works 
OK at small scales. Stuff like DUNDI can work there, though you run into 
trust issues quickly as numbers are not verified.

> 
> The bigger issue as I see it is the tying of SIP federation in non-VOIP
> scenarios (e.g. Video, Web conferencing, IM&P) to PSTN and phone number
> identifiers.  In those scenarios, the proposed introduction and verification
> mechanisms aren't much better than existing "buddy list" techniques for
> trust establishment.  Those techniques have been in use for a long time
> (e.g. in XMPP federation) and have proven quite effective.  

Firstly, ViPR is equally applicable to video, specifically for cases 
where video endpoints aren't 'special' - they are the same endpoint I 
use for my voip calling. When video is just a feature of my 
pstn-reachable voice device, ViPR works well.

IM&P - which usually doesn't use phone numbers - can be federated with 
existing DNS mechanisms and buddy-list based trust. However, as I 
mention in the draft, the reality of the situation is that *most* folks 
doing voip are still using phone numbers, and that is the problem we're 
trying to address here. Do you not agree that the majority of voice 
connections are still being established to phone numbers today?


> 
> Also, if one buys the arguments, then one could conclude that all that is
> necessary for VOIP bypass is a PRI and an Internet connection.  Why bother
> with SIP trunking?

SIP trunking would still get you cost savings to non-vipr connected 
domains and non-voip installtions - all of the rest of the existing PSTN 
endpoints. That'll continue to be useful for a really long time.

ViPR + SIP trunking then further optimizes (and cost reduces) for the 
domains that you can vipr federate with.

  However, that argument assumes that as the decline in the
> PSTN accelerates that SIP Trunking providers will utilize infrastructure
> ENUM to their advantage to route calls directly over the Internet without
> passing on the savings to customers. 
> 
> However, the evolution of Internet peering would indicate this not the most
> likely outcome.  Back in the mid-1990s, peering was a significant issue for
> second Tier ISPs and access cost much more than it does today.  Today recent
> measurements show that 50 percent of Internet traffic is originating in only
> 150 ASes, and that kind of consolidation also tends to go along with major
> declines in consumer costs and an lessening of importance of peering
> concerns.  
> 
> So my own guess is that as SIP trunking becomes a commodity that the
> industry will grow more and more competitive and that in a decade
> infrastructure ENUM will apply to a very large fraction of all calls, with
> the benefits being passed on to customers. 

I believe there will be infrastructure enum, absolutely. However, I 
believe it will be used to create peering relationships amongst the 
carries which are similar to the ones that exist today - with small 
regional operators federating with larger international ones, which then 
federate again off to more local ones. IP federation follows a similar 
model. These federation agreements are likely in many cases to mirror 
the existing peering relationships on the PSTN. This is because the 
peering agreements are likely to include actual RTP transport to, along 
with the CAC/QoS features that are the IP equivalents of what is done in 
the PSTN today.

That means, in turn, that a call from company A to company B would go as 
follows:

a.com -> SP1 -> SP2 -> SP3 -> b.com

and, along the way, pass through the myriad of SBCs and SIP 
infrastructure in each of these providers:

a.com -> SP1-SBC-ingres -> SP1-SBC-egress -> SP2-SBC-ingress -> 
SP2-SBC-egress -> SP3-SBC-ingress -> SP3-SBC-egress -> b.com

That eliminates the trapezoidal model of SIP, which I still believe is 
essential for feature  transparency, feature velocity, and security, 
amongst many other reasons. The feature velocity one is really 
important. The more that we have intermediaries in the path, the less 
likely (exponentially so, in fact) it is that any kind of new feature 
will ever work. There is a ton of precedent for this.

With ViPR, you get:

a.com -> b.com

And as a consequence, no inderdepdency on anything else for a new 
feature to work. Thats how the Internet was meant to be. Can you imagine 
what the web would be like if my HTTP traffic had to be handled by web 
proxies at each intermediate ISP?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   SkypeID: jdrosen
Chief Technology Strategist                    PHONE: +1 (732) 766-2496
Skype
jdrosen@skype.net                              http://www.skype.com
jdrosen@jdrosen.net                            http://www.jdrosen.net



From bernarda@microsoft.com  Tue Nov 10 00:30:33 2009
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From: Bernard Aboba <bernarda@microsoft.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
Thread-Topic: Re: [dispatch] Introducing ViPR: a new federation technology
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--_000_F839FBA415E97B4DBD5BA437162E0603201E993ATK5EX14MBXW652w_
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Jonathan Rosenberg said:

"Do you not agree that the majority of voice connections are still being es=
tablished to phone numbers today?"

[BA] Sure.  And I expect this will be the case for going forward too.  My q=
uestion was more about the application of ViPR to media *other* than voice.

"Firstly, ViPR is equally applicable to video, specifically for cases where=
 video endpoints aren't 'special' - they are the same endpoint I use for my=
 voip calling. When video is just a feature of my pstn-reachable voice devi=
ce, ViPR works well."

[BA] The question in my mind is whether video really is "special" (in the I=
M&P sense) or not.  That is, today we accept the idea that anyone should be=
 able to complete a phone call to anyone else.  We also accept this for SMS=
.  However, we don't accept that idea for Instant Messaging (which typicall=
y requires being in the buddy list). Do we accept that principle for video?=
  I'm not sure.  There are some inherent disadvantages to accepting video c=
alls from anywhere even if we have a verified phone number on the other end=
.

So even if video is a feature of a voice device, it still might be sufficie=
nt to control the ability to complete a video call via a "buddy list".  Tha=
t's essentially the model for XMPP-based federation with Jingle.  Not good =
enough to provide telephony as we think of it today.  But this does provide=
 a degree of spam control while retaining something more akin to the RFC 32=
61 federation model (though perhaps with some loosening).

"I believe there will be infrastructure enum, absolutely. However, I believ=
e it will be used to create peering relationships amongst the carries which=
 are similar to the ones that exist today - with small regional operators f=
ederating with larger international ones, which then federate again off to =
more local ones. IP federation follows a similar model. These federation ag=
reements are likely in many cases to mirror the existing peering relationsh=
ips on the PSTN. This is because the peering agreements are likely to inclu=
de actual RTP transport to, along with the CAC/QoS features that are the IP=
 equivalents of what is done in the PSTN today.

That means, in turn, that a call from company A to company B would go as fo=
llows:

a.com -> SP1 -> SP2 -> SP3 -> b.com

and, along the way, pass through the myriad of SBCs and SIP infrastructure =
in each of these providers:

a.com -> SP1-SBC-ingres -> SP1-SBC-egress -> SP2-SBC-ingress -> SP2-SBC-egr=
ess -> SP3-SBC-ingress -> SP3-SBC-egress -> b.com

That eliminates the trapezoidal model of SIP, which I still believe is esse=
ntial for feature transparency, feature velocity, and security, amongst man=
y other reasons. The feature velocity one is really important. The more tha=
t we have intermediaries in the path, the less likely (exponentially so, in=
 fact) it is that any kind of new feature will ever work. There is a ton of=
 precedent for this.

With ViPR, you get:

a.com -> b.com

"

[BA] This seems like the crux of the matter.  Even if the costs are equal t=
o the customer, a ViPR call could be more direct.  If all that's being prov=
ided is basic telephony minutes, it might not be very noticeable.  But if t=
here are services involved, you end up with a more evolvable infrastructure=
.

"SIP trunking would still get you cost savings to non-vipr connected domain=
s and non-voip installations - all of the rest of the existing PSTN endpoin=
ts. That'll continue to be useful for a really long time.

ViPR + SIP trunking then further optimizes (and cost reduces) for the domai=
ns that you can vipr federate with."

[BA] Infrastructure ENUM potentially can count in its camp phone numbers fr=
om the major consumer VOIP providers, as well as phone numbers provided by =
major SIP trunking providers (who are also presumably utilizing infrastruct=
ure ENUM).

While that seems like an advantage, if the SIP trunking provider still char=
ges on a per-minute basis, even for calls which may be carried entirely ove=
r the Internet, then there will still be an incentive for a customer to dep=
loy a direct alternative (ViPR, DUNDI, etc.) since that will allow calls ma=
de entirely over the Internet to be made for free.  Although we do see cell=
ular operators throwing in "in-networking calling", this is a far cry from =
throwing in intra-domain or "out of network" calls as well.  This seems a l=
ot less likely.



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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal>Jonathan Rosenberg said:<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>&#8220;<tt><span style=3D'font-size:10.0pt'>Do you not=
 agree
that the majority of voice connections are still being established to phone
numbers today?&#8221;<o:p></o:p></span></tt></p>

<p class=3DMsoNormal><tt><span style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p>=
</span></tt></p>

<p class=3DMsoNormal><tt><span style=3D'font-size:10.0pt'>[BA] Sure.&nbsp; =
And I
expect this will be the case for going forward too.&nbsp; My question was m=
ore
about the application of ViPR to media *<b>other</b>* than voice. </span></=
tt><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>&#8220;<span style=3D'font-size:10.0pt;font-family:"Co=
urier New"'>Firstly,
ViPR is equally applicable to video, specifically for cases where video
endpoints aren't 'special' - they are the same endpoint I use for my voip
calling. When video is just a feature of my pstn-reachable voice device, Vi=
PR
works well.&#8221;<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>[BA]
The question in my mind is whether video really is &#8220;special&#8221; (i=
n
the IM&amp;P sense) or not.&nbsp; That is, today we accept the idea that an=
yone
should be able to complete a phone call to anyone else.&nbsp; We also accep=
t
this for SMS.&nbsp; However, we don&#8217;t accept that idea for Instant
Messaging (which typically requires being in the buddy list). Do we accept =
that
principle for video?&nbsp; I&#8217;m not sure.&nbsp; There are some inheren=
t
disadvantages to accepting video calls from anywhere even if we have a veri=
fied
phone number on the other end. &nbsp;<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>So even
if video is a feature of a voice device, it still might be sufficient to co=
ntrol
the ability to complete a video call via a &#8220;buddy list&#8221;.&nbsp; =
That&#8217;s
essentially the model for XMPP-based federation with Jingle.&nbsp; Not good
enough to provide telephony as we think of it today.&nbsp; But this does pr=
ovide
a degree of spam control while retaining something more akin to the RFC 326=
1
federation model (though perhaps with some loosening). &nbsp;<o:p></o:p></s=
pan></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>&#8220;I
believe there will be infrastructure enum, absolutely. However, I believe i=
t
will be used to create peering relationships amongst the carries which are
similar to the ones that exist today - with small regional operators federa=
ting
with larger international ones, which then federate again off to more local
ones. IP federation follows a similar model. These federation agreements ar=
e
likely in many cases to mirror the existing peering relationships on the PS=
TN.
This is because the peering agreements are likely to include actual RTP
transport to, along with the CAC/QoS features that are the IP equivalents o=
f
what is done in the PSTN today. </span><span style=3D'font-size:12.0pt;
font-family:"Times New Roman","serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>That
means, in turn, that a call from company A to company B would go as follows=
: </span><span
style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'><o:p></o:p=
></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>a.com
-&gt; SP1 -&gt; SP2 -&gt; SP3 -&gt; b.com<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>and,
along the way, pass through the myriad of SBCs and SIP infrastructure in ea=
ch
of these providers: </span><span style=3D'font-size:12.0pt;font-family:"Tim=
es New Roman","serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>a.com
-&gt; SP1-SBC-ingres -&gt; SP1-SBC-egress -&gt; SP2-SBC-ingress -&gt;
SP2-SBC-egress -&gt; SP3-SBC-ingress -&gt; SP3-SBC-egress -&gt; b.com </spa=
n><span
style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'><o:p></o:p=
></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>That
eliminates the trapezoidal model of SIP, which I still believe is essential=
 for
feature transparency, feature velocity, and security, amongst many other re=
asons.
The feature velocity one is really important. The more that we have
intermediaries in the path, the less likely (exponentially so, in fact) it =
is
that any kind of new feature will ever work. There is a ton of precedent fo=
r
this. </span><span style=3D'font-size:12.0pt;font-family:"Times New Roman",=
"serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>With
ViPR, you get:<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>a.com
-&gt; b.com<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>&#8221;<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>[BA]
This seems like the crux of the matter.&nbsp; Even if the costs are equal t=
o
the customer, a ViPR call could be more direct. &nbsp;If all that&#8217;s b=
eing
provided is basic telephony minutes, it might not be very noticeable. &nbsp=
;But
if there are services involved, you end up with a more evolvable
infrastructure.&nbsp; &nbsp;<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>&#8220;SIP
trunking would still get you cost savings to non-vipr connected domains and
non-voip installations - all of the rest of the existing PSTN endpoints.
That'll continue to be useful for a really long time. </span><span
style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'><o:p></o:p=
></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>ViPR
+ SIP trunking then further optimizes (and cost reduces) for the domains th=
at
you can vipr federate with.&#8221;<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>[BA]
Infrastructure ENUM potentially can count in its camp phone numbers from th=
e
major consumer VOIP providers, as well as phone numbers provided by major S=
IP
trunking providers (who are also presumably utilizing infrastructure ENUM).=
&nbsp;
<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>While
that seems like an advantage, if the SIP trunking provider still charges on=
 a
per-minute basis, even for calls which may be carried entirely over the
Internet, then there will still be an incentive for a customer to deploy a =
direct
alternative (ViPR, DUNDI, etc.) since that will allow calls made entirely o=
ver
the Internet to be made for free.&nbsp; Although we do see cellular operato=
rs throwing
in &#8220;in-networking calling&#8221;, this is a far cry from throwing in =
intra-domain
or &#8220;out of network&#8221; calls as well.&nbsp; This seems a lot less
likely.&nbsp; <o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier N=
ew"'>&nbsp;</span><span
style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'><o:p></o:p=
></span></p>

</div>

</body>

</html>

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Jonathan Rosenberg said:

 

"Do you not agree that the majority of voice connections are still being
established to phone numbers today?"

 

[BA] Sure.  And I expect this will be the case for going forward too.  My
question was more about the application of ViPR to media *other* than voice.


 

"Firstly, ViPR is equally applicable to video, specifically for cases where
video endpoints aren't 'special' - they are the same endpoint I use for my
voip calling. When video is just a feature of my pstn-reachable voice
device, ViPR works well."

 

[BA] The question in my mind is whether video really is "special" (in the
IM&P sense) or not.  That is, today we accept the idea that anyone should be
able to complete a phone call to anyone else.  We also accept this for SMS.
However, we don't accept that idea for Instant Messaging (which typically
requires being in the buddy list). Do we accept that principle for video?
I'm not sure.  There are some inherent disadvantages to accepting video
calls from anywhere even if we have a verified phone number on the other
end.  

 

So even if video is a feature of a voice device, it still might be
sufficient to control the ability to complete a video call via a "buddy
list".  That's essentially the model for XMPP-based federation with Jingle.
Not good enough to provide telephony as we think of it today.  But this does
provide a degree of spam control while retaining something more akin to the
RFC 3261 federation model (though perhaps with some loosening).  

 

"I believe there will be infrastructure enum, absolutely. However, I believe
it will be used to create peering relationships amongst the carries which
are similar to the ones that exist today - with small regional operators
federating with larger international ones, which then federate again off to
more local ones. IP federation follows a similar model. These federation
agreements are likely in many cases to mirror the existing peering
relationships on the PSTN. This is because the peering agreements are likely
to include actual RTP transport to, along with the CAC/QoS features that are
the IP equivalents of what is done in the PSTN today. 

 

That means, in turn, that a call from company A to company B would go as
follows: 

 

a.com -> SP1 -> SP2 -> SP3 -> b.com

 

and, along the way, pass through the myriad of SBCs and SIP infrastructure
in each of these providers: 

 

a.com -> SP1-SBC-ingres -> SP1-SBC-egress -> SP2-SBC-ingress ->
SP2-SBC-egress -> SP3-SBC-ingress -> SP3-SBC-egress -> b.com 

 

That eliminates the trapezoidal model of SIP, which I still believe is
essential for feature transparency, feature velocity, and security, amongst
many other reasons. The feature velocity one is really important. The more
that we have intermediaries in the path, the less likely (exponentially so,
in fact) it is that any kind of new feature will ever work. There is a ton
of precedent for this. 

 

With ViPR, you get:

 

a.com -> b.com

 

"

 

[BA] This seems like the crux of the matter.  Even if the costs are equal to
the customer, a ViPR call could be more direct.  If all that's being
provided is basic telephony minutes, it might not be very noticeable.  But
if there are services involved, you end up with a more evolvable
infrastructure.   

 

"SIP trunking would still get you cost savings to non-vipr connected domains
and non-voip installations - all of the rest of the existing PSTN endpoints.
That'll continue to be useful for a really long time. 

 

ViPR + SIP trunking then further optimizes (and cost reduces) for the
domains that you can vipr federate with."

 

[BA] Infrastructure ENUM potentially can count in its camp phone numbers
from the major consumer VOIP providers, as well as phone numbers provided by
major SIP trunking providers (who are also presumably utilizing
infrastructure ENUM).  

 

While that seems like an advantage, if the SIP trunking provider still
charges on a per-minute basis, even for calls which may be carried entirely
over the Internet, then there will still be an incentive for a customer to
deploy a direct alternative (ViPR, DUNDI, etc.) since that will allow calls
made entirely over the Internet to be made for free.  Although we do see
cellular operators throwing in "in-networking calling", this is a far cry
from throwing in intra-domain or "out of network" calls as well.  This seems
a lot less likely.  

 

 


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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal>Jonathan Rosenberg said:<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>&#8220;<tt><span style=3D'font-size:10.0pt'>Do you =
not agree that
the majority of voice connections are still being established to phone =
numbers
today?&#8221;<o:p></o:p></span></tt></p>

<p class=3DMsoNormal><tt><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></tt></p>

<p class=3DMsoNormal><tt><span style=3D'font-size:10.0pt'>[BA] =
Sure.&nbsp; And I
expect this will be the case for going forward too.&nbsp; My question =
was more
about the application of ViPR to media *<b>other</b>* than voice. =
</span></tt><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>&#8220;<span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Firstly,
ViPR is equally applicable to video, specifically for cases where video
endpoints aren't 'special' - they are the same endpoint I use for my =
voip
calling. When video is just a feature of my pstn-reachable voice device, =
ViPR
works well.&#8221;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>[BA]
The question in my mind is whether video really is &#8220;special&#8221; =
(in the IM&amp;P
sense) or not.&nbsp; That is, today we accept the idea that anyone =
should be
able to complete a phone call to anyone else.&nbsp; We also accept this =
for
SMS.&nbsp; However, we don&#8217;t accept that idea for Instant =
Messaging (which
typically requires being in the buddy list). Do we accept that principle =
for
video?&nbsp; I&#8217;m not sure.&nbsp; There are some inherent =
disadvantages to
accepting video calls from anywhere even if we have a verified phone =
number on
the other end. &nbsp;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>So
even if video is a feature of a voice device, it still might be =
sufficient to
control the ability to complete a video call via a &#8220;buddy =
list&#8221;.&nbsp; That&#8217;s
essentially the model for XMPP-based federation with Jingle.&nbsp; Not =
good
enough to provide telephony as we think of it today.&nbsp; But this does
provide a degree of spam control while retaining something more akin to =
the RFC
3261 federation model (though perhaps with some loosening). =
&nbsp;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&#8220;I
believe there will be infrastructure enum, absolutely. However, I =
believe it
will be used to create peering relationships amongst the carries which =
are
similar to the ones that exist today - with small regional operators =
federating
with larger international ones, which then federate again off to more =
local
ones. IP federation follows a similar model. These federation agreements =
are
likely in many cases to mirror the existing peering relationships on the =
PSTN.
This is because the peering agreements are likely to include actual RTP
transport to, along with the CAC/QoS features that are the IP =
equivalents of
what is done in the PSTN today. </span><span style=3D'font-size:12.0pt;
font-family:"Times New Roman","serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>That
means, in turn, that a call from company A to company B would go as =
follows: </span><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>a.com
-&gt; SP1 -&gt; SP2 -&gt; SP3 -&gt; b.com<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>and,
along the way, pass through the myriad of SBCs and SIP infrastructure in =
each
of these providers: </span><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>a.com
-&gt; SP1-SBC-ingres -&gt; SP1-SBC-egress -&gt; SP2-SBC-ingress -&gt;
SP2-SBC-egress -&gt; SP3-SBC-ingress -&gt; SP3-SBC-egress -&gt; b.com =
</span><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>That
eliminates the trapezoidal model of SIP, which I still believe is =
essential for
feature transparency, feature velocity, and security, amongst many other
reasons. The feature velocity one is really important. The more that we =
have
intermediaries in the path, the less likely (exponentially so, in fact) =
it is
that any kind of new feature will ever work. There is a ton of precedent =
for
this. </span><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>With
ViPR, you get:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>a.com
-&gt; b.com<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&#8221;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>[BA]
This seems like the crux of the matter.&nbsp; Even if the costs are =
equal to
the customer, a ViPR call could be more direct. &nbsp;If all =
that&#8217;s being
provided is basic telephony minutes, it might not be very noticeable. =
&nbsp;But
if there are services involved, you end up with a more evolvable
infrastructure.&nbsp; &nbsp;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&#8220;SIP
trunking would still get you cost savings to non-vipr connected domains =
and
non-voip installations - all of the rest of the existing PSTN endpoints.
That'll continue to be useful for a really long time. </span><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>ViPR
+ SIP trunking then further optimizes (and cost reduces) for the domains =
that
you can vipr federate with.&#8221;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>[BA]
Infrastructure ENUM potentially can count in its camp phone numbers from =
the
major consumer VOIP providers, as well as phone numbers provided by =
major SIP
trunking providers (who are also presumably utilizing infrastructure
ENUM).&nbsp; <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>While
that seems like an advantage, if the SIP trunking provider still charges =
on a
per-minute basis, even for calls which may be carried entirely over the
Internet, then there will still be an incentive for a customer to deploy =
a
direct alternative (ViPR, DUNDI, etc.) since that will allow calls made
entirely over the Internet to be made for free.&nbsp; Although we do see
cellular operators throwing in &#8220;in-networking calling&#8221;, this =
is a far cry from
throwing in intra-domain or &#8220;out of network&#8221; calls as =
well.&nbsp; This seems a
lot less likely.&nbsp; <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;</span><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

</div>

</body>

</html>

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From BeckW@telekom.de  Tue Nov 10 00:38:29 2009
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From: <BeckW@telekom.de>
To: <adam@nostrum.com>, <christer.holmberg@ericsson.com>
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Adam wrote:
> My point is that increasing the amount of ISUP information that we=20
> include in SIP message headers is going down a slippery slope, the=20
> ultimate conclusion of which is including SIP header fields for every=20
> ISUP parameter.
This is a valid point, but the proposed alternative has severe problems.
If the reason header is the only piece of ISUP information we need,
carrying the whole ISUP message is not really elegant. SIP application
servers (AS) would have to implement ISUP parsing just to check a simple
value. The content of a Reason header could be checked and used by any
plain AS. Do we really want to perpetuate ISUP code in SIP networks?

> We have a means for sending ISUP fields transparently through a SIP=20
> network (RFC 3398). I think it's folly to revisit that set of
decisions=20
> simply because someone wants a less-general point solution.
We were talking about scenarios where the PSTN gateway does not know
whether a call will be terminated by a different PSTN gateway, or by a
SIP user agent. So the originating gateway would have to encapsulate
ISUP in SIP, and SIP user agents would receive it. This would violate
regulatory requirements in various countries. As S/MIME is not feasible,
we would have to introduce yet another B2BUA just to filter ISUP from
SIP messages for end user devices.


Wolfgang=20

From john.elwell@siemens-enterprise.com  Tue Nov 10 00:49:53 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "BeckW@telekom.de" <BeckW@telekom.de>, "adam@nostrum.com" <adam@nostrum.com>, "christer.holmberg@ericsson.com" <christer.holmberg@ericsson.com>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, "gonzalo.camarillo@ericsson.com" <gonzalo.camarillo@ericsson.com>, "R.Jesske@telekom.de" <R.Jesske@telekom.de>
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of BeckW@telekom.de
> Sent: 10 November 2009 08:39
> To: adam@nostrum.com; christer.holmberg@ericsson.com
> Cc: dispatch@ietf.org; gonzalo.camarillo@ericsson.com;=20
> R.Jesske@telekom.de
> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
>=20
>=20
> Adam wrote:
> > My point is that increasing the amount of ISUP information that we=20
> > include in SIP message headers is going down a slippery slope, the=20
> > ultimate conclusion of which is including SIP header fields=20
> for every=20
> > ISUP parameter.
> This is a valid point, but the proposed alternative has=20
> severe problems.
> If the reason header is the only piece of ISUP information we need,
> carrying the whole ISUP message is not really elegant. SIP application
> servers (AS) would have to implement ISUP parsing just to=20
> check a simple
> value. The content of a Reason header could be checked and used by any
> plain AS. Do we really want to perpetuate ISUP code in SIP networks?
>=20
> > We have a means for sending ISUP fields transparently through a SIP=20
> > network (RFC 3398). I think it's folly to revisit that set of
> decisions=20
> > simply because someone wants a less-general point solution.
> We were talking about scenarios where the PSTN gateway does not know
> whether a call will be terminated by a different PSTN gateway, or by a
> SIP user agent. So the originating gateway would have to encapsulate
> ISUP in SIP, and SIP user agents would receive it. This would violate
> regulatory requirements in various countries. As S/MIME is=20
> not feasible,
> we would have to introduce yet another B2BUA just to filter ISUP from
> SIP messages for end user devices.
[JRE] SIP UAs would receive the Reason header field too, unless the last pr=
oxy/B2BUA strips it out. The only difference is that a body part can only b=
e stripped by a B2BUA, not by a proxy. Do you need a solution that will wor=
k with proxies?

John

From R.Jesske@telekom.de  Tue Nov 10 00:53:57 2009
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Date: Tue, 10 Nov 2009 09:54:12 +0100
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Thread-Topic: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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References: <9886E5FCA6D76549A3011068483A4BD40498CFB8@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F050471@zrc2hxm0.corp.nortel.com>	<1247764118.4085.24.camel@victoria-pingtel-com.us.nortel.com><1ECE0EB50388174790F9694F77522CCF1F05050C@zrc2hxm0.corp.nortel.com><4A643B95.3060800@ericsson.com><9886E5FCA6D76549A3011068483A4BD404A9C2B7@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F155AC5@zrc2hxm0.corp.nortel.com>	<CA9998CD4A020D418654FCDEF4E707DF0B1683CC@esealmw113.eemea.ericsson.se>	<1ECE0EB50388174790F9694F77522CCF1F556A65@zrc2hxm0.corp.nortel.com>	<9886E5FCA6D76549A3011068483A4BD404BFFC37@S4DE8PSAAQB.mitte.t-com.de>	<4AF37113.8030908@nostrum.com><9886E5FCA6D76549A3011068483A4BD405319E68@S4DE8PSAAQB.mitte.t-com.de><4AF7934B.7080902@cisco.com><9886E5FCA6D76549A3011068483A4BD40537238D@S4DE8PSAAQB.mitte.t-com.de>	<4AF 8AE73.40 50405@cisco.com> <CA9998CD4A020D418654FCDEF4E707DF0B16864C@esealmw113.eemea.ericsson.se> <4AF8CA10.9010502@nostrum.com>
From: <R.Jesske@telekom.de>
To: <adam@nostrum.com>, <christer.holmberg@ericsson.com>
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Cc: dispatch@ietf.org, audet@nortel.com, gonzalo.camarillo@ericsson.com
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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My intension  is to avoid your shown ABNF for new INVITE header fields.
Instead we want to use the ABNF defined within RFC3326 as follows:

 Reason            =3D  "Reason" HCOLON reason-value *(COMMA =
reason-value)
 reason-value      =3D  protocol *(SEMI reason-params)
 protocol          =3D  "SIP" / "Q.850" / token
 reason-params     =3D  protocol-cause / reason-text
                      / reason-extension
 protocol-cause    =3D  "cause" EQUAL cause
 cause             =3D  1*DIGIT
 reason-text       =3D  "text" EQUAL quoted-string
 reason-extension  =3D  generic-param

The only one missing is to allow this header within Responses.

=20

-----Urspr=FCngliche Nachricht-----
Von: Adam Roach [mailto:adam@nostrum.com]=20
Gesendet: Dienstag, 10. November 2009 03:04
An: Christer Holmberg
Cc: Paul Kyzivat; Jesske, Roland; audet@nortel.com; dispatch@ietf.org; =
Gonzalo Camarillo
Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

To clarify the point I was trying to make in the meeting today -- if=20
we've decided to go down this path, I'll save you the trouble of=20
creating the new ABNF for the new INVITE header fields. Here's what they =

look like:

Nature-Of-Connection-Indicators          =3D =
"Nature-Of-Connection-Indicators"
                                            HCOLON quoted-string

Forward-Call-Indicators                  =3D "Forward-Call-Indicators"
                                            HCOLON quoted-string

Calling-Party's-Category                 =3D "Calling-Party's-Category"
                                            HCOLON quoted-string

Transmission-Medium-Requirement          =3D =
"Transmission-Medium-Requirement"
                                            HCOLON quoted-string

Called-Party-Number                      =3D "Called-Party-Number"
                                            HCOLON quoted-string

Transit-Network-Selection                =3D "Transit-Network-Selection"
                                            HCOLON quoted-string

Call-Reference                           =3D "Call-Reference"
                                            HCOLON quoted-string

Calling-Party-Number                     =3D "Calling-Party-Number"
                                            HCOLON quoted-string

Optional-Forward-Call-Indicators         =3D=20
"Optional-Forward-Call-Indicators"
                                            HCOLON quoted-string

Redirecting-Number                       =3D "Redirecting-Number"
                                            HCOLON quoted-string

Redirection-Information                  =3D "Redirection-Information"
                                            HCOLON quoted-string

Closed-User-Group-Interlock-Code         =3D=20
"Closed-User-Group-Interlock-Code"
                                            HCOLON quoted-string

Connection-Request                       =3D "Connection-Request"
                                            HCOLON quoted-string

Original-Called-Number                   =3D "Original-Called-Number"
                                            HCOLON quoted-string

User-to-user-Information                 =3D "User-to-user-Information"
                                            HCOLON quoted-string

Access-Transport                         =3D "Access-Transport"
                                            HCOLON quoted-string

User-Service-Information                 =3D "User-Service-Information"
                                            HCOLON quoted-string

User-to-user-Indicators                  =3D "User-to-user-Indicators"
                                            HCOLON quoted-string

Generic-Number                           =3D "Generic-Number"
                                            HCOLON quoted-string

Propagation-Delay-Counter                =3D "Propagation-Delay-Counter"
                                            HCOLON quoted-string

User-Service-Information-Prime           =3D =
"User-Service-Information-Prime"
                                            HCOLON quoted-string

Network-Specific-Facility                =3D "Network-Specific-Facility"
                                            HCOLON quoted-string

Generic-Digits                           =3D "Generic-Digits"
                                            HCOLON quoted-string

Origination-ISC-Point-Code               =3D =
"Origination-ISC-Point-Code"
                                            HCOLON quoted-string

User-Teleservice-Information             =3D =
"User-Teleservice-Information"
                                            HCOLON quoted-string

Remote-Operations                        =3D "Remote-Operations"
                                            HCOLON quoted-string

Parameter-Compatibility-Information      =3D=20
"Parameter-Compatibility-Information"
                                            HCOLON quoted-string

Generic-Notification-Indicator           =3D =
"Generic-Notification-Indicator"
                                            HCOLON quoted-string

Service-Activation                       =3D "Service-Activation"
                                            HCOLON quoted-string

Generic-Reference                        =3D "Generic-Reference"
                                            HCOLON quoted-string

MLPP-Precedence                          =3D "MLPP-Precedence"
                                            HCOLON quoted-string

Transmission-Medium-Requirement-Prime    =3D=20
"Transmission-Medium-Requirement-Prime"
                                            HCOLON quoted-string

Location-Number                          =3D "Location-Number"
                                            HCOLON quoted-string

Forward-GVNS                             =3D "Forward-GVNS"
                                            HCOLON quoted-string

CCSS                                     =3D "CCSS"
                                            HCOLON quoted-string

Network-Management-Controls              =3D =
"Network-Management-Controls"
                                            HCOLON quoted-string

Circuit-Assignment-Map                   =3D "Circuit-Assignment-Map"
                                            HCOLON quoted-string

Correlation-Id                           =3D "Correlation-Id"
                                            HCOLON quoted-string

Call-Diversion-Treatment-Indicators      =3D=20
"Call-Diversion-Treatment-Indicators"
                                            HCOLON quoted-string

Called-IN-Number                         =3D "Called-IN-Number"
                                            HCOLON quoted-string

Call-Offering-Treatment-Indicators       =3D=20
"Call-Offering-Treatment-Indicators"
                                            HCOLON quoted-string

Conference-Treatment-Indicators          =3D =
"Conference-Treatment-Indicators"
                                            HCOLON quoted-string

SCF-Id                                   =3D "SCF-Id"
                                            HCOLON quoted-string

UID-Capability-Indicators                =3D "UID-Capability-Indicators"
                                            HCOLON quoted-string

Echo-Control-Information                 =3D "Echo-Control-Information"
                                            HCOLON quoted-string

Hop-Counter                              =3D "Hop-Counter"
                                            HCOLON quoted-string

Collect-Call-Request                     =3D "Collect-Call-Request"
                                            HCOLON quoted-string

Application-Transport-Parameter          =3D =
"Application-Transport-Parameter"
                                            HCOLON quoted-string

Pivot-Capability                         =3D "Pivot-Capability"
                                            HCOLON quoted-string

Called-Directory-Number                  =3D "Called-Directory-Number"
                                            HCOLON quoted-string

Original-Called-IN-Number                =3D "Original-Called-IN-Number"
                                            HCOLON quoted-string

Calling-Geodetic-Location                =3D "Calling-Geodetic-Location"
                                            HCOLON quoted-string

Network-Routing-Number                   =3D "Network-Routing-Number"
                                            HCOLON quoted-string

QoR-Capability                           =3D "QoR-Capability"
                                            HCOLON quoted-string

Pivot-Counter                            =3D "Pivot-Counter"
                                            HCOLON quoted-string

Pivot-Routing-Forward-Information        =3D=20
"Pivot-Routing-Forward-Information"
                                            HCOLON quoted-string

Redirect-Capability                      =3D "Redirect-Capability"
                                            HCOLON quoted-string

Redirect-Counter                         =3D "Redirect-Counter"
                                            HCOLON quoted-string

Redirect-Status                          =3D "Redirect-Status"
                                            HCOLON quoted-string

Redirect-Forward-Information             =3D =
"Redirect-Forward-Information"
                                            HCOLON quoted-string

Number-Portability-Forward-Information   =3D
                                       =20
"Number-Portability-Forward-Information"
                                            HCOLON quoted-string

Perhaps we can refine some the the quoted strings with tokens, but I=20
think this is a good start.

/a

From john.elwell@siemens-enterprise.com  Tue Nov 10 01:07:03 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: DISPATCH <dispatch@ietf.org>
Date: Tue, 10 Nov 2009 10:07:21 +0100
Thread-Topic: SIP Forum UA config work - status update
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Subject: [dispatch] SIP Forum UA config work - status update
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The SIP Forum work on UA config is nearing completion and "last call" of th=
e specification is expected to start in a couple of weeks. The latest draft=
 (v17) is available here:
http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,3=
61/Itemid,261/

There is an accompanying presentation that Scott Lawrence put together and =
presented at SIPit to an interested audience. This is available here:
http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,3=
21/Itemid,261/

Note that the specification has changed considerably since we last presente=
d to DISPATCH.

We were hoping to find a room to present the work to interested IETF folks,=
 but we failed to do so. However, I would be prepared to talk it through in=
formally to anybody interested. We can meet at 08.00 Friday at IETF registr=
ation. Let me know if interested, and also bring your own downloaded copy o=
f the presentation, since there won't be a projector.

John=

From thomas.belling@nsn.com  Tue Nov 10 04:14:14 2009
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References: <9886E5FCA6D76549A3011068483A4BD40498CFB8@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F050471@zrc2hxm0.corp.nortel.com>	<1247764118.4085.24.camel@victoria-pingtel-com.us.nortel.com><1ECE0EB50388174790F9694F77522CCF1F05050C@zrc2hxm0.corp.nortel.com><4A643B95.3060800@ericsson.com><9886E5FCA6D76549A3011068483A4BD404A9C2B7@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F155AC5@zrc2hxm0.corp.nortel.com>	<CA9998CD4A020D418654FCDEF4E707DF0B1683CC@esealmw113.eemea.ericsson.se>	<1ECE0EB50388174790F9694F77522CCF1F556A65@zrc2hxm0.corp.nortel.com>	<9886E5FCA6D76549A3011068483A4BD404BFFC37@S4DE8PSAAQB.mitte.t-com.de>	<4AF37113.8030908@nostrum.com><9886E5FCA6D76549A3011068483A4BD405319E68@S4DE8PSAAQB.mitte.t-com.de><4AF7934B.7080902@cisco.com><9886E5FCA6D76549A3011068483A4BD40537238D@S4DE8PSAAQB.mitte.t-com.de>	<4AF8AE73.40 50405@cisco.com><CA9998CD4A020D418654FCDEF4E707DF0B16864C@esealmw113.eemea.ericsson.se><4AF8CA10.9010502@nostrum.com> <988! 6E5FCA6D7 6549A3011068483A4BD4053724C3@S4DE8PSAAQB.mitte.t-com.de>
From: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
To: <R.Jesske@telekom.de>, <adam@nostrum.com>, <christer.holmberg@ericsson.com>
X-OriginalArrivalTime: 10 Nov 2009 12:14:29.0773 (UTC) FILETIME=[5A94B3D0:01CA61FF]
Cc: audet@nortel.com, dispatch@ietf.org, gonzalo.camarillo@ericsson.com
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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I am bit surprised that the principle of the reason haeder in general =
and reason headers in requests are discussed. I thought this has been =
agreeed in RFC 3326.
In my understnding the intention of the draft is to allow reason headers =
in responses.

Thomas B

----------------------------------=20
Dr. Thomas Belling=20
3GPP Standardisation
Nokia Siemens Networks=20



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of ext R.Jesske@telekom.de
Sent: Tuesday, November 10, 2009 4:54 PM
To: adam@nostrum.com; christer.holmberg@ericsson.com
Cc: dispatch@ietf.org; audet@nortel.com; gonzalo.camarillo@ericsson.com
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

My intension  is to avoid your shown ABNF for new INVITE header fields.
Instead we want to use the ABNF defined within RFC3326 as follows:

 Reason            =3D  "Reason" HCOLON reason-value *(COMMA =
reason-value)
 reason-value      =3D  protocol *(SEMI reason-params)
 protocol          =3D  "SIP" / "Q.850" / token
 reason-params     =3D  protocol-cause / reason-text
                      / reason-extension
 protocol-cause    =3D  "cause" EQUAL cause
 cause             =3D  1*DIGIT
 reason-text       =3D  "text" EQUAL quoted-string
 reason-extension  =3D  generic-param

The only one missing is to allow this header within Responses.

=20

-----Urspr=FCngliche Nachricht-----
Von: Adam Roach [mailto:adam@nostrum.com]
Gesendet: Dienstag, 10. November 2009 03:04
An: Christer Holmberg
Cc: Paul Kyzivat; Jesske, Roland; audet@nortel.com; dispatch@ietf.org; =
Gonzalo Camarillo
Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

To clarify the point I was trying to make in the meeting today -- if =
we've decided to go down this path, I'll save you the trouble of =
creating the new ABNF for the new INVITE header fields. Here's what they =
look like:

Nature-Of-Connection-Indicators          =3D =
"Nature-Of-Connection-Indicators"
                                            HCOLON quoted-string

Forward-Call-Indicators                  =3D "Forward-Call-Indicators"
                                            HCOLON quoted-string

Calling-Party's-Category                 =3D "Calling-Party's-Category"
                                            HCOLON quoted-string

Transmission-Medium-Requirement          =3D =
"Transmission-Medium-Requirement"
                                            HCOLON quoted-string

Called-Party-Number                      =3D "Called-Party-Number"
                                            HCOLON quoted-string

Transit-Network-Selection                =3D "Transit-Network-Selection"
                                            HCOLON quoted-string

Call-Reference                           =3D "Call-Reference"
                                            HCOLON quoted-string

Calling-Party-Number                     =3D "Calling-Party-Number"
                                            HCOLON quoted-string

Optional-Forward-Call-Indicators         =3D=20
"Optional-Forward-Call-Indicators"
                                            HCOLON quoted-string

Redirecting-Number                       =3D "Redirecting-Number"
                                            HCOLON quoted-string

Redirection-Information                  =3D "Redirection-Information"
                                            HCOLON quoted-string

Closed-User-Group-Interlock-Code         =3D=20
"Closed-User-Group-Interlock-Code"
                                            HCOLON quoted-string

Connection-Request                       =3D "Connection-Request"
                                            HCOLON quoted-string

Original-Called-Number                   =3D "Original-Called-Number"
                                            HCOLON quoted-string

User-to-user-Information                 =3D "User-to-user-Information"
                                            HCOLON quoted-string

Access-Transport                         =3D "Access-Transport"
                                            HCOLON quoted-string

User-Service-Information                 =3D "User-Service-Information"
                                            HCOLON quoted-string

User-to-user-Indicators                  =3D "User-to-user-Indicators"
                                            HCOLON quoted-string

Generic-Number                           =3D "Generic-Number"
                                            HCOLON quoted-string

Propagation-Delay-Counter                =3D "Propagation-Delay-Counter"
                                            HCOLON quoted-string

User-Service-Information-Prime           =3D =
"User-Service-Information-Prime"
                                            HCOLON quoted-string

Network-Specific-Facility                =3D "Network-Specific-Facility"
                                            HCOLON quoted-string

Generic-Digits                           =3D "Generic-Digits"
                                            HCOLON quoted-string

Origination-ISC-Point-Code               =3D =
"Origination-ISC-Point-Code"
                                            HCOLON quoted-string

User-Teleservice-Information             =3D =
"User-Teleservice-Information"
                                            HCOLON quoted-string

Remote-Operations                        =3D "Remote-Operations"
                                            HCOLON quoted-string

Parameter-Compatibility-Information      =3D=20
"Parameter-Compatibility-Information"
                                            HCOLON quoted-string

Generic-Notification-Indicator           =3D =
"Generic-Notification-Indicator"
                                            HCOLON quoted-string

Service-Activation                       =3D "Service-Activation"
                                            HCOLON quoted-string

Generic-Reference                        =3D "Generic-Reference"
                                            HCOLON quoted-string

MLPP-Precedence                          =3D "MLPP-Precedence"
                                            HCOLON quoted-string

Transmission-Medium-Requirement-Prime    =3D=20
"Transmission-Medium-Requirement-Prime"
                                            HCOLON quoted-string

Location-Number                          =3D "Location-Number"
                                            HCOLON quoted-string

Forward-GVNS                             =3D "Forward-GVNS"
                                            HCOLON quoted-string

CCSS                                     =3D "CCSS"
                                            HCOLON quoted-string

Network-Management-Controls              =3D =
"Network-Management-Controls"
                                            HCOLON quoted-string

Circuit-Assignment-Map                   =3D "Circuit-Assignment-Map"
                                            HCOLON quoted-string

Correlation-Id                           =3D "Correlation-Id"
                                            HCOLON quoted-string

Call-Diversion-Treatment-Indicators      =3D=20
"Call-Diversion-Treatment-Indicators"
                                            HCOLON quoted-string

Called-IN-Number                         =3D "Called-IN-Number"
                                            HCOLON quoted-string

Call-Offering-Treatment-Indicators       =3D=20
"Call-Offering-Treatment-Indicators"
                                            HCOLON quoted-string

Conference-Treatment-Indicators          =3D =
"Conference-Treatment-Indicators"
                                            HCOLON quoted-string

SCF-Id                                   =3D "SCF-Id"
                                            HCOLON quoted-string

UID-Capability-Indicators                =3D "UID-Capability-Indicators"
                                            HCOLON quoted-string

Echo-Control-Information                 =3D "Echo-Control-Information"
                                            HCOLON quoted-string

Hop-Counter                              =3D "Hop-Counter"
                                            HCOLON quoted-string

Collect-Call-Request                     =3D "Collect-Call-Request"
                                            HCOLON quoted-string

Application-Transport-Parameter          =3D =
"Application-Transport-Parameter"
                                            HCOLON quoted-string

Pivot-Capability                         =3D "Pivot-Capability"
                                            HCOLON quoted-string

Called-Directory-Number                  =3D "Called-Directory-Number"
                                            HCOLON quoted-string

Original-Called-IN-Number                =3D "Original-Called-IN-Number"
                                            HCOLON quoted-string

Calling-Geodetic-Location                =3D "Calling-Geodetic-Location"
                                            HCOLON quoted-string

Network-Routing-Number                   =3D "Network-Routing-Number"
                                            HCOLON quoted-string

QoR-Capability                           =3D "QoR-Capability"
                                            HCOLON quoted-string

Pivot-Counter                            =3D "Pivot-Counter"
                                            HCOLON quoted-string

Pivot-Routing-Forward-Information        =3D=20
"Pivot-Routing-Forward-Information"
                                            HCOLON quoted-string

Redirect-Capability                      =3D "Redirect-Capability"
                                            HCOLON quoted-string

Redirect-Counter                         =3D "Redirect-Counter"
                                            HCOLON quoted-string

Redirect-Status                          =3D "Redirect-Status"
                                            HCOLON quoted-string

Redirect-Forward-Information             =3D =
"Redirect-Forward-Information"
                                            HCOLON quoted-string

Number-Portability-Forward-Information   =3D
                                       =20
"Number-Portability-Forward-Information"
                                            HCOLON quoted-string

Perhaps we can refine some the the quoted strings with tokens, but I =
think this is a good start.

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

From klaus.mailinglists@pernau.at  Tue Nov 10 06:51:10 2009
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Subject: Re: [dispatch] Introducing ViPR: a new federation technology
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Richard Shockey schrieb:
> OK Asterisk DUNDi on steroids. 
> 
> Useful for enterprise to enterprise supply chain federations, since TRIP was
> never going to work there, but to a SIP service provider it provides no real
> value.

I guess it depends on the provider's business model. If the provider 
wants to receive termination fees for incoming calls, then he wont use 
ViPR. If the provider does not care about termination fees, then he can 
allow incoming calls with ViPR, e.g. to save gateway resource.

Anyway, it can use ViPR on outgoing calls to avoid transit costs.

regards
Klaus

> Did I miss something?
> 
>>  -----Original Message-----
>>  From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>>  Behalf Of Jonathan Rosenberg
>>  Sent: Monday, November 09, 2009 2:44 AM
>>  To: dispatch@ietf.org
>>  Subject: [dispatch] Introducing ViPR: a new federation technology
>>  
>>  Folks,
>>  
>>  As you are all well aware, SIP has been very successful in the
>>  marketplace as a tool for voice and video within a single domain. Its
>>  success for inter-domain federation has been more limited, and it has
>>  seen almost no deployment over the public Internet for any-to-any
>>  federation, even though this was the model originally envisaged for
>>  SIP,
>>  and a model I still believe in.
>>  
>>  This is something that needed to be fixed, and I have fixed it.
>>  
>>  Today, Cullen and I submitted documents describing a new technology
>>  called ViPR. ViPR is a new federation technique which will finally
>>  enable the any-to-any SIP federation that we have always wanted for
>>  SIP.
>>  It solves several hard problems that have prevented widespread
>>  federation. In particular, it solves the phone number routing problem,
>>  providing a non-centralized technique that securely maps phone numbers
>>  to domains. It also provides a built-in solution for the VoIP anti-
>>  spam
>>  problem - a new one that is specific to VoIP. As such, it enables
>>  worldwide, scalable, any-to-any federation over the public Internet,
>>  securely, for phone numbers, and in a way that supports incremental
>>  deployability.
>>  
>>  This technology is not just documents - it is working code, and was
>>  announced by Cisco today as part of their products shipping early next
>>  year.
>>  
>>  I'm extremely excited about ViPR, and I think others will be too.
>>  
>>  The documents are:
>>  
>>  
>>  http://www.ietf.org/id/draft-rosenberg-dispatch-vipr-overview-01.txt
>>  http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-vap-
>>  00.txt
>>  http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-pvp-
>>  00.txt
>>  http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-sip-
>>  antispam-00.txt
>>  http://www.ietf.org/internet-drafts/draft-rosenberg-dispatch-vipr-
>>  reload-usage-00.txt
>>  
>>  
>>  Thanks,
>>  Jonathan Rosenberg
>>  jdrosen@jdrosen.net
>>  _______________________________________________
>>  dispatch mailing list
>>  dispatch@ietf.org
>>  https://www.ietf.org/mailman/listinfo/dispatch
> 
> 

From klaus.mailinglists@pernau.at  Tue Nov 10 07:02:26 2009
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From: Klaus Darilion <klaus.mailinglists@pernau.at>
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Bernard Aboba schrieb:
> Also, if one buys the arguments, then one could conclude that all that is
> necessary for VOIP bypass is a PRI and an Internet connection.  Why bother
> with SIP trunking? However, that argument assumes that as the decline in the

It is not about PRI vs. SIP trunking, but about PSTN connectivity vs. 
adhoc SIP peering.

The trust is based on a phone call that happened on the PSTN (PSTN means 
telcos/carriers with interconnect and customers connected to those telcos).

It does not matter which technology the telcos use for interconnection 
or for connections of its customers. E.g. an enterprise which already 
uses a SIP trunk to a SIP service provider is also connected to the 
PSTN. That mean for routes which are not routable with ViPR the SIP 
trunk as used as the SIP trunk is part of the PSTN.

regards
klaus

> PSTN accelerates that SIP Trunking providers will utilize infrastructure
> ENUM to their advantage to route calls directly over the Internet without
> passing on the savings to customers. 
> 
> However, the evolution of Internet peering would indicate this not the most
> likely outcome.  Back in the mid-1990s, peering was a significant issue for
> second Tier ISPs and access cost much more than it does today.  Today recent
> measurements show that 50 percent of Internet traffic is originating in only
> 150 ASes, and that kind of consolidation also tends to go along with major
> declines in consumer costs and an lessening of importance of peering
> concerns.  
> 
> So my own guess is that as SIP trunking becomes a commodity that the
> industry will grow more and more competitive and that in a decade
> infrastructure ENUM will apply to a very large fraction of all calls, with
> the benefits being passed on to customers. 
> 
> In that kind of scenario, ViPR would be unnecessary. 
> 
> 

From victor.pascual.avila@gmail.com  Tue Nov 10 08:55:54 2009
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Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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As a very brief summary:

- Unanimous Agreement that we do have a problem to be solved
- Next step is to work the requirements until we have better agreement
- Several folks indicated interest and willingness to be involved in
ongoing work

Cheers,
-Victor


On Mon, Nov 9, 2009 at 11:21 PM, Mary Barnes <mary.barnes@nortel.com> wrote=
:
>
> Hi all,
>
> I have uploaded Jim McEachern's notes from the Identity Adhoc yesterday:
> http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/Identity-AdHoc-N=
otes-jmce.doc
>
> Regards,
> Mary.
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>



--=20
Victor Pascual =C3=81vila

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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Introducing ViPR: a new federation technology - VIDEO
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Bernard Aboba wrote:

> [BA] The question in my mind is whether video really is “special” (in 
> the IM&P sense) or not.  That is, today we accept the idea that anyone 
> should be able to complete a phone call to anyone else.  We also accept 
> this for SMS.  However, we don’t accept that idea for Instant Messaging 
> (which typically requires being in the buddy list). Do we accept that 
> principle for video?  I’m not sure.  There are some inherent 
> disadvantages to accepting video calls from anywhere even if we have a 
> verified phone number on the other end.  

I don't think this has much of anything to do with buddy lists.
Even when my mother calls, whether I want to transmit video will vary 
depending on current state. (Am I dressed, is there anything/anybody in 
sight that I don't want her to see, ...)

There are mechanisms and conventions yet to be developed around this. 
There has been little motivation to do it so far because there has not 
been use cases that demand it.

It seems very likely to me that a simple answer is that a video phone 
will typically *answer* in video recvonly mode, and require some manual 
intervention to start transmitting. And when initiating a call from a 
video phone there of course must also be options as to whether to 
transmit by default.

Other more sophisticated features can evolve. But I think they are all 
compatible with a "call anybody" model.

It may however put more of a premium on accurate callerid and calleeid 
information.

	Thanks,
	Paul

From BeckW@telekom.de  Tue Nov 10 17:04:58 2009
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From: <BeckW@telekom.de>
To: <adam@nostrum.com>
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04; why encapsulation is problematic
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Adam wrote:
> S/MIME for a general purpose network suffers from the classical key=20
> distribution problem. It's not used for the same reason encrypted
email=20
> is not used (except among the geek community).
>=20
> But you're not talking about general purpose networks.
> You're talking about tightly constrained networks with PSTN gateways=20
> that are either all within a single administrative domain, or that=20
> communicate with each other through bilateral agreements. In these=20
> circumstances, key distribution is no more difficult a problem than IP
> address assignment. It's a simple matter of configuration.
In theory, S/MIME should be doable in an administrative domain. It's the
lack of implementations that troubles me.=20
=20
Adam wrote:
> B2BUAs will break whatever they break. Until someone sits down and=20
> documents their behavior, we can't really do too much to work around=20
> their behavior. It's the same problem we're having all over the place=20
> (identity, MSRP ACM, etc). There's no point in talking about something

> as nebulous as B2BUA behavior.
Sure. I was referring to an argument made (not by Adam) in the meeting.=20

Adam wrote:
> While many UAs won't be able to ignore the multitype body (aside: this

> is sad -- email clients all can), you'd be hard pressed to find one
that=20
> can't understand multipart that also won't respond to a multipart body

> in a request with a 415.
Right, but connection setup times are already a concern now, without the
extra transaction to reject multipart content.

Adam wrote:
> At one point, you said "IMS." I don't think you'll find an IMS client=20
> that has any hope of receiving a UDP SIP message -- show me an INVITE
on=20
> the terminating side of an IMS network that is somehow smaller than
1500=20
> bytes, and I'll be shocked into silence. TCP is a foregone conclusion.
The universal deployment of IMS may take some time. Until that glorious
day, we may have networks that would profit from the Reason header and
don't have big SIP messages.=20

Wolfgang

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From: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
To: <R.Jesske@telekom.de>, <dispatch@ietf.org>
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Cc: "ext Calme, James A \(Jim\)" <calme@alcatel-lucent.com>, Paul While <paul.while@ericsson.com>
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Dear Roland,

we discussed the interworking of ISUP ACM and CPG causes to SIP reasons =
headers in provisional responses in 3GPP CT3 yesterday.
The conclusion was that we will only insert a reason header in a 183 =
provisional response.

May I therefor suggest the following wording:

        The Reason header is applicable to 3xx, 4xx, 5xx and 6xx final =
responses
        and 183 and 199 <reference> provisional responses.

Further, it would be very helpfull for 3GPP if you could make the =
updated draft available very quickly as we have several dependent CRs in =
the 3GPP CTx meeting which is ongoing this week.

Thomas


----------------------------------=20
Dr. Thomas Belling=20
3GPP Standardisation
Nokia Siemens Networks=20



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of ext Christer Holmberg
Sent: Tuesday, November 10, 2009 8:12 AM
To: Paul Kyzivat; R.Jesske@telekom.de
Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04


Hi,

A small editorial change proposal:

"The appearance of the Reason header is applicable to 3xx, 4xx, 5xx and =
6xx final responses and 18x and 199 <reference> provisional responsess."

Regards,

Christer



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Paul Kyzivat
Sent: 10. marraskuuta 2009 3:06
To: R.Jesske@telekom.de
Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

Roland,

Thanks! That addresses my concern.

	Thanks,
	Paul

R.Jesske@telekom.de wrote:
> Hello Paul,
> Thanks for that hint. I will change this too.=20
> Like:
>=20
>  The appearance of the Reason header is applicable to final responses
>        3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as =
defined within
>        ietf-sipcore-199.txt. Also the Reason header is
>        applicable to provisional responses18x.
>=20
> Best Regards
>=20
> Roland
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Gesendet: Montag, 9. November 2009 12:58
> An: Jesske, Roland
> Cc: adam@nostrum.com; audet@nortel.com; dispatch@ietf.org;=20
> gonzalo.camarillo@ericsson.com
> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
>=20
>=20
> R.Jesske@telekom.de wrote:
>> Hi Adam,
>> Thank you for your mail. So I propose the following text then:
>>
>> The appearance of the Reason header is applicable to final responses
>>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as =
defined within
>>       ietf-sipcore-199.txt. Also the Reason header is
>>       applicable to provisional responses18x which are generated by =
an
>>       interworking gateway from ISUP to SIP.</t>
>=20
> I think this is ok *except* if you are going to allow one kind of UAS=20
> to use Reason in a 18x response, then it should be allowed for *any*=20
> UAS to do so.
>=20
> For instance, suppose the gateway is replaced by a B2BUA to another=20
> sip environment, and the request eventually arrives as a gateway. Then =

> a 18x response with a Reason header may arrive at the B2BUA and need=20
> to be replicated on the other side.
>=20
> 	Thanks,
> 	Paul
>=20
>> Best Beagards
>>
>> Roland
>>
>> -----Urspr=FCngliche Nachricht-----
>> Von: Adam Roach [mailto:adam@nostrum.com]
>> Gesendet: Freitag, 6. November 2009 01:43
>> An: Jesske, Roland
>> Cc: audet@nortel.com; christer.holmberg@ericsson.com;=20
>> gonzalo.camarillo@ericsson.com; dispatch@ietf.org
>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
>>> Hi Christer and Francois,
>>> I have added a sentence under section Overall Applicability:
>>>
>>>
>>> The appearance of the Reason header is applicable to final responses =
4xx, 5xx and 6xx and in addition for 199 Responses.
>>>
>>> Is this proper enough? Or do you have more in mind?
>>>   =20
>> Given that RFC 3398 specifies the generation of a 301 in response to=20
>> a cause code of 22 under certain circumstances, it seems pretty=20
>> appropriate to allow the inclusion of a Q.850 cause code in 300-class =

>> responses also. Or, at least, as appropriate as it would be to=20
>> include them in 4xx, 5xx, and 6xx responses.
>>
>> /a
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>=20
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

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The DREGS ad hoc will be in Orchard East, 11.45 - 12.40 JT tomorrow  
(Thursday).

We do need scribes and jabber scribes.  Please respond directly to me  
to volunteer.  Thanks.

From mary.barnes@nortel.com  Wed Nov 11 02:45:10 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Eric Burger" <eburger@standardstrack.com>, <dispatch@ietf.org>
Subject: Re: [dispatch] DREGS ad hoc
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Orchard -> Orchid  =20

-----Original Message-----
From: dispatch-bounces@ietf.org on behalf of Eric Burger
Sent: Wed 11/11/2009 2:26 AM
To: dispatch@ietf.org
Subject: [dispatch] DREGS ad hoc
=20
The DREGS ad hoc will be in Orchard East, 11.45 - 12.40 JT tomorrow =20
(Thursday).

We do need scribes and jabber scribes.  Please respond directly to me =20
to volunteer.  Thanks.
_______________________________________________
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https://www.ietf.org/mailman/listinfo/dispatch


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<P><FONT SIZE=3D2>Orchard -&gt; Orchid&nbsp;&nbsp;<BR>
<BR>
-----Original Message-----<BR>
From: dispatch-bounces@ietf.org on behalf of Eric Burger<BR>
Sent: Wed 11/11/2009 2:26 AM<BR>
To: dispatch@ietf.org<BR>
Subject: [dispatch] DREGS ad hoc<BR>
<BR>
The DREGS ad hoc will be in Orchard East, 11.45 - 12.40 JT =
tomorrow&nbsp;<BR>
(Thursday).<BR>
<BR>
We do need scribes and jabber scribes.&nbsp; Please respond directly to =
me&nbsp;<BR>
to volunteer.&nbsp; Thanks.<BR>
_______________________________________________<BR>
dispatch mailing list<BR>
dispatch@ietf.org<BR>
<A =
HREF=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
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From pkyzivat@cisco.com  Wed Nov 11 05:57:14 2009
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Date: Wed, 11 Nov 2009 08:57:51 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
References: <9886E5FCA6D76549A3011068483A4BD40498CFB8@S4DE8PSAAQB.mitte.t-com.de>	<1247764118.4085.24.camel@victoria-pingtel-com.us.nortel.com><1ECE0EB50388174790F9694F77522CCF1F05050C@zrc2hxm0.corp.nortel.com><4A643B95.3060800@ericsson.com><9886E5FCA6D76549A3011068483A4BD404A9C2B7@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F155AC5@zrc2hxm0.corp.nortel.com>	<CA9998CD4A020D418654FCDEF4E707DF0B1683CC@esealmw113.eemea.ericsson.se>	<1ECE0EB50388174790F9694F77522CCF1F556A65@zrc2hxm0.corp.nortel.com>	<9886E5FCA6D76549A3011068483A4BD404BFFC37@S4DE8PSAAQB.mitte.t-com.de>	<4AF37113.8030908@nostrum.com><9886E5FCA6D76549A3011068483A4BD405319E68@S4DE8PSAAQB.mitte.t-com.de><4AF7934B.7080902@cisco.com><9886E5FCA6D76549A3011068483A4BD40537238D@S4DE8PSAAQB.mitte.t-com.de><4AF!	8AE73.40 50405@cisco .com>	<CA9998CD4A020D418654FCDEF4E707DF0B16864C@esealmw113.eemea.ericsson.se> <744A66DF31B5B34EA8B08BBD8187A722C6423E@DEMUEXC014.nsn-intra.net>
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Cc: Paul While <paul.while@ericsson.com>, dispatch@ietf.org, R.Jesske@telekom.de, "ext Calme, James A \(Jim\)" <calme@alcatel-lucent.com>
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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I hate to see *anything* restricted to 183, rather than being applicable 
to all 1xx (or at least 18x) responses.

Restricting to 183 implies that there is something *special* about 183.
I have seen that erroneous assumption in the wild too - and it is 
distressing. Anything you can do with 183 should also work with 180.

	Thanks,
	Paul

Belling, Thomas (NSN - DE/Munich) wrote:
> Dear Roland,
> 
> we discussed the interworking of ISUP ACM and CPG causes to SIP reasons headers in provisional responses in 3GPP CT3 yesterday.
> The conclusion was that we will only insert a reason header in a 183 provisional response.
> 
> May I therefor suggest the following wording:
> 
>         The Reason header is applicable to 3xx, 4xx, 5xx and 6xx final responses
>         and 183 and 199 <reference> provisional responses.
> 
> Further, it would be very helpfull for 3GPP if you could make the updated draft available very quickly as we have several dependent CRs in the 3GPP CTx meeting which is ongoing this week.
> 
> Thomas
> 
> 
> ---------------------------------- 
> Dr. Thomas Belling 
> 3GPP Standardisation
> Nokia Siemens Networks 
> 
> 
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of ext Christer Holmberg
> Sent: Tuesday, November 10, 2009 8:12 AM
> To: Paul Kyzivat; R.Jesske@telekom.de
> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> 
> 
> Hi,
> 
> A small editorial change proposal:
> 
> "The appearance of the Reason header is applicable to 3xx, 4xx, 5xx and 6xx final responses and 18x and 199 <reference> provisional responsess."
> 
> Regards,
> 
> Christer
> 
> 
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 10. marraskuuta 2009 3:06
> To: R.Jesske@telekom.de
> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> 
> Roland,
> 
> Thanks! That addresses my concern.
> 
> 	Thanks,
> 	Paul
> 
> R.Jesske@telekom.de wrote:
>> Hello Paul,
>> Thanks for that hint. I will change this too. 
>> Like:
>>
>>  The appearance of the Reason header is applicable to final responses
>>        3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as defined within
>>        ietf-sipcore-199.txt. Also the Reason header is
>>        applicable to provisional responses18x.
>>
>> Best Regards
>>
>> Roland
>>
>> -----Ursprüngliche Nachricht-----
>> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Gesendet: Montag, 9. November 2009 12:58
>> An: Jesske, Roland
>> Cc: adam@nostrum.com; audet@nortel.com; dispatch@ietf.org; 
>> gonzalo.camarillo@ericsson.com
>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>>
>>
>> R.Jesske@telekom.de wrote:
>>> Hi Adam,
>>> Thank you for your mail. So I propose the following text then:
>>>
>>> The appearance of the Reason header is applicable to final responses
>>>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as defined within
>>>       ietf-sipcore-199.txt. Also the Reason header is
>>>       applicable to provisional responses18x which are generated by an
>>>       interworking gateway from ISUP to SIP.</t>
>> I think this is ok *except* if you are going to allow one kind of UAS 
>> to use Reason in a 18x response, then it should be allowed for *any* 
>> UAS to do so.
>>
>> For instance, suppose the gateway is replaced by a B2BUA to another 
>> sip environment, and the request eventually arrives as a gateway. Then 
>> a 18x response with a Reason header may arrive at the B2BUA and need 
>> to be replicated on the other side.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Best Beagards
>>>
>>> Roland
>>>
>>> -----Ursprüngliche Nachricht-----
>>> Von: Adam Roach [mailto:adam@nostrum.com]
>>> Gesendet: Freitag, 6. November 2009 01:43
>>> An: Jesske, Roland
>>> Cc: audet@nortel.com; christer.holmberg@ericsson.com; 
>>> gonzalo.camarillo@ericsson.com; dispatch@ietf.org
>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>
>>> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
>>>> Hi Christer and Francois,
>>>> I have added a sentence under section Overall Applicability:
>>>>
>>>>
>>>> The appearance of the Reason header is applicable to final responses 4xx, 5xx and 6xx and in addition for 199 Responses.
>>>>
>>>> Is this proper enough? Or do you have more in mind?
>>>>    
>>> Given that RFC 3398 specifies the generation of a 301 in response to 
>>> a cause code of 22 under certain circumstances, it seems pretty 
>>> appropriate to allow the inclusion of a Q.850 cause code in 300-class 
>>> responses also. Or, at least, as appropriate as it would be to 
>>> include them in 4xx, 5xx, and 6xx responses.
>>>
>>> /a
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From R.Jesske@telekom.de  Wed Nov 11 06:25:19 2009
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Thread-Topic: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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References: <9886E5FCA6D76549A3011068483A4BD40498CFB8@S4DE8PSAAQB.mitte.t-com.de>	<1247764118.4085.24.camel@victoria-pingtel-com.us.nortel.com><1ECE0EB50388174790F9694F77522CCF1F05050C@zrc2hxm0.corp.nortel.com><4A643B95.3060800@ericsson.com><9886E5FCA6D76549A3011068483A4BD404A9C2B7@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F155AC5@zrc2hxm0.corp.nortel.com>	<CA9998CD4A020D418654FCDEF4E707DF0B1683CC@esealmw113.eemea.ericsson.se>	<1ECE0EB50388174790F9694F77522CCF1F556A65@zrc2hxm0.corp.nortel.com>	<9886E5FCA6D76549A3011068483A4BD404BFFC37@S4DE8PSAAQB.mitte.t-com.de>	<4AF37113.8030908@nostrum.com><9886E5FCA6D76549A3011068483A4BD405319E68@S4DE8PSAAQB.mitte.t-com.de><4AF7934B.7080902@cisco.com><9886E5FCA6D76549A3011068483A4BD40537238D@S4DE8PSAAQB.mitte.t-com.de><4AF!	8AE73.4050405@cisco.com>	<CA9998CD4A020D418654FCDEF4E707DF0B16864C@esealmw113.eemea.ericsson.se><744A66DF31B5B34EA8B08BBD8187A722C6423E@DEMUEXC014.nsn-intra.net> <4AFAC2DF.3090001@cisco.com>
From: <R.Jesske@telekom.de>
To: <pkyzivat@cisco.com>, <thomas.belling@nsn.com>
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Cc: calme@alcatel-lucent.com, dispatch@ietf.org, paul.while@ericsson.com
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Hello Paul,
Thank you for your comment. I have the sam opinion. Due to the fact that =
1xx is not done end to end  I have put the following sentence into the =
draft.

The appearance of the Reason header is applicable to final responses
   3xx, 4xx, 5xx and 6xx and 18x and 199 provisional responses
   [I-D.ietf-sipcore-199].

I hope that fits.=20

Best Regards

Roland

-----Urspr=FCngliche Nachricht-----
Von: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] Im =
Auftrag von Paul Kyzivat
Gesendet: Mittwoch, 11. November 2009 22:58
An: Belling, Thomas (NSN - DE/Munich)
Cc: Paul While; dispatch@ietf.org; Jesske, Roland; ext Calme,James A =
(Jim)
Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

I hate to see *anything* restricted to 183, rather than being applicable =

to all 1xx (or at least 18x) responses.

Restricting to 183 implies that there is something *special* about 183.
I have seen that erroneous assumption in the wild too - and it is=20
distressing. Anything you can do with 183 should also work with 180.

	Thanks,
	Paul

Belling, Thomas (NSN - DE/Munich) wrote:
> Dear Roland,
>=20
> we discussed the interworking of ISUP ACM and CPG causes to SIP =
reasons headers in provisional responses in 3GPP CT3 yesterday.
> The conclusion was that we will only insert a reason header in a 183 =
provisional response.
>=20
> May I therefor suggest the following wording:
>=20
>         The Reason header is applicable to 3xx, 4xx, 5xx and 6xx final =
responses
>         and 183 and 199 <reference> provisional responses.
>=20
> Further, it would be very helpfull for 3GPP if you could make the =
updated draft available very quickly as we have several dependent CRs in =
the 3GPP CTx meeting which is ongoing this week.
>=20
> Thomas
>=20
>=20
> ----------------------------------=20
> Dr. Thomas Belling=20
> 3GPP Standardisation
> Nokia Siemens Networks=20
>=20
>=20
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of ext Christer Holmberg
> Sent: Tuesday, November 10, 2009 8:12 AM
> To: Paul Kyzivat; R.Jesske@telekom.de
> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
>=20
> Hi,
>=20
> A small editorial change proposal:
>=20
> "The appearance of the Reason header is applicable to 3xx, 4xx, 5xx =
and 6xx final responses and 18x and 199 <reference> provisional =
responsess."
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Paul Kyzivat
> Sent: 10. marraskuuta 2009 3:06
> To: R.Jesske@telekom.de
> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> Roland,
>=20
> Thanks! That addresses my concern.
>=20
> 	Thanks,
> 	Paul
>=20
> R.Jesske@telekom.de wrote:
>> Hello Paul,
>> Thanks for that hint. I will change this too.=20
>> Like:
>>
>>  The appearance of the Reason header is applicable to final responses
>>        3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as =
defined within
>>        ietf-sipcore-199.txt. Also the Reason header is
>>        applicable to provisional responses18x.
>>
>> Best Regards
>>
>> Roland
>>
>> -----Urspr=FCngliche Nachricht-----
>> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Gesendet: Montag, 9. November 2009 12:58
>> An: Jesske, Roland
>> Cc: adam@nostrum.com; audet@nortel.com; dispatch@ietf.org;=20
>> gonzalo.camarillo@ericsson.com
>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>>
>>
>> R.Jesske@telekom.de wrote:
>>> Hi Adam,
>>> Thank you for your mail. So I propose the following text then:
>>>
>>> The appearance of the Reason header is applicable to final responses
>>>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as =
defined within
>>>       ietf-sipcore-199.txt. Also the Reason header is
>>>       applicable to provisional responses18x which are generated by =
an
>>>       interworking gateway from ISUP to SIP.</t>
>> I think this is ok *except* if you are going to allow one kind of UAS =

>> to use Reason in a 18x response, then it should be allowed for *any*=20
>> UAS to do so.
>>
>> For instance, suppose the gateway is replaced by a B2BUA to another=20
>> sip environment, and the request eventually arrives as a gateway. =
Then=20
>> a 18x response with a Reason header may arrive at the B2BUA and need=20
>> to be replicated on the other side.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Best Beagards
>>>
>>> Roland
>>>
>>> -----Urspr=FCngliche Nachricht-----
>>> Von: Adam Roach [mailto:adam@nostrum.com]
>>> Gesendet: Freitag, 6. November 2009 01:43
>>> An: Jesske, Roland
>>> Cc: audet@nortel.com; christer.holmberg@ericsson.com;=20
>>> gonzalo.camarillo@ericsson.com; dispatch@ietf.org
>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>
>>> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
>>>> Hi Christer and Francois,
>>>> I have added a sentence under section Overall Applicability:
>>>>
>>>>
>>>> The appearance of the Reason header is applicable to final =
responses 4xx, 5xx and 6xx and in addition for 199 Responses.
>>>>
>>>> Is this proper enough? Or do you have more in mind?
>>>>   =20
>>> Given that RFC 3398 specifies the generation of a 301 in response to =

>>> a cause code of 22 under certain circumstances, it seems pretty=20
>>> appropriate to allow the inclusion of a Q.850 cause code in =
300-class=20
>>> responses also. Or, at least, as appropriate as it would be to=20
>>> include them in 4xx, 5xx, and 6xx responses.
>>>
>>> /a
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> 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 pkyzivat@cisco.com  Wed Nov 11 06:59:05 2009
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Date: Wed, 11 Nov 2009 09:59:28 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: R.Jesske@telekom.de
References: <9886E5FCA6D76549A3011068483A4BD40498CFB8@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F05050C@zrc2hxm0.corp.nortel.com><4A643B95.3060800@ericsson.com><9886E5FCA6D76549A3011068483A4BD404A9C2B7@S4DE8PSAAQB.mitte.t-com.de>	<1ECE0EB50388174790F9694F77522CCF1F155AC5@zrc2hxm0.corp.nortel.com>	<CA9998CD4A020D418654FCDEF4E707DF0B1683CC@esealmw113.eemea.ericsson.se>	<1ECE0EB50388174790F9694F77522CCF1F556A65@zrc2hxm0.corp.nortel.com>	<9886E5FCA6D76549A3011068483A4BD404BFFC37@S4DE8PSAAQB.mitte.t-com.de>	<4AF37113.8030908@nostrum.com><9886E5FCA6D76549A3011068483A4BD405319E68@S4DE8PSAAQB.mitte.t-com.de><4AF7934B.7080902@cisco.com><9886E5FCA6D76549A3011068483A4BD40537238D@S4DE8PSAAQB.mitte.t-com.de><4AF!	8AE73.4050405@cisco.com>	<CA9998CD4A020D418654FCDEF4E707DF0B16864C@esealmw113.eemea.ericsson.se><744A66DF31B5B34EA8B08BBD8187A722C6423E@DEMUEXC014.nsn-intra.net> <4AFAC2DF.3090001@cisco.com> <9886E5FCA6D76549A3011068483A4BD405372C90@S4DE8PSAAQB.mitte.t-com.de>
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Cc: calme@alcatel-lucent.com, dispatch@ietf.org, paul.while@ericsson.com
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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R.Jesske@telekom.de wrote:
> Hello Paul,
> Thank you for your comment. I have the sam opinion. Due to the fact that 1xx is not done end to end

1xx means 101-199, excluding 100. Only 100 is hop by hop. All the 1xx 
are end to end.

> I have put the following sentence into the draft.
> 
> The appearance of the Reason header is applicable to final responses
>    3xx, 4xx, 5xx and 6xx and 18x and 199 provisional responses
>    [I-D.ietf-sipcore-199].

I'd prefer to see it just say 1xx.

	Thanks,
	Paul

> I hope that fits. 
> 
> Best Regards
> 
> Roland
> 
> -----Ursprüngliche Nachricht-----
> Von: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] Im Auftrag von Paul Kyzivat
> Gesendet: Mittwoch, 11. November 2009 22:58
> An: Belling, Thomas (NSN - DE/Munich)
> Cc: Paul While; dispatch@ietf.org; Jesske, Roland; ext Calme,James A (Jim)
> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> 
> I hate to see *anything* restricted to 183, rather than being applicable 
> to all 1xx (or at least 18x) responses.
> 
> Restricting to 183 implies that there is something *special* about 183.
> I have seen that erroneous assumption in the wild too - and it is 
> distressing. Anything you can do with 183 should also work with 180.
> 
> 	Thanks,
> 	Paul
> 
> Belling, Thomas (NSN - DE/Munich) wrote:
>> Dear Roland,
>>
>> we discussed the interworking of ISUP ACM and CPG causes to SIP reasons headers in provisional responses in 3GPP CT3 yesterday.
>> The conclusion was that we will only insert a reason header in a 183 provisional response.
>>
>> May I therefor suggest the following wording:
>>
>>         The Reason header is applicable to 3xx, 4xx, 5xx and 6xx final responses
>>         and 183 and 199 <reference> provisional responses.
>>
>> Further, it would be very helpfull for 3GPP if you could make the updated draft available very quickly as we have several dependent CRs in the 3GPP CTx meeting which is ongoing this week.
>>
>> Thomas
>>
>>
>> ---------------------------------- 
>> Dr. Thomas Belling 
>> 3GPP Standardisation
>> Nokia Siemens Networks 
>>
>>
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of ext Christer Holmberg
>> Sent: Tuesday, November 10, 2009 8:12 AM
>> To: Paul Kyzivat; R.Jesske@telekom.de
>> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
>> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>>
>> Hi,
>>
>> A small editorial change proposal:
>>
>> "The appearance of the Reason header is applicable to 3xx, 4xx, 5xx and 6xx final responses and 18x and 199 <reference> provisional responsess."
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
>> Sent: 10. marraskuuta 2009 3:06
>> To: R.Jesske@telekom.de
>> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
>> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>> Roland,
>>
>> Thanks! That addresses my concern.
>>
>> 	Thanks,
>> 	Paul
>>
>> R.Jesske@telekom.de wrote:
>>> Hello Paul,
>>> Thanks for that hint. I will change this too. 
>>> Like:
>>>
>>>  The appearance of the Reason header is applicable to final responses
>>>        3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as defined within
>>>        ietf-sipcore-199.txt. Also the Reason header is
>>>        applicable to provisional responses18x.
>>>
>>> Best Regards
>>>
>>> Roland
>>>
>>> -----Ursprüngliche Nachricht-----
>>> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>> Gesendet: Montag, 9. November 2009 12:58
>>> An: Jesske, Roland
>>> Cc: adam@nostrum.com; audet@nortel.com; dispatch@ietf.org; 
>>> gonzalo.camarillo@ericsson.com
>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>
>>>
>>>
>>> R.Jesske@telekom.de wrote:
>>>> Hi Adam,
>>>> Thank you for your mail. So I propose the following text then:
>>>>
>>>> The appearance of the Reason header is applicable to final responses
>>>>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as defined within
>>>>       ietf-sipcore-199.txt. Also the Reason header is
>>>>       applicable to provisional responses18x which are generated by an
>>>>       interworking gateway from ISUP to SIP.</t>
>>> I think this is ok *except* if you are going to allow one kind of UAS 
>>> to use Reason in a 18x response, then it should be allowed for *any* 
>>> UAS to do so.
>>>
>>> For instance, suppose the gateway is replaced by a B2BUA to another 
>>> sip environment, and the request eventually arrives as a gateway. Then 
>>> a 18x response with a Reason header may arrive at the B2BUA and need 
>>> to be replicated on the other side.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>>> Best Beagards
>>>>
>>>> Roland
>>>>
>>>> -----Ursprüngliche Nachricht-----
>>>> Von: Adam Roach [mailto:adam@nostrum.com]
>>>> Gesendet: Freitag, 6. November 2009 01:43
>>>> An: Jesske, Roland
>>>> Cc: audet@nortel.com; christer.holmberg@ericsson.com; 
>>>> gonzalo.camarillo@ericsson.com; dispatch@ietf.org
>>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>>
>>>> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
>>>>> Hi Christer and Francois,
>>>>> I have added a sentence under section Overall Applicability:
>>>>>
>>>>>
>>>>> The appearance of the Reason header is applicable to final responses 4xx, 5xx and 6xx and in addition for 199 Responses.
>>>>>
>>>>> Is this proper enough? Or do you have more in mind?
>>>>>    
>>>> Given that RFC 3398 specifies the generation of a 301 in response to 
>>>> a cause code of 22 under certain circumstances, it seems pretty 
>>>> appropriate to allow the inclusion of a Q.850 cause code in 300-class 
>>>> responses also. Or, at least, as appropriate as it would be to 
>>>> include them in 4xx, 5xx, and 6xx responses.
>>>>
>>>> /a
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> 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 brett@broadsoft.com  Wed Nov 11 07:07:41 2009
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From: Brett Tate <brett@broadsoft.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 11 Nov 2009 07:07:56 -0800
Thread-Topic: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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References: <9886E5FCA6D76549A3011068483A4BD40498CFB8@S4DE8PSAAQB.mitte.t-com.de> <1ECE0EB50388174790F9694F77522CCF1F05050C@zrc2hxm0.corp.nortel.com><4A643B95.3060800@ericsson.com><9886E5FCA6D76549A3011068483A4BD404A9C2B7@S4DE8PSAAQB.mitte.t-com.de> <1ECE0EB50388174790F9694F77522CCF1F155AC5@zrc2hxm0.corp.nortel.com> <CA9998CD4A020D418654FCDEF4E707DF0B1683CC@esealmw113.eemea.ericsson.se> <1ECE0EB50388174790F9694F77522CCF1F556A65@zrc2hxm0.corp.nortel.com> <9886E5FCA6D76549A3011068483A4BD404BFFC37@S4DE8PSAAQB.mitte.t-com.de> <4AF37113.8030908@nostrum.com><9886E5FCA6D76549A3011068483A4BD405319E68@S4DE8PSAAQB.mitte.t-com.de><4AF7934B.7080902@cisco.com><9886E5FCA6D76549A3011068483A4BD40537238D@S4DE8PSAAQB.mitte.t-com.de><4AF! 8AE73.4050405@cisco.com> <CA9998CD4A020D418654FCDEF4E707DF0B16864C@esealmw113.eemea.ericsson.se><744A66DF31B5B34EA8B08BBD8187A722C6423E@DEMUEXC014.nsn-intra.net> <4AFAC2DF.3090001@cisco.com> <9886E5FCA6D76549A3011068483A4BD405372C90@S4DE8PSAAQB.mitte.t-com.de> <4AFAD150.9070900@cisco.com>
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Although some people/documents might use 1xx when they really mean "101-199=
", rfc3261 section 7.2 indicates that 100 is included within 1xx.

"For this reason, any response with a status code between 100 and 199 is re=
ferred to as a "1xx response","

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: Wednesday, November 11, 2009 9:59 AM
> To: R.Jesske@telekom.de
> Cc: calme@alcatel-lucent.com; dispatch@ietf.org;
> paul.while@ericsson.com
> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
>=20
>=20
> R.Jesske@telekom.de wrote:
> > Hello Paul,
> > Thank you for your comment. I have the sam opinion. Due to the fact
> that 1xx is not done end to end
>=20
> 1xx means 101-199, excluding 100. Only 100 is hop by hop. All the 1xx
> are end to end.
>=20
> > I have put the following sentence into the draft.
> >
> > The appearance of the Reason header is applicable to final responses
> >    3xx, 4xx, 5xx and 6xx and 18x and 199 provisional responses
> >    [I-D.ietf-sipcore-199].
>=20
> I'd prefer to see it just say 1xx.
>=20
> 	Thanks,
> 	Paul
>=20
> > I hope that fits.
> >
> > Best Regards
> >
> > Roland
> >
> > -----Urspr=FCngliche Nachricht-----
> > Von: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] Im
> Auftrag von Paul Kyzivat
> > Gesendet: Mittwoch, 11. November 2009 22:58
> > An: Belling, Thomas (NSN - DE/Munich)
> > Cc: Paul While; dispatch@ietf.org; Jesske, Roland; ext Calme,James A
> (Jim)
> > Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> >
> > I hate to see *anything* restricted to 183, rather than being
> applicable
> > to all 1xx (or at least 18x) responses.
> >
> > Restricting to 183 implies that there is something *special* about
> 183.
> > I have seen that erroneous assumption in the wild too - and it is
> > distressing. Anything you can do with 183 should also work with 180.
> >
> > 	Thanks,
> > 	Paul
> >
> > Belling, Thomas (NSN - DE/Munich) wrote:
> >> Dear Roland,
> >>
> >> we discussed the interworking of ISUP ACM and CPG causes to SIP
> reasons headers in provisional responses in 3GPP CT3 yesterday.
> >> The conclusion was that we will only insert a reason header in a 183
> provisional response.
> >>
> >> May I therefor suggest the following wording:
> >>
> >>         The Reason header is applicable to 3xx, 4xx, 5xx and 6xx
> final responses
> >>         and 183 and 199 <reference> provisional responses.
> >>
> >> Further, it would be very helpfull for 3GPP if you could make the
> updated draft available very quickly as we have several dependent CRs
> in the 3GPP CTx meeting which is ongoing this week.
> >>
> >> Thomas
> >>
> >>
> >> ----------------------------------
> >> Dr. Thomas Belling
> >> 3GPP Standardisation
> >> Nokia Siemens Networks
> >>
> >>
> >>
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
> On Behalf Of ext Christer Holmberg
> >> Sent: Tuesday, November 10, 2009 8:12 AM
> >> To: Paul Kyzivat; R.Jesske@telekom.de
> >> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
> >> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> >>
> >>
> >> Hi,
> >>
> >> A small editorial change proposal:
> >>
> >> "The appearance of the Reason header is applicable to 3xx, 4xx, 5xx
> and 6xx final responses and 18x and 199 <reference> provisional
> responsess."
> >>
> >> Regards,
> >>
> >> Christer
> >>
> >>
> >>
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
> On Behalf Of Paul Kyzivat
> >> Sent: 10. marraskuuta 2009 3:06
> >> To: R.Jesske@telekom.de
> >> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
> >> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> >>
> >> Roland,
> >>
> >> Thanks! That addresses my concern.
> >>
> >> 	Thanks,
> >> 	Paul
> >>
> >> R.Jesske@telekom.de wrote:
> >>> Hello Paul,
> >>> Thanks for that hint. I will change this too.
> >>> Like:
> >>>
> >>>  The appearance of the Reason header is applicable to final
> responses
> >>>        3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as
> defined within
> >>>        ietf-sipcore-199.txt. Also the Reason header is
> >>>        applicable to provisional responses18x.
> >>>
> >>> Best Regards
> >>>
> >>> Roland
> >>>
> >>> -----Urspr=FCngliche Nachricht-----
> >>> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >>> Gesendet: Montag, 9. November 2009 12:58
> >>> An: Jesske, Roland
> >>> Cc: adam@nostrum.com; audet@nortel.com; dispatch@ietf.org;
> >>> gonzalo.camarillo@ericsson.com
> >>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> >>>
> >>>
> >>>
> >>> R.Jesske@telekom.de wrote:
> >>>> Hi Adam,
> >>>> Thank you for your mail. So I propose the following text then:
> >>>>
> >>>> The appearance of the Reason header is applicable to final
> responses
> >>>>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as
> defined within
> >>>>       ietf-sipcore-199.txt. Also the Reason header is
> >>>>       applicable to provisional responses18x which are generated
> by an
> >>>>       interworking gateway from ISUP to SIP.</t>
> >>> I think this is ok *except* if you are going to allow one kind of
> UAS
> >>> to use Reason in a 18x response, then it should be allowed for
> *any*
> >>> UAS to do so.
> >>>
> >>> For instance, suppose the gateway is replaced by a B2BUA to another
> >>> sip environment, and the request eventually arrives as a gateway.
> Then
> >>> a 18x response with a Reason header may arrive at the B2BUA and
> need
> >>> to be replicated on the other side.
> >>>
> >>> 	Thanks,
> >>> 	Paul
> >>>
> >>>> Best Beagards
> >>>>
> >>>> Roland
> >>>>
> >>>> -----Urspr=FCngliche Nachricht-----
> >>>> Von: Adam Roach [mailto:adam@nostrum.com]
> >>>> Gesendet: Freitag, 6. November 2009 01:43
> >>>> An: Jesske, Roland
> >>>> Cc: audet@nortel.com; christer.holmberg@ericsson.com;
> >>>> gonzalo.camarillo@ericsson.com; dispatch@ietf.org
> >>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> >>>>
> >>>> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
> >>>>> Hi Christer and Francois,
> >>>>> I have added a sentence under section Overall Applicability:
> >>>>>
> >>>>>
> >>>>> The appearance of the Reason header is applicable to final
> responses 4xx, 5xx and 6xx and in addition for 199 Responses.
> >>>>>
> >>>>> Is this proper enough? Or do you have more in mind?
> >>>>>
> >>>> Given that RFC 3398 specifies the generation of a 301 in response
> to
> >>>> a cause code of 22 under certain circumstances, it seems pretty
> >>>> appropriate to allow the inclusion of a Q.850 cause code in 300-
> class
> >>>> responses also. Or, at least, as appropriate as it would be to
> >>>> include them in 4xx, 5xx, and 6xx responses.
> >>>>
> >>>> /a
> >>>> _______________________________________________
> >>>> dispatch mailing list
> >>>> dispatch@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/dispatch
> >>>>
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From R.Jesske@telekom.de  Wed Nov 11 07:10:33 2009
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From: <R.Jesske@telekom.de>
To: <pkyzivat@cisco.com>
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Cc: calme@alcatel-lucent.com, dispatch@ietf.org, paul.while@ericsson.com
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Hi Paul,
Sorry. I was to fast with my writing an sending.=20
What I meant to write was the 100. But I dont know any other 1xx beyond =
18x and 199 that would fit. Or did I forgot one.

Or is it more the general avalibility for 1xx for future.

Best Regards

Roland=20

-----Urspr=FCngliche Nachricht-----
Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Gesendet: Mittwoch, 11. November 2009 23:59
An: Jesske, Roland
Cc: thomas.belling@nsn.com; paul.while@ericsson.com; dispatch@ietf.org; =
calme@alcatel-lucent.com
Betreff: Re: AW: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04



R.Jesske@telekom.de wrote:
> Hello Paul,
> Thank you for your comment. I have the sam opinion. Due to the fact =
that 1xx is not done end to end

1xx means 101-199, excluding 100. Only 100 is hop by hop. All the 1xx=20
are end to end.

> I have put the following sentence into the draft.
>=20
> The appearance of the Reason header is applicable to final responses
>    3xx, 4xx, 5xx and 6xx and 18x and 199 provisional responses
>    [I-D.ietf-sipcore-199].

I'd prefer to see it just say 1xx.

	Thanks,
	Paul

> I hope that fits.=20
>=20
> Best Regards
>=20
> Roland
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] Im =
Auftrag von Paul Kyzivat
> Gesendet: Mittwoch, 11. November 2009 22:58
> An: Belling, Thomas (NSN - DE/Munich)
> Cc: Paul While; dispatch@ietf.org; Jesske, Roland; ext Calme,James A =
(Jim)
> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> I hate to see *anything* restricted to 183, rather than being =
applicable=20
> to all 1xx (or at least 18x) responses.
>=20
> Restricting to 183 implies that there is something *special* about =
183.
> I have seen that erroneous assumption in the wild too - and it is=20
> distressing. Anything you can do with 183 should also work with 180.
>=20
> 	Thanks,
> 	Paul
>=20
> Belling, Thomas (NSN - DE/Munich) wrote:
>> Dear Roland,
>>
>> we discussed the interworking of ISUP ACM and CPG causes to SIP =
reasons headers in provisional responses in 3GPP CT3 yesterday.
>> The conclusion was that we will only insert a reason header in a 183 =
provisional response.
>>
>> May I therefor suggest the following wording:
>>
>>         The Reason header is applicable to 3xx, 4xx, 5xx and 6xx =
final responses
>>         and 183 and 199 <reference> provisional responses.
>>
>> Further, it would be very helpfull for 3GPP if you could make the =
updated draft available very quickly as we have several dependent CRs in =
the 3GPP CTx meeting which is ongoing this week.
>>
>> Thomas
>>
>>
>> ----------------------------------=20
>> Dr. Thomas Belling=20
>> 3GPP Standardisation
>> Nokia Siemens Networks=20
>>
>>
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of ext Christer Holmberg
>> Sent: Tuesday, November 10, 2009 8:12 AM
>> To: Paul Kyzivat; R.Jesske@telekom.de
>> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
>> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>>
>> Hi,
>>
>> A small editorial change proposal:
>>
>> "The appearance of the Reason header is applicable to 3xx, 4xx, 5xx =
and 6xx final responses and 18x and 199 <reference> provisional =
responsess."
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Paul Kyzivat
>> Sent: 10. marraskuuta 2009 3:06
>> To: R.Jesske@telekom.de
>> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
>> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>> Roland,
>>
>> Thanks! That addresses my concern.
>>
>> 	Thanks,
>> 	Paul
>>
>> R.Jesske@telekom.de wrote:
>>> Hello Paul,
>>> Thanks for that hint. I will change this too.=20
>>> Like:
>>>
>>>  The appearance of the Reason header is applicable to final =
responses
>>>        3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as =
defined within
>>>        ietf-sipcore-199.txt. Also the Reason header is
>>>        applicable to provisional responses18x.
>>>
>>> Best Regards
>>>
>>> Roland
>>>
>>> -----Urspr=FCngliche Nachricht-----
>>> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>> Gesendet: Montag, 9. November 2009 12:58
>>> An: Jesske, Roland
>>> Cc: adam@nostrum.com; audet@nortel.com; dispatch@ietf.org;=20
>>> gonzalo.camarillo@ericsson.com
>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>
>>>
>>>
>>> R.Jesske@telekom.de wrote:
>>>> Hi Adam,
>>>> Thank you for your mail. So I propose the following text then:
>>>>
>>>> The appearance of the Reason header is applicable to final =
responses
>>>>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as =
defined within
>>>>       ietf-sipcore-199.txt. Also the Reason header is
>>>>       applicable to provisional responses18x which are generated by =
an
>>>>       interworking gateway from ISUP to SIP.</t>
>>> I think this is ok *except* if you are going to allow one kind of =
UAS=20
>>> to use Reason in a 18x response, then it should be allowed for *any* =

>>> UAS to do so.
>>>
>>> For instance, suppose the gateway is replaced by a B2BUA to another=20
>>> sip environment, and the request eventually arrives as a gateway. =
Then=20
>>> a 18x response with a Reason header may arrive at the B2BUA and need =

>>> to be replicated on the other side.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>>> Best Beagards
>>>>
>>>> Roland
>>>>
>>>> -----Urspr=FCngliche Nachricht-----
>>>> Von: Adam Roach [mailto:adam@nostrum.com]
>>>> Gesendet: Freitag, 6. November 2009 01:43
>>>> An: Jesske, Roland
>>>> Cc: audet@nortel.com; christer.holmberg@ericsson.com;=20
>>>> gonzalo.camarillo@ericsson.com; dispatch@ietf.org
>>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>>
>>>> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
>>>>> Hi Christer and Francois,
>>>>> I have added a sentence under section Overall Applicability:
>>>>>
>>>>>
>>>>> The appearance of the Reason header is applicable to final =
responses 4xx, 5xx and 6xx and in addition for 199 Responses.
>>>>>
>>>>> Is this proper enough? Or do you have more in mind?
>>>>>   =20
>>>> Given that RFC 3398 specifies the generation of a 301 in response =
to=20
>>>> a cause code of 22 under certain circumstances, it seems pretty=20
>>>> appropriate to allow the inclusion of a Q.850 cause code in =
300-class=20
>>>> responses also. Or, at least, as appropriate as it would be to=20
>>>> include them in 4xx, 5xx, and 6xx responses.
>>>>
>>>> /a
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Brett Tate wrote:
> Although some people/documents might use 1xx when they really mean "101-199", rfc3261 section 7.2 indicates that 100 is included within 1xx.
> 
> "For this reason, any response with a status code between 100 and 199 is referred to as a "1xx response","

Damn! Sorry.

Ok. Then lets go for 101-199.

	Thanks,
	Paul

>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Paul Kyzivat
>> Sent: Wednesday, November 11, 2009 9:59 AM
>> To: R.Jesske@telekom.de
>> Cc: calme@alcatel-lucent.com; dispatch@ietf.org;
>> paul.while@ericsson.com
>> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>>
>>
>> R.Jesske@telekom.de wrote:
>>> Hello Paul,
>>> Thank you for your comment. I have the sam opinion. Due to the fact
>> that 1xx is not done end to end
>>
>> 1xx means 101-199, excluding 100. Only 100 is hop by hop. All the 1xx
>> are end to end.
>>
>>> I have put the following sentence into the draft.
>>>
>>> The appearance of the Reason header is applicable to final responses
>>>    3xx, 4xx, 5xx and 6xx and 18x and 199 provisional responses
>>>    [I-D.ietf-sipcore-199].
>> I'd prefer to see it just say 1xx.
>>
>> 	Thanks,
>> 	Paul
>>
>>> I hope that fits.
>>>
>>> Best Regards
>>>
>>> Roland
>>>
>>> -----Ursprüngliche Nachricht-----
>>> Von: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] Im
>> Auftrag von Paul Kyzivat
>>> Gesendet: Mittwoch, 11. November 2009 22:58
>>> An: Belling, Thomas (NSN - DE/Munich)
>>> Cc: Paul While; dispatch@ietf.org; Jesske, Roland; ext Calme,James A
>> (Jim)
>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>
>>> I hate to see *anything* restricted to 183, rather than being
>> applicable
>>> to all 1xx (or at least 18x) responses.
>>>
>>> Restricting to 183 implies that there is something *special* about
>> 183.
>>> I have seen that erroneous assumption in the wild too - and it is
>>> distressing. Anything you can do with 183 should also work with 180.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>> Belling, Thomas (NSN - DE/Munich) wrote:
>>>> Dear Roland,
>>>>
>>>> we discussed the interworking of ISUP ACM and CPG causes to SIP
>> reasons headers in provisional responses in 3GPP CT3 yesterday.
>>>> The conclusion was that we will only insert a reason header in a 183
>> provisional response.
>>>> May I therefor suggest the following wording:
>>>>
>>>>         The Reason header is applicable to 3xx, 4xx, 5xx and 6xx
>> final responses
>>>>         and 183 and 199 <reference> provisional responses.
>>>>
>>>> Further, it would be very helpfull for 3GPP if you could make the
>> updated draft available very quickly as we have several dependent CRs
>> in the 3GPP CTx meeting which is ongoing this week.
>>>> Thomas
>>>>
>>>>
>>>> ----------------------------------
>>>> Dr. Thomas Belling
>>>> 3GPP Standardisation
>>>> Nokia Siemens Networks
>>>>
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
>> On Behalf Of ext Christer Holmberg
>>>> Sent: Tuesday, November 10, 2009 8:12 AM
>>>> To: Paul Kyzivat; R.Jesske@telekom.de
>>>> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
>>>> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>>
>>>>
>>>> Hi,
>>>>
>>>> A small editorial change proposal:
>>>>
>>>> "The appearance of the Reason header is applicable to 3xx, 4xx, 5xx
>> and 6xx final responses and 18x and 199 <reference> provisional
>> responsess."
>>>> Regards,
>>>>
>>>> Christer
>>>>
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
>> On Behalf Of Paul Kyzivat
>>>> Sent: 10. marraskuuta 2009 3:06
>>>> To: R.Jesske@telekom.de
>>>> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
>>>> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>>
>>>> Roland,
>>>>
>>>> Thanks! That addresses my concern.
>>>>
>>>> 	Thanks,
>>>> 	Paul
>>>>
>>>> R.Jesske@telekom.de wrote:
>>>>> Hello Paul,
>>>>> Thanks for that hint. I will change this too.
>>>>> Like:
>>>>>
>>>>>  The appearance of the Reason header is applicable to final
>> responses
>>>>>        3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as
>> defined within
>>>>>        ietf-sipcore-199.txt. Also the Reason header is
>>>>>        applicable to provisional responses18x.
>>>>>
>>>>> Best Regards
>>>>>
>>>>> Roland
>>>>>
>>>>> -----Ursprüngliche Nachricht-----
>>>>> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>>> Gesendet: Montag, 9. November 2009 12:58
>>>>> An: Jesske, Roland
>>>>> Cc: adam@nostrum.com; audet@nortel.com; dispatch@ietf.org;
>>>>> gonzalo.camarillo@ericsson.com
>>>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>>>
>>>>>
>>>>>
>>>>> R.Jesske@telekom.de wrote:
>>>>>> Hi Adam,
>>>>>> Thank you for your mail. So I propose the following text then:
>>>>>>
>>>>>> The appearance of the Reason header is applicable to final
>> responses
>>>>>>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as
>> defined within
>>>>>>       ietf-sipcore-199.txt. Also the Reason header is
>>>>>>       applicable to provisional responses18x which are generated
>> by an
>>>>>>       interworking gateway from ISUP to SIP.</t>
>>>>> I think this is ok *except* if you are going to allow one kind of
>> UAS
>>>>> to use Reason in a 18x response, then it should be allowed for
>> *any*
>>>>> UAS to do so.
>>>>>
>>>>> For instance, suppose the gateway is replaced by a B2BUA to another
>>>>> sip environment, and the request eventually arrives as a gateway.
>> Then
>>>>> a 18x response with a Reason header may arrive at the B2BUA and
>> need
>>>>> to be replicated on the other side.
>>>>>
>>>>> 	Thanks,
>>>>> 	Paul
>>>>>
>>>>>> Best Beagards
>>>>>>
>>>>>> Roland
>>>>>>
>>>>>> -----Ursprüngliche Nachricht-----
>>>>>> Von: Adam Roach [mailto:adam@nostrum.com]
>>>>>> Gesendet: Freitag, 6. November 2009 01:43
>>>>>> An: Jesske, Roland
>>>>>> Cc: audet@nortel.com; christer.holmberg@ericsson.com;
>>>>>> gonzalo.camarillo@ericsson.com; dispatch@ietf.org
>>>>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>>>>
>>>>>> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
>>>>>>> Hi Christer and Francois,
>>>>>>> I have added a sentence under section Overall Applicability:
>>>>>>>
>>>>>>>
>>>>>>> The appearance of the Reason header is applicable to final
>> responses 4xx, 5xx and 6xx and in addition for 199 Responses.
>>>>>>> Is this proper enough? Or do you have more in mind?
>>>>>>>
>>>>>> Given that RFC 3398 specifies the generation of a 301 in response
>> to
>>>>>> a cause code of 22 under certain circumstances, it seems pretty
>>>>>> appropriate to allow the inclusion of a Q.850 cause code in 300-
>> class
>>>>>> responses also. Or, at least, as appropriate as it would be to
>>>>>> include them in 4xx, 5xx, and 6xx responses.
>>>>>>
>>>>>> /a
>>>>>> _______________________________________________
>>>>>> dispatch mailing list
>>>>>> dispatch@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>>
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
> 

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Cc: calme@alcatel-lucent.com, dispatch@ietf.org, paul.while@ericsson.com
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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R.Jesske@telekom.de wrote:
> Hi Paul,
> Sorry. I was to fast with my writing an sending. 
> What I meant to write was the 100. But I dont know any other 1xx beyond 18x and 199 that would fit. Or did I forgot one.
> 
> Or is it more the general avalibility for 1xx for future.

Enumerating which ones are meaningful isn't future proof.
Maybe this is a special case since there is a requirement to explicitly 
document which responses this header may appear in (why?),
but IMO it would be better not to second guess, and just allow them all.

My fallback from that would be all 18x and 199.

	Thanks,
	Paul

> Best Regards
> 
> Roland 
> 
> -----Ursprüngliche Nachricht-----
> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Gesendet: Mittwoch, 11. November 2009 23:59
> An: Jesske, Roland
> Cc: thomas.belling@nsn.com; paul.while@ericsson.com; dispatch@ietf.org; calme@alcatel-lucent.com
> Betreff: Re: AW: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> 
> 
> 
> R.Jesske@telekom.de wrote:
>> Hello Paul,
>> Thank you for your comment. I have the sam opinion. Due to the fact that 1xx is not done end to end
> 
> 1xx means 101-199, excluding 100. Only 100 is hop by hop. All the 1xx 
> are end to end.
> 
>> I have put the following sentence into the draft.
>>
>> The appearance of the Reason header is applicable to final responses
>>    3xx, 4xx, 5xx and 6xx and 18x and 199 provisional responses
>>    [I-D.ietf-sipcore-199].
> 
> I'd prefer to see it just say 1xx.
> 
> 	Thanks,
> 	Paul
> 
>> I hope that fits. 
>>
>> Best Regards
>>
>> Roland
>>
>> -----Ursprüngliche Nachricht-----
>> Von: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] Im Auftrag von Paul Kyzivat
>> Gesendet: Mittwoch, 11. November 2009 22:58
>> An: Belling, Thomas (NSN - DE/Munich)
>> Cc: Paul While; dispatch@ietf.org; Jesske, Roland; ext Calme,James A (Jim)
>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>> I hate to see *anything* restricted to 183, rather than being applicable 
>> to all 1xx (or at least 18x) responses.
>>
>> Restricting to 183 implies that there is something *special* about 183.
>> I have seen that erroneous assumption in the wild too - and it is 
>> distressing. Anything you can do with 183 should also work with 180.
>>
>> 	Thanks,
>> 	Paul
>>
>> Belling, Thomas (NSN - DE/Munich) wrote:
>>> Dear Roland,
>>>
>>> we discussed the interworking of ISUP ACM and CPG causes to SIP reasons headers in provisional responses in 3GPP CT3 yesterday.
>>> The conclusion was that we will only insert a reason header in a 183 provisional response.
>>>
>>> May I therefor suggest the following wording:
>>>
>>>         The Reason header is applicable to 3xx, 4xx, 5xx and 6xx final responses
>>>         and 183 and 199 <reference> provisional responses.
>>>
>>> Further, it would be very helpfull for 3GPP if you could make the updated draft available very quickly as we have several dependent CRs in the 3GPP CTx meeting which is ongoing this week.
>>>
>>> Thomas
>>>
>>>
>>> ---------------------------------- 
>>> Dr. Thomas Belling 
>>> 3GPP Standardisation
>>> Nokia Siemens Networks 
>>>
>>>
>>>
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of ext Christer Holmberg
>>> Sent: Tuesday, November 10, 2009 8:12 AM
>>> To: Paul Kyzivat; R.Jesske@telekom.de
>>> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
>>> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>
>>>
>>> Hi,
>>>
>>> A small editorial change proposal:
>>>
>>> "The appearance of the Reason header is applicable to 3xx, 4xx, 5xx and 6xx final responses and 18x and 199 <reference> provisional responsess."
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>>
>>>
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
>>> Sent: 10. marraskuuta 2009 3:06
>>> To: R.Jesske@telekom.de
>>> Cc: audet@nortel.com; dispatch@ietf.org; Gonzalo Camarillo
>>> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>
>>> Roland,
>>>
>>> Thanks! That addresses my concern.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>> R.Jesske@telekom.de wrote:
>>>> Hello Paul,
>>>> Thanks for that hint. I will change this too. 
>>>> Like:
>>>>
>>>>  The appearance of the Reason header is applicable to final responses
>>>>        3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as defined within
>>>>        ietf-sipcore-199.txt. Also the Reason header is
>>>>        applicable to provisional responses18x.
>>>>
>>>> Best Regards
>>>>
>>>> Roland
>>>>
>>>> -----Ursprüngliche Nachricht-----
>>>> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>> Gesendet: Montag, 9. November 2009 12:58
>>>> An: Jesske, Roland
>>>> Cc: adam@nostrum.com; audet@nortel.com; dispatch@ietf.org; 
>>>> gonzalo.camarillo@ericsson.com
>>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>>
>>>>
>>>>
>>>> R.Jesske@telekom.de wrote:
>>>>> Hi Adam,
>>>>> Thank you for your mail. So I propose the following text then:
>>>>>
>>>>> The appearance of the Reason header is applicable to final responses
>>>>>       3xx, 4xx, 5xx and 6xx and in addition for 199 Responses as defined within
>>>>>       ietf-sipcore-199.txt. Also the Reason header is
>>>>>       applicable to provisional responses18x which are generated by an
>>>>>       interworking gateway from ISUP to SIP.</t>
>>>> I think this is ok *except* if you are going to allow one kind of UAS 
>>>> to use Reason in a 18x response, then it should be allowed for *any* 
>>>> UAS to do so.
>>>>
>>>> For instance, suppose the gateway is replaced by a B2BUA to another 
>>>> sip environment, and the request eventually arrives as a gateway. Then 
>>>> a 18x response with a Reason header may arrive at the B2BUA and need 
>>>> to be replicated on the other side.
>>>>
>>>> 	Thanks,
>>>> 	Paul
>>>>
>>>>> Best Beagards
>>>>>
>>>>> Roland
>>>>>
>>>>> -----Ursprüngliche Nachricht-----
>>>>> Von: Adam Roach [mailto:adam@nostrum.com]
>>>>> Gesendet: Freitag, 6. November 2009 01:43
>>>>> An: Jesske, Roland
>>>>> Cc: audet@nortel.com; christer.holmberg@ericsson.com; 
>>>>> gonzalo.camarillo@ericsson.com; dispatch@ietf.org
>>>>> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>>>
>>>>> On 8/6/09 7:28 PM, R.Jesske@telekom.de wrote:
>>>>>> Hi Christer and Francois,
>>>>>> I have added a sentence under section Overall Applicability:
>>>>>>
>>>>>>
>>>>>> The appearance of the Reason header is applicable to final responses 4xx, 5xx and 6xx and in addition for 199 Responses.
>>>>>>
>>>>>> Is this proper enough? Or do you have more in mind?
>>>>>>    
>>>>> Given that RFC 3398 specifies the generation of a 301 in response to 
>>>>> a cause code of 22 under certain circumstances, it seems pretty 
>>>>> appropriate to allow the inclusion of a Q.850 cause code in 300-class 
>>>>> responses also. Or, at least, as appropriate as it would be to 
>>>>> include them in 4xx, 5xx, and 6xx responses.
>>>>>
>>>>> /a
>>>>> _______________________________________________
>>>>> dispatch mailing list
>>>>> dispatch@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>> _______________________________________________
>>> 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 mary.barnes@nortel.com  Wed Nov 11 07:41:04 2009
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The charts for this session have been uploaded to the meeting repository =
under the DISPATCH WG - entitled "DREGS Adhoc".
https://datatracker.ietf.org/meeting/76/materials.html

Regards,
Mary.=20


-----Original Message-----
From: dispatch-bounces@ietf.org on behalf of Eric Burger
Sent: Wed 11/11/2009 2:26 AM
To: dispatch@ietf.org
Subject: [dispatch] DREGS ad hoc
=20
The DREGS ad hoc will be in Orchid East, 11.45 - 12.40 JT tomorrow =20
(Thursday).

We do need scribes and jabber scribes.  Please respond directly to me =20
to volunteer.  Thanks.
_______________________________________________
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<A =
HREF=3D"https://datatracker.ietf.org/meeting/76/materials.html">https://d=
atatracker.ietf.org/meeting/76/materials.html</A><BR>
<BR>
Regards,<BR>
Mary.<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: dispatch-bounces@ietf.org on behalf of Eric Burger<BR>
Sent: Wed 11/11/2009 2:26 AM<BR>
To: dispatch@ietf.org<BR>
Subject: [dispatch] DREGS ad hoc<BR>
<BR>
The DREGS ad hoc will be in Orchid East, 11.45 - 12.40 JT =
tomorrow&nbsp;<BR>
(Thursday).<BR>
<BR>
We do need scribes and jabber scribes.&nbsp; Please respond directly to =
me&nbsp;<BR>
to volunteer.&nbsp; Thanks.<BR>
_______________________________________________<BR>
dispatch mailing list<BR>
dispatch@ietf.org<BR>
<A =
HREF=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>
<BR>
<BR>
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From eburger@standardstrack.com  Wed Nov 11 13:09:47 2009
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We still need Scribes!!!

On Nov 11, 2009, at 7:41 PM, Mary Barnes wrote:

>
> Orchard -> Orchid
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org on behalf of Eric Burger
> Sent: Wed 11/11/2009 2:26 AM
> To: dispatch@ietf.org
> Subject: [dispatch] DREGS ad hoc
>
> The DREGS ad hoc will be in Orchard East, 11.45 - 12.40 JT tomorrow
> (Thursday).
>
> We do need scribes and jabber scribes.  Please respond directly to me
> to volunteer.  Thanks.
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>


From david.middleton@nsn.com  Wed Nov 11 13:13:17 2009
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Thread-Topic: [dispatch] To-URI in domain-registration
Thread-Index: AcpZue9KRbQOCEeXRx6AR/epLJfHpAJV/yZw
References: <E6C2E8958BA59A4FB960963D475F7AC31A0A7C9F34@mail>	<09B7DBFE70A9E24BBB21689DAD3A06141B09D891@zcarhxm1.corp.nortel.com><9AAEDF491EF7CA48AB587781B8F5D7C6024FEFD3@srvxchg3.cablelabs.com>	<024101ca528e$fbd0a8b0$f371fa10$@us> <159B27ECF9D16E40BD7ABD5778522B0103792C9C@USCHEXC006.nsn-intra.net> <4AEB78AD.5070608@cisco.com>
From: "Middleton, David (NSN - US/Boca Raton)" <david.middleton@nsn.com>
To: "ext Paul Kyzivat" <pkyzivat@cisco.com>
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Subject: Re: [dispatch] To-URI in domain-registration
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Paul,

Sorry for the delay in responding.

I think you are on target as far as describing many (perhaps most) SP / =
PBX interconnect arrangements.  However, the functionality of having the =
SP route calls per user is a real-world scenario which proves there is =
merit for having the capability.

Also see inline.

David



-----Original Message-----
From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Friday, October 30, 2009 7:37 PM
To: Middleton, David (NSN - US/Boca Raton)
Cc: dispatch@ietf.org
Subject: Re: [dispatch] To-URI in domain-registration

David,

IIUC, you are proposing a case such as:

- a customer of sp.net has domain cust.sp.net
- the customer has two PBX systems, east and west.
- you propose that they register, respectively
   To: east@cust.sp.net, and
   To: west@cust.sp.net.
- then numbers assigned to the customer are routed
   something like:
   1-212-555-88xx -> contact from east@cust.sp.net
   1-408-555-88xx -> contact from west@cust.sp.net
- also, routing for names be something like:
   bob@cust.sp.net   -> contact from east@cust.sp.net
   carol@cust.sp.net -> "
   ted@cust.sp.net   -> contact from west@cust.sp.net
   alice@cust.sp.net -> "
   others...

Is that right?
<DM> Yes

You then can't describe that as setting up routing paths for=20
cust.sp.net. You are instead describing some entirely different routing=20
algorithm based on the user part.
<DM>  Yes, it requires a local policy to determine the location of the =
users.

Subdomains east.cust.sp.net and west.cust.sp.net solve this problem in=20
large part - at least for the numbers.

You clearly don't find that acceptable for the names. If those names=20
were exposed to end users I wouldn't find it acceptable either. (I want=20
to be pkyzivat@cisco.com, not pkyzivat@east.cisco.com.)
<DM> Agreed

The question is whether this is the best way to solve that problem.

I think that solution won't be very acceptable to an SP, or to the=20
customer of that SP. Having the SP provision every one of your users=20
just isn't going to be popular with anyone. (Unless the SP gets a big=20
enough per-user fee.)
<DM> I agree but if the parties involved reach an acceptable agreement, =
the technology should support it IMO.

Its not quite as much of a problem with numbers, because they are likely =

to be managed in blocks. And the numbers are really owned/managed by the =

SP anyway so they are forced to do something about them.

This is also a problem that is not unique to this SP interconnect=20
mechanism. It is pretty much the same problem in any big company with a=20
lot of sites, even if they have a private network connecting all the =
sites.

What you need in this case is a proxy at the company domain level=20
(cust.sp.net in this case perhaps) that has the roster of all usernames=20
and a mapping to a subdomain - mapping bob@cust.sp.net to=20
bob@east.cust.sp.net.
<DM> Yes but some enterprise customers do not want to do this and are =
able to convince the SP to do this.

Then, the company has one server registering To:cust.sp.net to provide=20
this mapping, and other servers registering To:east.cust.sp.net, ...

	Thanks,
	Paul


Middleton, David (NSN - US/Boca Raton) wrote:
> Some colleagues have pointed out an aspect that I had not considered =
concerning the domain registration To: user part.  There are cases where =
an Enterprise has multiple PBXs that register and calls should be routed =
to one of the PBXs based on which user is called.  In other words, =
instead of the Enterprise with multiple PBXs appearing as one logical =
PBX, the Enterprise customer wants the service provider to take =
responsibility for distributing the calls to the specific PBX that hosts =
the called user.
>=20
> While this problem could be solved with a local policy to choose one =
of several contacts it is not as clean an approach as being able to =
treat each registration as a separate entity with it's own contact(s).  =
While one could propose that these be treated separately by making =
different sub-domains, the Enterprise most assuredly will not want to =
publish this sub-domain on business cards, etc. when using e-mail style =
addresses.  Besides, it would cause the address to change if the user =
moves from one PBX to another as might happen with a physical move.
>=20
> Another aspect to consider is that the Enterprise user, in this =
scenario, is only reachable via one registered "To: user@host".  So not =
having separate registrations would result in a more complex definition =
and implementation of things like (a.) is there an unexpired binding and =
(b.) picking appropriate contacts by q-value.
>=20
> I think this makes a strong case for having a user part in the To: =
header and treating each unique user@host as a separate registration =E0 =
la RFC 3264.
>=20
> In summary, I support Option 1) as well.
>=20
> - David
>=20
> =20
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of ext Richard Shockey
> Sent: Wednesday, October 21, 2009 4:42 PM
> To: dispatch@ietf.org
> Subject: Re: [dispatch] To-URI in domain-registration
>=20
> +1 to Option 1) as well
>=20
>>  -----Original Message-----
>>  From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] =
On
>>  Behalf Of David Hancock
>>  Sent: Wednesday, October 21, 2009 4:05 PM
>>  To: Brian Lindsay; Hadriel Kaplan; dispatch@ietf.org
>>  Subject: Re: [dispatch] To-URI in domain-registration
>> =20
>> =20
>>  Hadriel - I'd prefer option 1) as well.
>> =20
>>  David
>> =20
>>  > -----Original Message-----
>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
>>  On
>>  > Behalf Of Brian Lindsay
>>  > Sent: Wednesday, October 21, 2009 1:53 PM
>>  > To: Hadriel Kaplan; dispatch@ietf.org
>>  > Subject: Re: [dispatch] To-URI in domain-registration
>>  >
>>  >  Hi Hadriel,
>>  >
>>  >     Per my comment yesterday, I'd prefer (1).
>>  >
>>  >     Thanks.
>>  >
>>  > -----Original Message-----
>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
>>  On
>>  > Behalf Of Hadriel Kaplan
>>  > Sent: Wednesday, October 21, 2009 3:43 PM
>>  > To: dispatch@ietf.org
>>  > Subject: [dispatch] To-URI in domain-registration
>>  >
>>  > Howdy,
>>  > I'd like to get a consensus call on the question of what the =
To-URI
>>  > should be for the REGISTER request of a domain registration, =
before
>>  > finishing the next rev of the draft.  This will affect quite a bit
>>  of
>>  > text in the draft (e.g., examples), and I think that the decision
>>  about
>>  > that will also affect whether we need more than a =
Require:option-tag
>>  as
>>  > well.
>>  >
>>  > The choices right now are:
>>  > 1) Mandate the To-URI be in the syntactical form of an AoR,
>>  including
>>  > the user@ portion, the same as the From-URI.
>>  > 	Pros: closer to what current proprietary and other-SDO
>>  > mechanisms do, on the wire.  Therefore, presumably less of a =
change
>>  and
>>  > less barrier to adoption.
>>  > 2) Mandate the To-URI be a domain only (no user@), while the From-
>>  URI
>>  > remains the "AoR" of the PBX registering.
>>  > 	Pros: semantically accurate (you're registering a domain, not an
>>  > AoR), and may obviate the need for a new header or param to =
indicate
>>  > this new mechanism is being used.
>>  >
>>  > Can people please indicate which they'd prefer?
>>  >
>>  > Thanks!
>>  > -hadriel
>>  > _______________________________________________
>>  > dispatch mailing list
>>  > dispatch@ietf.org
>>  > https://www.ietf.org/mailman/listinfo/dispatch
>>  > _______________________________________________
>>  > dispatch mailing list
>>  > dispatch@ietf.org
>>  > https://www.ietf.org/mailman/listinfo/dispatch
>>  _______________________________________________
>>  dispatch mailing list
>>  dispatch@ietf.org
>>  https://www.ietf.org/mailman/listinfo/dispatch
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From spencer@wonderhamster.org  Wed Nov 11 14:24:50 2009
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From: "Spencer Dawkins" <spencer@wonderhamster.org>
To: "Eric Burger" <eburger@standardstrack.com>, <dispatch@ietf.org>
References: <CB843946-3E0A-4552-BF5C-506B4DCAE983@standardstrack.com><F592E36A5C943E4E91F25880D05AD1140CC8E20E@zrc2hxm0.corp.nortel.com> <BAFF6291-2E01-4F00-8254-E7226F48A603@standardstrack.com>
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Subject: Re: [dispatch] DREGS ad hoc
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And at least one of The Usual Suspects is helping to present!!! so can't 
reasonably be drafted at the beginning of the session without this turning 
into a DoS attack!!!

:-)


> We still need Scribes!!!

Spencer!!! 


From pkyzivat@cisco.com  Wed Nov 11 14:34:36 2009
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] To-URI in domain-registration
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Middleton, David (NSN - US/Boca Raton) wrote:
> Paul,
> 
> Sorry for the delay in responding.
> 
> I think you are on target as far as describing many (perhaps most) SP / PBX interconnect arrangements.  However, the functionality of having the SP route calls per user is a real-world scenario which proves there is merit for having the capability.

This could still be dealt with by having the pbxes register to separate 
sub-domains and the SP knowing (via provisioning) which things to route 
to each.

So instead of having:

foo-east-pbx register to east@foo.sp.com
foo-west-pbx register to west@foo.sp.com

you could simply have

foo-east-pbx register to east.foo.sp.com
foo-west-pbx register to west.foo.sp.com

Then the question of how requests addressed to xyz@foo.sp.com are mapped 
to something@east.foo.sp.com or something@west.foo.sp.com is simply out 
of scope of this document.

	Thanks,
	Paul


> Also see inline.
> 
> David
> 
> 
> 
> -----Original Message-----
> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Friday, October 30, 2009 7:37 PM
> To: Middleton, David (NSN - US/Boca Raton)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] To-URI in domain-registration
> 
> David,
> 
> IIUC, you are proposing a case such as:
> 
> - a customer of sp.net has domain cust.sp.net
> - the customer has two PBX systems, east and west.
> - you propose that they register, respectively
>    To: east@cust.sp.net, and
>    To: west@cust.sp.net.
> - then numbers assigned to the customer are routed
>    something like:
>    1-212-555-88xx -> contact from east@cust.sp.net
>    1-408-555-88xx -> contact from west@cust.sp.net
> - also, routing for names be something like:
>    bob@cust.sp.net   -> contact from east@cust.sp.net
>    carol@cust.sp.net -> "
>    ted@cust.sp.net   -> contact from west@cust.sp.net
>    alice@cust.sp.net -> "
>    others...
> 
> Is that right?
> <DM> Yes
> 
> You then can't describe that as setting up routing paths for 
> cust.sp.net. You are instead describing some entirely different routing 
> algorithm based on the user part.
> <DM>  Yes, it requires a local policy to determine the location of the users.
> 
> Subdomains east.cust.sp.net and west.cust.sp.net solve this problem in 
> large part - at least for the numbers.
> 
> You clearly don't find that acceptable for the names. If those names 
> were exposed to end users I wouldn't find it acceptable either. (I want 
> to be pkyzivat@cisco.com, not pkyzivat@east.cisco.com.)
> <DM> Agreed
> 
> The question is whether this is the best way to solve that problem.
> 
> I think that solution won't be very acceptable to an SP, or to the 
> customer of that SP. Having the SP provision every one of your users 
> just isn't going to be popular with anyone. (Unless the SP gets a big 
> enough per-user fee.)
> <DM> I agree but if the parties involved reach an acceptable agreement, the technology should support it IMO.
> 
> Its not quite as much of a problem with numbers, because they are likely 
> to be managed in blocks. And the numbers are really owned/managed by the 
> SP anyway so they are forced to do something about them.
> 
> This is also a problem that is not unique to this SP interconnect 
> mechanism. It is pretty much the same problem in any big company with a 
> lot of sites, even if they have a private network connecting all the sites.
> 
> What you need in this case is a proxy at the company domain level 
> (cust.sp.net in this case perhaps) that has the roster of all usernames 
> and a mapping to a subdomain - mapping bob@cust.sp.net to 
> bob@east.cust.sp.net.
> <DM> Yes but some enterprise customers do not want to do this and are able to convince the SP to do this.
> 
> Then, the company has one server registering To:cust.sp.net to provide 
> this mapping, and other servers registering To:east.cust.sp.net, ...
> 
> 	Thanks,
> 	Paul
> 
> 
> Middleton, David (NSN - US/Boca Raton) wrote:
>> Some colleagues have pointed out an aspect that I had not considered concerning the domain registration To: user part.  There are cases where an Enterprise has multiple PBXs that register and calls should be routed to one of the PBXs based on which user is called.  In other words, instead of the Enterprise with multiple PBXs appearing as one logical PBX, the Enterprise customer wants the service provider to take responsibility for distributing the calls to the specific PBX that hosts the called user.
>>
>> While this problem could be solved with a local policy to choose one of several contacts it is not as clean an approach as being able to treat each registration as a separate entity with it's own contact(s).  While one could propose that these be treated separately by making different sub-domains, the Enterprise most assuredly will not want to publish this sub-domain on business cards, etc. when using e-mail style addresses.  Besides, it would cause the address to change if the user moves from one PBX to another as might happen with a physical move.
>>
>> Another aspect to consider is that the Enterprise user, in this scenario, is only reachable via one registered "To: user@host".  So not having separate registrations would result in a more complex definition and implementation of things like (a.) is there an unexpired binding and (b.) picking appropriate contacts by q-value.
>>
>> I think this makes a strong case for having a user part in the To: header and treating each unique user@host as a separate registration à la RFC 3264.
>>
>> In summary, I support Option 1) as well.
>>
>> - David
>>
>>  
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of ext Richard Shockey
>> Sent: Wednesday, October 21, 2009 4:42 PM
>> To: dispatch@ietf.org
>> Subject: Re: [dispatch] To-URI in domain-registration
>>
>> +1 to Option 1) as well
>>
>>>  -----Original Message-----
>>>  From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>>>  Behalf Of David Hancock
>>>  Sent: Wednesday, October 21, 2009 4:05 PM
>>>  To: Brian Lindsay; Hadriel Kaplan; dispatch@ietf.org
>>>  Subject: Re: [dispatch] To-URI in domain-registration
>>>  
>>>  
>>>  Hadriel - I'd prefer option 1) as well.
>>>  
>>>  David
>>>  
>>>  > -----Original Message-----
>>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
>>>  On
>>>  > Behalf Of Brian Lindsay
>>>  > Sent: Wednesday, October 21, 2009 1:53 PM
>>>  > To: Hadriel Kaplan; dispatch@ietf.org
>>>  > Subject: Re: [dispatch] To-URI in domain-registration
>>>  >
>>>  >  Hi Hadriel,
>>>  >
>>>  >     Per my comment yesterday, I'd prefer (1).
>>>  >
>>>  >     Thanks.
>>>  >
>>>  > -----Original Message-----
>>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
>>>  On
>>>  > Behalf Of Hadriel Kaplan
>>>  > Sent: Wednesday, October 21, 2009 3:43 PM
>>>  > To: dispatch@ietf.org
>>>  > Subject: [dispatch] To-URI in domain-registration
>>>  >
>>>  > Howdy,
>>>  > I'd like to get a consensus call on the question of what the To-URI
>>>  > should be for the REGISTER request of a domain registration, before
>>>  > finishing the next rev of the draft.  This will affect quite a bit
>>>  of
>>>  > text in the draft (e.g., examples), and I think that the decision
>>>  about
>>>  > that will also affect whether we need more than a Require:option-tag
>>>  as
>>>  > well.
>>>  >
>>>  > The choices right now are:
>>>  > 1) Mandate the To-URI be in the syntactical form of an AoR,
>>>  including
>>>  > the user@ portion, the same as the From-URI.
>>>  > 	Pros: closer to what current proprietary and other-SDO
>>>  > mechanisms do, on the wire.  Therefore, presumably less of a change
>>>  and
>>>  > less barrier to adoption.
>>>  > 2) Mandate the To-URI be a domain only (no user@), while the From-
>>>  URI
>>>  > remains the "AoR" of the PBX registering.
>>>  > 	Pros: semantically accurate (you're registering a domain, not an
>>>  > AoR), and may obviate the need for a new header or param to indicate
>>>  > this new mechanism is being used.
>>>  >
>>>  > Can people please indicate which they'd prefer?
>>>  >
>>>  > Thanks!
>>>  > -hadriel
>>>  > _______________________________________________
>>>  > dispatch mailing list
>>>  > dispatch@ietf.org
>>>  > https://www.ietf.org/mailman/listinfo/dispatch
>>>  > _______________________________________________
>>>  > dispatch mailing list
>>>  > dispatch@ietf.org
>>>  > https://www.ietf.org/mailman/listinfo/dispatch
>>>  _______________________________________________
>>>  dispatch mailing list
>>>  dispatch@ietf.org
>>>  https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
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>> _______________________________________________
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>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> 

From stpeter@stpeter.im  Wed Nov 11 16:52:03 2009
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Date: Thu, 12 Nov 2009 09:52:25 +0900
From: Peter Saint-Andre <stpeter@stpeter.im>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Agenda for SIP/XMPP co-existence ad-hoc meeting on Friday Nov 13
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On 11/10/09 11:14 AM, Simo.Veikkolainen@nokia.com wrote:
> Hello,
> 
> Here is the proposed agenda for the DISPATCH WG ad-hoc session on
> SIP/XMPP co-existence scheduled for Friday Nov 13, 2009, at
> 13:00-15:00 at room Cattleya West. The ad-hoc session will be chaired
> by Gonzalo Camarillo.

I'm really sorry that I won't be there.

> 1. Motivation and objectives of SIP/XMPP co-existence (10 min)
> 
> 2. Overview of solution space (15 min) - baseline proposal
> http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01
> 
> 3. Technical issues discussed so far (15 min) -
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00845.html
> 
> 4. Discussion on charter proposal (30 min) -
> http://www.ietf.org/mail-archive/web/dispatch/current/msg00560.html
> 
> 5. Conclusion and next steps

My overriding concern is that we keep it simple (SIMPLE?) and focus on
some small practical tasks first, to solve "pain points" among existing
implementations and deployments. Clearly, many use cases and scenarios
might be interesting eventually, but first let's do things that might
have an immediate impact.

That said, I would be happy to put some time into the
draft-saintandre-sip-xmpp-* document series if we feel that's needed.

Peter

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



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From mary.barnes@nortel.com  Wed Nov 11 22:55:38 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: <dispatch@ietf.org>
Subject: [dispatch] Meeting notes: DREGS Adhoc
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Hi all,

The notes from the DREGS Adhoc held today from 11:45-12:50 are available =
on the DISPATCH WG supplemental webpage:
http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/DREGS-adhoc-note=
s-shida.txt
http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/DREGS Adhoc =
Meeting notes-jmce.doc=09


Regards,
Mary.=20

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<P><FONT SIZE=3D2>Hi all,<BR>
<BR>
The notes from the DREGS Adhoc held today from 11:45-12:50 are available =
on the DISPATCH WG supplemental webpage:<BR>
<A =
HREF=3D"http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/DREGS-ad=
hoc-notes-shida.txt">http://www.softarmor.com/dispatch/IETF76/Meeting%20n=
otes/DREGS-adhoc-notes-shida.txt</A><BR>
<A =
HREF=3D"http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/DREGS">h=
ttp://www.softarmor.com/dispatch/IETF76/Meeting%20notes/DREGS</A> Adhoc =
Meeting notes-jmce.doc&nbsp;&nbsp;&nbsp;&nbsp;<BR>
<BR>
<BR>
Regards,<BR>
Mary. </FONT>
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From eburger@standardstrack.com  Thu Nov 12 00:14:15 2009
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DREGS Adhoc

Thursday 12 November 2009, 11:30 =96 1:00

Chairs: Eric Burger and Mary Barnes

Scribes: Shida Schubert and Jim McEachern



John Elwell =96 Charter Discussion.

http://www.ietf.org/proceedings/09nov/slides/dispatch-6.ppt



Jon Peterson reacted to the John=92s statement that =93decided we needed =
a =20
new WG=94 and said that he had suggested it perhaps should be in =20
DRINKS.  Was concerned about the presumption of what needs to be done =20=

to address the open issue.



Cullen:  the purpose of dispatch is to recommend what should be done.  =20=

So we could recommend it go to DRINKS or wherever.  Let=92s see what we =20=

recommend.



Clarified that the proposal was that this should be looked at, not =20
that a WG was needed.

Trying to standardize a uniform way for PBXs to register their =20
domain.  Today everyone does it their own way, even though they are =20
all very similar.

Proposing to use REGISTER with an added header for requesting domain =20
registration.  They are proposing this approach because it builds on =20
what is already widely deployed.



Adam Roach:  Are we really proposing to write the use of REGISTER in =20
the charter?  Answer: Yes.  Adam stated that he was not prepared to =20
have that discussion today because he thought it would be discussed in =20=

the WG.  This feels like =93let=92s ratify this draft=94 rather than =
=93let=92s =20
work on this problem=94.



Spencer Dawkins: the intent is not to ratify existing draft.

Note:  In spite of Spencer=92s statement, the ensuing discussion made it =
=20
very clear that for at least some of the authors, the intent was in =20
fact very much to ratify this draft, or something very near it.  =20
Apparently the initial charter draft did not mention REGISTER and it =20
was included because SIP Forum participants explicitly asked for it to =20=

be added.



Eric: the subject of whether or not REGISTER should be included in the =20=

charter has been discussed on the list, but there is not consensus



Someone stated that REGISTER was proposed because =93that is what is =20
implemented=94 and we need to align with those implementations to be =20
relevant.  Jon countered that earlier it had been stated that the =20
entire reason for this is that there are multiple ways this is =20
implemented and that we need a standard to get interoperability.  So =20
if it is being done multiple ways today, the argument that REGISTER =20
needs to be in the charter to align with deployed equipment, simply =20
does not fly.



Cable Labs supported the use of REGISTER because that is what they use =20=

in their specifications.



Jim McEachern & Adam Roach both pointed out that even if you remove =20
the word REGISTER from the charter, it does not mean that the eventual =20=

solution will have REGISTER.  The bulk of the objections (dare I say =20
the allergic reaction?) is to the fact that the solution (REGISTER) is =20=

being specified in the charter, when we should be simply defining the =20=

problem we are trying to solve.



Anwar Siddiqi Asked about the definition of small =96 medium business, =20=

and said that this will not apply to large enterprises since they will =20=

never give their register information to the SP.  Therefore why not =20
make it optional.



Markus Isomaki, Nokia:  What does this imply about the addresses of =20
the terminals behind the PBX?  Answer: UA registers with the PBX, but =20=

this work deals with the PBX registering with the SP.  It is designed =20=

to help the SP work out where to send requests to =93example.com=94.



Jon Peterson:  Charter scope question.  Is the charter focused on the =20=

specific problem on the slide, or the more general problem? (Note: I =20
think this was referring to the bullets specifying =93a header field for =
=20
requesting domain registration=94 and =93a SIP option tag to detect non-=20=

support of this mechanism=94.)   Answer: The more general one.  (Note:  =20=

I think this meant the general problem of PBXs registering=85)



Jon Peterson:  The only way this charter will get approved is by =20
taking REGISTER out of the draft.  Lots of nods to that.



Spencer:  If this looks like tweaking existing implementations, it =20
will get more interest than if it looked like a new method.  =20
Preference is to be up front about this.



Cullen (channeling Hadriel):  If Hadriel were here he would insist it =20=

must be REGISTER or it will never be deployed.



Cullen: would the possible solution space now include dynamic dns?  =20
The only reason to specify a solution in the charter is to explicitly =20=

exclude alternate classes of solution =96 isn=92t it?



Chris Stanley =96 cable labs.  We need to deploy this for businesses and =
=20
need to move quickly.  (And REGISTER is the mechanism we use.)



Adam Roach:  how far are we going to go down this path of specifying =20
the answer in the charter?  If we allow it for REGISTER for this WG, =20
what other things are we going to do it for?



Cullen as AD:  He was the one who earlier encouraged them to be very =20
specific to see if they could get consensus on the approach to shorten =20=

the process, even though he suspected the proposal would get the =20
reaction that it is currently getting.  However, if the group cannot =20
get consensus to specify the solution in the charter, then the only =20
approach is to remove it.  This is clearly the case.



Hum:  Should we leave the charter exactly as it is?  Result: Moderate =20=

hum



Hum:  Should we leave the charter largely as is,  but remove the =20
solution (REGISTER) from the charter.  Result: Strong hum.



Discussed a vote on =93Is there a critical mass in the IETF to work on =20=

this problem?=94, but instead decided on the following wording.



Hum:  Does anyone think that we don=92t have the critical mass and the =20=

expertise to tackle this.  Result:  no one objected.  So consensus for =20=

this.



Hum: is the IETF the place to work on this?  Answer: yes



Note:  The following happened after the meeting had officially closed, =20=

but is included here because it pointed toward a possible way forward.

Jon: Discussion of the precedent setting risks of just adopting an =20
existing solution.  One way forward may be to think about this as a =20
telephone number problem rather than a domain problem, which would =20
make it a much easier problem.  If we can scope it that way then it =20
will make it a much quicker problem to solve.

Alan Johnston:  All the SIP Forum discussion is in fact about =20
telephone numbers, so restricting it to telephone numbers is what we =20
are looking for



Cullen:  Recognize that we do need to find ways to work faster.  =93If =20=

you send me a charter by midnight tonight, I will have it on the =20
agenda for the next IESG meeting=94



Meeting officially closed.



Since we had time, we had an open discussion of how to make progress=85



Discussion of what would be realistic dates for this?  The discussion =20=

indicated that March and June might be more believable dates.



Cullen:  Recognize that we do need to find ways to work faster.  =93If =20=

you send me a charter by midnight tonight, I will have it on the =20
agenda for the next IESG meeting=94



Robert Sparks:  Recipe for success in SDOs cooperating is having a =20
significant overlap of common participants working on the problem.  He =20=

is concerned that the overlap is not enough.



Spencer:  Last SIP Forum meeting had 12 people and 7 of them were also =20=

in the IETF.  Is that enough?



Jon: Discussion of the precedent setting risks of just adopting an =20
existing solution.  One way forward may be to think about this as a =20
telephone number problem rather than a domain problem, then it is a =20
much easier problem.  If we can scope it that way then it will make it =20=

a much easier problem to solve.



Alan Johnston:  All the SIP Forum discussion is in fact about =20
telephone numbers, so restricting it to telephone numbers would be =20
acceptable.=20
  =20=

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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><!--StartFragment--><p =
class=3D"MsoNormal">DREGS Adhoc</p><p class=3D"MsoNormal">Thursday 12 =
November 2009, 11:30 =96 1:00</p><p class=3D"MsoNormal">Chairs: Eric =
Burger and Mary Barnes</p><p class=3D"MsoNormal">Scribes: Shida Schubert =
and Jim McEachern</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal">John Elwell =96 Charter Discussion.</p><p =
class=3D"MsoNormal"><a =
href=3D"http://www.ietf.org/proceedings/09nov/slides/dispatch-6.ppt">http:=
//www.ietf.org/proceedings/09nov/slides/dispatch-6.ppt</a>
</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Jon Peterson</b> reacted to the John=92s =
statement that =93decided
we needed a new WG=94 and said that he had suggested it perhaps should =
be in
DRINKS.<span style=3D"mso-spacerun: yes">&nbsp; </span>Was concerned =
about the
presumption of what needs to be done to address the open issue.</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Cullen</b>:<span style=3D"mso-spacerun: =
yes">&nbsp;
</span>the purpose of dispatch is to recommend what should be done.<span =
style=3D"mso-spacerun: yes">&nbsp; </span>So we could recommend it go to =
DRINKS
or wherever.<span style=3D"mso-spacerun: yes">&nbsp; </span>Let=92s see =
what we
recommend.</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal">Clarified that the proposal was that this should be =
looked
at, not that a WG was needed.</p><p class=3D"MsoNormal">Trying to =
standardize a uniform way for PBXs to register
their domain.<span style=3D"mso-spacerun: yes">&nbsp; </span>Today =
everyone does
it their own way, even though they are all very similar.</p><p =
class=3D"MsoNormal">Proposing to use REGISTER with an added header for
requesting domain registration.<span style=3D"mso-spacerun: yes">&nbsp;
</span>They are proposing this approach because it builds on what is =
already
widely deployed.</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Adam Roach</b>:<span style=3D"mso-spacerun: =
yes">&nbsp;
</span>Are we really proposing to write the use of REGISTER in the
charter?<span style=3D"mso-spacerun: yes">&nbsp; </span><b>Answer</b>: =
Yes.<span style=3D"mso-spacerun: yes">&nbsp; </span>Adam stated that he =
was not prepared to
have that discussion today because he thought it would be discussed in =
the
WG.<span style=3D"mso-spacerun: yes">&nbsp; </span>This feels like =
=93let=92s ratify
this draft=94 rather than =93let=92s work on this problem=94.</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p class=3D"MsoNormal"><b>Spencer=
 Dawkins</b>: the intent is not to ratify existing
draft.</p><p class=3D"MsoNormal" =
style=3D"margin-left:.5in"><b><i>Note</i></b><i>:<span =
style=3D"mso-spacerun: yes">&nbsp; </span>In spite of Spencer=92s =
statement, the
ensuing discussion made it very clear that for at least some of the =
authors,
the intent was in fact very much to ratify this draft, or something very =
near
it.<span style=3D"mso-spacerun: yes">&nbsp; </span>Apparently the =
initial charter
draft did not mention REGISTER and it was included because SIP Forum
participants explicitly asked for it to be added.<o:p></o:p></i></p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Eric</b>: the subject of whether or not REGISTER =
should
be included in the charter has been discussed on the list, but there is =
not
consensus</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal">Someone stated that REGISTER was proposed because =
=93that is
what is implemented=94 and we need to align with those implementations =
to be
relevant.<span style=3D"mso-spacerun: yes">&nbsp; </span><b>Jon</b> =
countered
that earlier it had been stated that the entire reason for this is that =
there
are multiple ways this is implemented and that we need a standard to get
interoperability.<span style=3D"mso-spacerun: yes">&nbsp; </span>So if =
it is
being done multiple ways today, the argument that REGISTER needs to be =
in the
charter to align with deployed equipment, simply does not fly.</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p class=3D"MsoNormal"><b>Cable =
Labs</b> supported the use of REGISTER because that
is what they use in their specifications.</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p class=3D"MsoNormal"><b>Jim =
McEachern &amp; Adam Roach</b> both pointed out that
even if you remove the word REGISTER from the charter, it does not mean =
that
the eventual solution will have REGISTER.<span style=3D"mso-spacerun: =
yes">&nbsp;
</span>The bulk of the objections (dare I say the allergic reaction?) is =
to the
fact that the solution (REGISTER) is being specified in the charter, =
when we
should be simply defining the problem we are trying to solve.</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p class=3D"MsoNormal"><b>Anwar =
Siddiqi </b>Asked about the definition of small =96
medium business, and said that this will not apply to large enterprises =
since
they will never give their register information to the SP.<span =
style=3D"mso-spacerun: yes">&nbsp; </span>Therefore why not make it =
optional.</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Markus Isomaki</b>, Nokia:<span =
style=3D"mso-spacerun:
yes">&nbsp; </span>What does this imply about the addresses of the =
terminals
behind the PBX?<span style=3D"mso-spacerun: yes">&nbsp; =
</span><b>Answer</b>: UA
registers with the PBX, but this work deals with the PBX registering =
with the
SP.<span style=3D"mso-spacerun: yes">&nbsp; </span>It is designed to =
help the SP
work out where to send requests to =93example.com=94.</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p class=3D"MsoNormal"><b>Jon =
Peterson</b>:<span style=3D"mso-spacerun: yes">&nbsp;
</span>Charter scope question.<span style=3D"mso-spacerun: yes">&nbsp; =
</span>Is
the charter focused on the specific problem on the slide, or the more =
general
problem? (<i>Note: I think this was referring to the bullets specifying =
=93a
header field for requesting domain registration=94 and =93a SIP option =
tag to detect
non-support of this mechanism=94</i>.) <span style=3D"mso-spacerun:
yes">&nbsp;&nbsp;</span><b>Answer</b>: The more general one.<span =
style=3D"mso-spacerun: yes">&nbsp; </span>(<i>Note:<span =
style=3D"mso-spacerun:
yes">&nbsp; </span>I think this meant the general problem of PBXs =
registering=85</i>)</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Jon Peterson</b>:<span style=3D"mso-spacerun: =
yes">&nbsp;
</span>The only way this charter will get approved is by taking REGISTER =
out of
the draft.<span style=3D"mso-spacerun: yes">&nbsp; </span>Lots of nods =
to that.</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Spencer</b>:<span style=3D"mso-spacerun: =
yes">&nbsp;
</span>If this looks like tweaking existing implementations, it will get =
more
interest than if it looked like a new method.<span style=3D"mso-spacerun:
yes">&nbsp; </span>Preference is to be up front about this.</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p class=3D"MsoNormal"><b>Cullen =
(channeling Hadriel)</b>:<span style=3D"mso-spacerun: yes">&nbsp; =
</span>If Hadriel were here he would insist it
must be REGISTER or it will never be deployed.</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Cullen</b>: would the possible solution space now =
include
dynamic dns?<span style=3D"mso-spacerun: yes">&nbsp; </span>The only =
reason to specify
a solution in the charter is to explicitly exclude alternate classes of
solution =96 isn=92t it?</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p=
 class=3D"MsoNormal"><b>Chris Stanley</b> =96 cable labs.<span =
style=3D"mso-spacerun:
yes">&nbsp; </span>We need to deploy this for businesses and need to =
move
quickly.<span style=3D"mso-spacerun: yes">&nbsp; </span>(And REGISTER is =
the
mechanism we use.)</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Adam Roach</b>:<span style=3D"mso-spacerun: =
yes">&nbsp;
</span>how far are we going to go down this path of specifying the =
answer in
the charter?<span style=3D"mso-spacerun: yes">&nbsp; </span>If we allow =
it for
REGISTER for this WG, what other things are we going to do it for?</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p class=3D"MsoNormal"><b>Cullen =
as AD</b>:<span style=3D"mso-spacerun: yes">&nbsp;
</span>He was the one who earlier encouraged them to be very specific to =
see if
they could get consensus on the approach to shorten the process, even =
though he
suspected the proposal would get the reaction that it is currently =
getting.<span style=3D"mso-spacerun: yes">&nbsp; </span>However, if the =
group cannot get
consensus to specify the solution in the charter, then the only approach =
is to
remove it. <span style=3D"mso-spacerun: yes">&nbsp;</span>This is =
clearly the
case.</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Hum</b>:<span style=3D"mso-spacerun: yes">&nbsp; =
</span>Should
we leave the charter exactly as it is?<span style=3D"mso-spacerun: =
yes">&nbsp;
</span><b>Result</b>: Moderate hum</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Hum</b>:<span style=3D"mso-spacerun: yes">&nbsp; =
</span>Should
we leave the charter largely as is,<span style=3D"mso-spacerun: =
yes">&nbsp;
</span>but remove the solution (REGISTER) from the charter.<span =
style=3D"mso-spacerun: yes">&nbsp; </span><b>Result</b>: Strong =
hum.</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal">Discussed a vote on =93Is there a critical mass in =
the IETF to
work on this problem?=94, but instead decided on the following =
wording.</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Hum</b>:<span style=3D"mso-spacerun: yes">&nbsp;
</span>Does anyone think that we don=92t have the critical mass and the =
expertise
to tackle this.<span style=3D"mso-spacerun: yes">&nbsp; =
</span><b>Result</b>:<span style=3D"mso-spacerun: yes">&nbsp; </span>no =
one objected.<span style=3D"mso-spacerun: yes">&nbsp; </span>So =
consensus for this.</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Hum</b>: is the IETF the place to work on =
this?<span style=3D"mso-spacerun: yes">&nbsp; </span><b>Answer</b>: =
yes</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><i>Note:<span style=3D"mso-spacerun: yes">&nbsp; =
</span>The
following happened after the meeting had officially closed, but is =
included
here because it pointed toward a possible way =
forward.<o:p></o:p></i></p><p class=3D"MsoNormal" =
style=3D"margin-left:9.0pt"><b><i>Jon</i></b><i>: Discussion
of the precedent setting risks of just adopting an existing =
solution.<span style=3D"mso-spacerun: yes">&nbsp; </span>One way forward =
may be to think about
this as a telephone number problem rather than a domain problem, which =
would
make it a much easier problem.<span style=3D"mso-spacerun: yes">&nbsp; =
</span>If
we can scope it that way then it will make it a much quicker problem to =
solve.<o:p></o:p></i></p><p class=3D"MsoNormal" =
style=3D"margin-left:9.0pt"><b><i>Alan Johnston</i></b><i>:<span =
style=3D"mso-spacerun: yes">&nbsp; </span>All the SIP Forum discussion =
is in fact
about telephone numbers, so restricting it to telephone numbers is what =
we are
looking for<o:p></o:p></i></p><p class=3D"MsoNormal" =
style=3D"margin-left:9.0pt"><i><o:p>&nbsp;</o:p></i></p><p =
class=3D"MsoNormal" =
style=3D"margin-left:9.0pt"><b><i>Cullen</i></b><i>:<span =
style=3D"mso-spacerun: yes">&nbsp; </span>Recognize that we do need to =
find ways
to work faster.<span style=3D"mso-spacerun: yes">&nbsp; </span>=93If you =
send me a
charter by midnight tonight, I will have it on the agenda for the next =
IESG
meeting=94<o:p></o:p></i></p><p class=3D"MsoNormal" =
style=3D"margin-left:9.0pt"><i><o:p>&nbsp;</o:p></i></p><p =
class=3D"MsoNormal"><b><u>Meeting officially =
closed.<o:p></o:p></u></b></p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p class=3D"MsoNormal"><b>Since =
we had time, we had an open discussion of how to
make progress=85<o:p></o:p></b></p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p class=3D"MsoNormal">Discussion=
 of what would be realistic dates for this?<span style=3D"mso-spacerun: =
yes">&nbsp; </span>The discussion indicated that March and
June might be more believable dates.</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Cullen</b>:<span style=3D"mso-spacerun: =
yes">&nbsp;
</span>Recognize that we do need to find ways to work faster.<span =
style=3D"mso-spacerun: yes">&nbsp; </span>=93If you send me a charter by =
midnight
tonight, I will have it on the agenda for the next IESG meeting=94</p><p =
class=3D"MsoNormal"><b><o:p>&nbsp;</o:p></b></p><p =
class=3D"MsoNormal"><b>Robert Sparks</b>:<span style=3D"mso-spacerun: =
yes">&nbsp;
</span>Recipe for success in SDOs cooperating is having a significant =
overlap
of common participants working on the problem.<span style=3D"mso-spacerun:=

yes">&nbsp; </span>He is concerned that the overlap is not enough.</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Spencer</b>:<span style=3D"mso-spacerun: =
yes">&nbsp;
</span>Last SIP Forum meeting had 12 people and 7 of them were also in =
the
IETF.<span style=3D"mso-spacerun: yes">&nbsp; </span>Is that =
enough?</p><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p =
class=3D"MsoNormal"><b>Jon</b>: Discussion of the precedent setting =
risks of
just adopting an existing solution.<span style=3D"mso-spacerun: =
yes">&nbsp;
</span>One way forward may be to think about this as a telephone number =
problem
rather than a domain problem, then it is a much easier problem.<span =
style=3D"mso-spacerun: yes">&nbsp; </span>If we can scope it that way =
then it
will make it a much easier problem to solve.</p><p =
class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p class=3D"MsoNormal"><b>Alan =
Johnston</b>:<span style=3D"mso-spacerun: yes">&nbsp;
</span>All the SIP Forum discussion is in fact about telephone numbers, =
so
restricting it to telephone numbers would be acceptable. <span =
style=3D"mso-spacerun: yes">&nbsp;&nbsp;</span></p>

<!--EndFragment-->


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From spencer@wonderhamster.org  Thu Nov 12 00:25:59 2009
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From: "Spencer Dawkins" <spencer@wonderhamster.org>
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Subject: Re: [dispatch] DREGS Ad Hoc minutes
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Hi, Shida and Jim,

Thanks for taking notes and publishing them quickly.

What *I* thought happened (very close to the end of the meeting), was =
that=20

- Robert expressed his concern that SIP Forum participation and IETF =
participation didn't overlap, and this was a problem because that's the =
IESG's experience with inter-SDO coordination is that nothing good =
happens unless there's sufficient overlap,

- Spencer said something like "we had about 12 (and John said 14) =
participants at the last SIPconnect face-to-face, and about 7 were =
regular IETF attendees, is that enough overlap?", and=20

- Robert said he didn't think so.

Others can, of course, confirm (and the audio will probably be available =
in a small number of days), but it's going to be real important to tease =
a way forward on this, and I don't want the notes to give the impression =
that Robert was convinced when I asked my question.

Thanks,

Spencer
  ----- Original Message -----=20
  From: Eric Burger=20
  To: dispatch@ietf.org=20
  Sent: Thursday, November 12, 2009 5:14 PM
  Subject: [dispatch] DREGS Ad Hoc minutes


  DREGS Adhoc

  Thursday 12 November 2009, 11:30 =96 1:00

  Chairs: Eric Burger and Mary Barnes

  Scribes: Shida Schubert and Jim McEachern



  John Elwell =96 Charter Discussion.

  http://www.ietf.org/proceedings/09nov/slides/dispatch-6.ppt=20



  Jon Peterson reacted to the John=92s statement that =93decided we =
needed a new WG=94 and said that he had suggested it perhaps should be =
in DRINKS.  Was concerned about the presumption of what needs to be done =
to address the open issue.



  Cullen:  the purpose of dispatch is to recommend what should be done.  =
So we could recommend it go to DRINKS or wherever.  Let=92s see what we =
recommend.



  Clarified that the proposal was that this should be looked at, not =
that a WG was needed.

  Trying to standardize a uniform way for PBXs to register their domain. =
 Today everyone does it their own way, even though they are all very =
similar.

  Proposing to use REGISTER with an added header for requesting domain =
registration.  They are proposing this approach because it builds on =
what is already widely deployed.



  Adam Roach:  Are we really proposing to write the use of REGISTER in =
the charter?  Answer: Yes.  Adam stated that he was not prepared to have =
that discussion today because he thought it would be discussed in the =
WG.  This feels like =93let=92s ratify this draft=94 rather than =
=93let=92s work on this problem=94.



  Spencer Dawkins: the intent is not to ratify existing draft.

  Note:  In spite of Spencer=92s statement, the ensuing discussion made =
it very clear that for at least some of the authors, the intent was in =
fact very much to ratify this draft, or something very near it.  =
Apparently the initial charter draft did not mention REGISTER and it was =
included because SIP Forum participants explicitly asked for it to be =
added.



  Eric: the subject of whether or not REGISTER should be included in the =
charter has been discussed on the list, but there is not consensus



  Someone stated that REGISTER was proposed because =93that is what is =
implemented=94 and we need to align with those implementations to be =
relevant.  Jon countered that earlier it had been stated that the entire =
reason for this is that there are multiple ways this is implemented and =
that we need a standard to get interoperability.  So if it is being done =
multiple ways today, the argument that REGISTER needs to be in the =
charter to align with deployed equipment, simply does not fly.



  Cable Labs supported the use of REGISTER because that is what they use =
in their specifications.



  Jim McEachern & Adam Roach both pointed out that even if you remove =
the word REGISTER from the charter, it does not mean that the eventual =
solution will have REGISTER.  The bulk of the objections (dare I say the =
allergic reaction?) is to the fact that the solution (REGISTER) is being =
specified in the charter, when we should be simply defining the problem =
we are trying to solve.



  Anwar Siddiqi Asked about the definition of small =96 medium business, =
and said that this will not apply to large enterprises since they will =
never give their register information to the SP.  Therefore why not make =
it optional.



  Markus Isomaki, Nokia:  What does this imply about the addresses of =
the terminals behind the PBX?  Answer: UA registers with the PBX, but =
this work deals with the PBX registering with the SP.  It is designed to =
help the SP work out where to send requests to =93example.com=94.



  Jon Peterson:  Charter scope question.  Is the charter focused on the =
specific problem on the slide, or the more general problem? (Note: I =
think this was referring to the bullets specifying =93a header field for =
requesting domain registration=94 and =93a SIP option tag to detect =
non-support of this mechanism=94.)   Answer: The more general one.  =
(Note:  I think this meant the general problem of PBXs registering=85)



  Jon Peterson:  The only way this charter will get approved is by =
taking REGISTER out of the draft.  Lots of nods to that.



  Spencer:  If this looks like tweaking existing implementations, it =
will get more interest than if it looked like a new method.  Preference =
is to be up front about this.



  Cullen (channeling Hadriel):  If Hadriel were here he would insist it =
must be REGISTER or it will never be deployed.



  Cullen: would the possible solution space now include dynamic dns?  =
The only reason to specify a solution in the charter is to explicitly =
exclude alternate classes of solution =96 isn=92t it?



  Chris Stanley =96 cable labs.  We need to deploy this for businesses =
and need to move quickly.  (And REGISTER is the mechanism we use.)



  Adam Roach:  how far are we going to go down this path of specifying =
the answer in the charter?  If we allow it for REGISTER for this WG, =
what other things are we going to do it for?



  Cullen as AD:  He was the one who earlier encouraged them to be very =
specific to see if they could get consensus on the approach to shorten =
the process, even though he suspected the proposal would get the =
reaction that it is currently getting.  However, if the group cannot get =
consensus to specify the solution in the charter, then the only approach =
is to remove it.  This is clearly the case.



  Hum:  Should we leave the charter exactly as it is?  Result: Moderate =
hum



  Hum:  Should we leave the charter largely as is,  but remove the =
solution (REGISTER) from the charter.  Result: Strong hum.



  Discussed a vote on =93Is there a critical mass in the IETF to work on =
this problem?=94, but instead decided on the following wording.



  Hum:  Does anyone think that we don=92t have the critical mass and the =
expertise to tackle this.  Result:  no one objected.  So consensus for =
this.



  Hum: is the IETF the place to work on this?  Answer: yes



  Note:  The following happened after the meeting had officially closed, =
but is included here because it pointed toward a possible way forward.

  Jon: Discussion of the precedent setting risks of just adopting an =
existing solution.  One way forward may be to think about this as a =
telephone number problem rather than a domain problem, which would make =
it a much easier problem.  If we can scope it that way then it will make =
it a much quicker problem to solve.

  Alan Johnston:  All the SIP Forum discussion is in fact about =
telephone numbers, so restricting it to telephone numbers is what we are =
looking for



  Cullen:  Recognize that we do need to find ways to work faster.  =93If =
you send me a charter by midnight tonight, I will have it on the agenda =
for the next IESG meeting=94



  Meeting officially closed.



  Since we had time, we had an open discussion of how to make =
progress=85



  Discussion of what would be realistic dates for this?  The discussion =
indicated that March and June might be more believable dates.



  Cullen:  Recognize that we do need to find ways to work faster.  =93If =
you send me a charter by midnight tonight, I will have it on the agenda =
for the next IESG meeting=94



  Robert Sparks:  Recipe for success in SDOs cooperating is having a =
significant overlap of common participants working on the problem.  He =
is concerned that the overlap is not enough.



  Spencer:  Last SIP Forum meeting had 12 people and 7 of them were also =
in the IETF.  Is that enough?



  Jon: Discussion of the precedent setting risks of just adopting an =
existing solution.  One way forward may be to think about this as a =
telephone number problem rather than a domain problem, then it is a much =
easier problem.  If we can scope it that way then it will make it a much =
easier problem to solve.



  Alan Johnston:  All the SIP Forum discussion is in fact about =
telephone numbers, so restricting it to telephone numbers would be =
acceptable.  =20



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


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

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<DIV><FONT size=3D2>Hi, Shida and Jim,</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Thanks for taking notes and publishing them=20
quickly.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>What *I* thought happened (very close to the end of =
the=20
meeting), was that </FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>- Robert expressed his concern that SIP Forum =
participation=20
and IETF participation didn't overlap, and this was a problem because =
that's the=20
IESG's experience with inter-SDO coordination is that nothing good =
happens=20
unless there's sufficient overlap,</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>- Spencer said something like "we had about 12 (and =
John said=20
14) participants at the last SIPconnect face-to-face, and about 7 were =
regular=20
IETF attendees, is that enough overlap?", and </FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>- Robert said he didn't think so.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Others can, of course, confirm (and the audio will =
probably be=20
available in a small number of days), but i</FONT><FONT size=3D2>t's =
going to be=20
real important to tease a way forward on this, and I don't want the =
notes to=20
give the impression that Robert was convinced when I asked my=20
question.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Thanks,</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=3Deburger@standardstrack.com=20
  href=3D"mailto:eburger@standardstrack.com">Eric Burger</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3Ddispatch@ietf.org=20
  href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Thursday, November 12, =
2009 5:14=20
  PM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [dispatch] DREGS Ad =
Hoc=20
  minutes</DIV>
  <DIV><BR></DIV><!--StartFragment-->
  <P class=3DMsoNormal>DREGS Adhoc</P>
  <P class=3DMsoNormal>Thursday 12 November 2009, 11:30 =96 1:00</P>
  <P class=3DMsoNormal>Chairs: Eric Burger and Mary Barnes</P>
  <P class=3DMsoNormal>Scribes: Shida Schubert and Jim McEachern</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal>John Elwell =96 Charter Discussion.</P>
  <P class=3DMsoNormal><A=20
  =
href=3D"http://www.ietf.org/proceedings/09nov/slides/dispatch-6.ppt">http=
://www.ietf.org/proceedings/09nov/slides/dispatch-6.ppt</A>=20
  </P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Jon Peterson</B> reacted to the John=92s =
statement that=20
  =93decided we needed a new WG=94 and said that he had suggested it =
perhaps should=20
  be in DRINKS.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>Was =
concerned about=20
  the presumption of what needs to be done to address the open =
issue.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Cullen</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>the purpose of dispatch is to recommend what should be =
done.<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>So we could recommend it go =
to DRINKS=20
  or wherever.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>Let=92s =
see what we=20
  recommend.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal>Clarified that the proposal was that this should =
be looked=20
  at, not that a WG was needed.</P>
  <P class=3DMsoNormal>Trying to standardize a uniform way for PBXs to =
register=20
  their domain.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>Today =
everyone does=20
  it their own way, even though they are all very similar.</P>
  <P class=3DMsoNormal>Proposing to use REGISTER with an added header =
for=20
  requesting domain registration.<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>They are proposing this approach because it builds on what is =
already=20
  widely deployed.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Adam Roach</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>Are we really proposing to write the use of REGISTER in the=20
  charter?<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN><B>Answer</B>: =
Yes.<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>Adam stated that he was not =
prepared=20
  to have that discussion today because he thought it would be discussed =
in the=20
  WG.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>This feels like =
=93let=92s ratify=20
  this draft=94 rather than =93let=92s work on this problem=94.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Spencer Dawkins</B>: the intent is not to =
ratify=20
  existing draft.</P>
  <P style=3D"MARGIN-LEFT: 0.5in" =
class=3DMsoNormal><B><I>Note</I></B><I>:<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>In spite of Spencer=92s =
statement, the=20
  ensuing discussion made it very clear that for at least some of the =
authors,=20
  the intent was in fact very much to ratify this draft, or something =
very near=20
  it.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>Apparently the =
initial=20
  charter draft did not mention REGISTER and it was included because SIP =
Forum=20
  participants explicitly asked for it to be added.<O:P></O:P></I></P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Eric</B>: the subject of whether or not =
REGISTER should=20
  be included in the charter has been discussed on the list, but there =
is not=20
  consensus</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal>Someone stated that REGISTER was proposed because =
=93that is=20
  what is implemented=94 and we need to align with those implementations =
to be=20
  relevant.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN><B>Jon</B> =
countered=20
  that earlier it had been stated that the entire reason for this is =
that there=20
  are multiple ways this is implemented and that we need a standard to =
get=20
  interoperability.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>So if =
it is=20
  being done multiple ways today, the argument that REGISTER needs to be =
in the=20
  charter to align with deployed equipment, simply does not fly.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Cable Labs</B> supported the use of REGISTER =
because=20
  that is what they use in their specifications.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Jim McEachern &amp; Adam Roach</B> both =
pointed out that=20
  even if you remove the word REGISTER from the charter, it does not =
mean that=20
  the eventual solution will have REGISTER.<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>The bulk of the objections =
(dare I say=20
  the allergic reaction?) is to the fact that the solution (REGISTER) is =
being=20
  specified in the charter, when we should be simply defining the =
problem we are=20
  trying to solve.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Anwar Siddiqi </B>Asked about the definition =
of small =96=20
  medium business, and said that this will not apply to large =
enterprises since=20
  they will never give their register information to the SP.<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>Therefore why not make it=20
optional.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Markus Isomaki</B>, Nokia:<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>What does this imply about =
the=20
  addresses of the terminals behind the PBX?<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN><B>Answer</B>: UA registers =
with the=20
  PBX, but this work deals with the PBX registering with the SP.<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>It is designed to help the =
SP work out=20
  where to send requests to =93example.com=94.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Jon Peterson</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>Charter scope question.<SPAN style=3D"mso-spacerun: yes">&nbsp; =
</SPAN>Is=20
  the charter focused on the specific problem on the slide, or the more =
general=20
  problem? (<I>Note: I think this was referring to the bullets =
specifying =93a=20
  header field for requesting domain registration=94 and =93a SIP option =
tag to=20
  detect non-support of this mechanism=94</I>.) <SPAN=20
  style=3D"mso-spacerun: yes">&nbsp;&nbsp;</SPAN><B>Answer</B>: The more =
general=20
  one.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>(<I>Note:<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>I think this meant the =
general problem=20
  of PBXs registering=85</I>)</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Jon Peterson</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>The only way this charter will get approved is by taking =
REGISTER out=20
  of the draft.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>Lots of =
nods to=20
  that.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Spencer</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>If this looks like tweaking existing implementations, it will =
get more=20
  interest than if it looked like a new method.<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>Preference is to be up front =
about=20
  this.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Cullen (channeling Hadriel)</B>:<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>If Hadriel were here he =
would insist=20
  it must be REGISTER or it will never be deployed.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Cullen</B>: would the possible solution space =
now=20
  include dynamic dns?<SPAN style=3D"mso-spacerun: yes">&nbsp; =
</SPAN>The only=20
  reason to specify a solution in the charter is to explicitly exclude =
alternate=20
  classes of solution =96 isn=92t it?</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Chris Stanley</B> =96 cable labs.<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>We need to deploy this for =
businesses=20
  and need to move quickly.<SPAN style=3D"mso-spacerun: yes">&nbsp; =
</SPAN>(And=20
  REGISTER is the mechanism we use.)</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Adam Roach</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>how far are we going to go down this path of specifying the =
answer in=20
  the charter?<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>If we =
allow it for=20
  REGISTER for this WG, what other things are we going to do it for?</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Cullen as AD</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>He was the one who earlier encouraged them to be very specific =
to see=20
  if they could get consensus on the approach to shorten the process, =
even=20
  though he suspected the proposal would get the reaction that it is =
currently=20
  getting.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>However, if =
the group=20
  cannot get consensus to specify the solution in the charter, then the =
only=20
  approach is to remove it. <SPAN style=3D"mso-spacerun: =
yes">&nbsp;</SPAN>This is=20
  clearly the case.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Hum</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>Should we leave the charter exactly as it is?<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN><B>Result</B>: Moderate =
hum</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Hum</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>Should we leave the charter largely as is,<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>but remove the solution =
(REGISTER)=20
  from the charter.<SPAN style=3D"mso-spacerun: yes">&nbsp; =
</SPAN><B>Result</B>:=20
  Strong hum.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal>Discussed a vote on =93Is there a critical mass =
in the IETF=20
  to work on this problem?=94, but instead decided on the following =
wording.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Hum</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>Does anyone think that we don=92t have the critical mass and =
the=20
  expertise to tackle this.<SPAN style=3D"mso-spacerun: yes">&nbsp;=20
  </SPAN><B>Result</B>:<SPAN style=3D"mso-spacerun: yes">&nbsp; =
</SPAN>no one=20
  objected.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>So consensus =
for=20
  this.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Hum</B>: is the IETF the place to work on =
this?<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN><B>Answer</B>: yes</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><I>Note:<SPAN style=3D"mso-spacerun: yes">&nbsp; =
</SPAN>The=20
  following happened after the meeting had officially closed, but is =
included=20
  here because it pointed toward a possible way =
forward.<O:P></O:P></I></P>
  <P style=3D"MARGIN-LEFT: 9pt" class=3DMsoNormal><B><I>Jon</I></B><I>: =
Discussion=20
  of the precedent setting risks of just adopting an existing =
solution.<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>One way forward may be to =
think about=20
  this as a telephone number problem rather than a domain problem, which =
would=20
  make it a much easier problem.<SPAN style=3D"mso-spacerun: yes">&nbsp; =
</SPAN>If=20
  we can scope it that way then it will make it a much quicker problem =
to=20
  solve.<O:P></O:P></I></P>
  <P style=3D"MARGIN-LEFT: 9pt" class=3DMsoNormal><B><I>Alan=20
  Johnston</I></B><I>:<SPAN style=3D"mso-spacerun: yes">&nbsp; =
</SPAN>All the SIP=20
  Forum discussion is in fact about telephone numbers, so restricting it =
to=20
  telephone numbers is what we are looking for<O:P></O:P></I></P>
  <P style=3D"MARGIN-LEFT: 9pt" class=3DMsoNormal><I><O:P></O:P></I></P>
  <P style=3D"MARGIN-LEFT: 9pt" =
class=3DMsoNormal><B><I>Cullen</I></B><I>:<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>Recognize that we do need to =
find ways=20
  to work faster.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>=93If =
you send me a=20
  charter by midnight tonight, I will have it on the agenda for the next =
IESG=20
  meeting=94<O:P></O:P></I></P>
  <P style=3D"MARGIN-LEFT: 9pt" class=3DMsoNormal><I><O:P></O:P></I></P>
  <P class=3DMsoNormal><B><U>Meeting officially =
closed.<O:P></O:P></U></B></P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Since we had time, we had an open discussion =
of how to=20
  make progress=85<O:P></O:P></B></P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal>Discussion of what would be realistic dates for =
this?<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>The discussion indicated =
that March=20
  and June might be more believable dates.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Cullen</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>Recognize that we do need to find ways to work faster.<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>=93If you send me a charter =
by midnight=20
  tonight, I will have it on the agenda for the next IESG meeting=94</P>
  <P class=3DMsoNormal><B><O:P></O:P></B></P>
  <P class=3DMsoNormal><B>Robert Sparks</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>Recipe for success in SDOs cooperating is having a significant =
overlap=20
  of common participants working on the problem.<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>He is concerned that the =
overlap is=20
  not enough.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Spencer</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>Last SIP Forum meeting had 12 people and 7 of them were also in =
the=20
  IETF.<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>Is that =
enough?</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Jon</B>: Discussion of the precedent setting =
risks of=20
  just adopting an existing solution.<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>One way forward may be to think about this as a telephone =
number=20
  problem rather than a domain problem, then it is a much easier =
problem.<SPAN=20
  style=3D"mso-spacerun: yes">&nbsp; </SPAN>If we can scope it that way =
then it=20
  will make it a much easier problem to solve.</P>
  <P class=3DMsoNormal><O:P></O:P></P>
  <P class=3DMsoNormal><B>Alan Johnston</B>:<SPAN style=3D"mso-spacerun: =
yes">&nbsp;=20
  </SPAN>All the SIP Forum discussion is in fact about telephone =
numbers, so=20
  restricting it to telephone numbers would be acceptable. <SPAN=20
  style=3D"mso-spacerun: yes">&nbsp;&nbsp;</SPAN></P><!--EndFragment-->
  <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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Very drafty. A bunch of errors already found. Will fix RSN. Comments  
welcome.

From david.middleton@nsn.com  Thu Nov 12 04:12:43 2009
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From: "Middleton, David (NSN - US/Boca Raton)" <david.middleton@nsn.com>
To: "ext Paul Kyzivat" <pkyzivat@cisco.com>
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I agree this is possible but having the SP cross-reference domains is =
not in the spirit of domain registration from my perspective.


-----Original Message-----
From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Wednesday, November 11, 2009 5:35 PM
To: Middleton, David (NSN - US/Boca Raton)
Cc: dispatch@ietf.org
Subject: Re: [dispatch] To-URI in domain-registration



Middleton, David (NSN - US/Boca Raton) wrote:
> Paul,
>=20
> Sorry for the delay in responding.
>=20
> I think you are on target as far as describing many (perhaps most) SP =
/ PBX interconnect arrangements.  However, the functionality of having =
the SP route calls per user is a real-world scenario which proves there =
is merit for having the capability.

This could still be dealt with by having the pbxes register to separate=20
sub-domains and the SP knowing (via provisioning) which things to route=20
to each.

So instead of having:

foo-east-pbx register to east@foo.sp.com
foo-west-pbx register to west@foo.sp.com

you could simply have

foo-east-pbx register to east.foo.sp.com
foo-west-pbx register to west.foo.sp.com

Then the question of how requests addressed to xyz@foo.sp.com are mapped =

to something@east.foo.sp.com or something@west.foo.sp.com is simply out=20
of scope of this document.

	Thanks,
	Paul


> Also see inline.
>=20
> David
>=20
>=20
>=20
> -----Original Message-----
> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Friday, October 30, 2009 7:37 PM
> To: Middleton, David (NSN - US/Boca Raton)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] To-URI in domain-registration
>=20
> David,
>=20
> IIUC, you are proposing a case such as:
>=20
> - a customer of sp.net has domain cust.sp.net
> - the customer has two PBX systems, east and west.
> - you propose that they register, respectively
>    To: east@cust.sp.net, and
>    To: west@cust.sp.net.
> - then numbers assigned to the customer are routed
>    something like:
>    1-212-555-88xx -> contact from east@cust.sp.net
>    1-408-555-88xx -> contact from west@cust.sp.net
> - also, routing for names be something like:
>    bob@cust.sp.net   -> contact from east@cust.sp.net
>    carol@cust.sp.net -> "
>    ted@cust.sp.net   -> contact from west@cust.sp.net
>    alice@cust.sp.net -> "
>    others...
>=20
> Is that right?
> <DM> Yes
>=20
> You then can't describe that as setting up routing paths for=20
> cust.sp.net. You are instead describing some entirely different =
routing=20
> algorithm based on the user part.
> <DM>  Yes, it requires a local policy to determine the location of the =
users.
>=20
> Subdomains east.cust.sp.net and west.cust.sp.net solve this problem in =

> large part - at least for the numbers.
>=20
> You clearly don't find that acceptable for the names. If those names=20
> were exposed to end users I wouldn't find it acceptable either. (I =
want=20
> to be pkyzivat@cisco.com, not pkyzivat@east.cisco.com.)
> <DM> Agreed
>=20
> The question is whether this is the best way to solve that problem.
>=20
> I think that solution won't be very acceptable to an SP, or to the=20
> customer of that SP. Having the SP provision every one of your users=20
> just isn't going to be popular with anyone. (Unless the SP gets a big=20
> enough per-user fee.)
> <DM> I agree but if the parties involved reach an acceptable =
agreement, the technology should support it IMO.
>=20
> Its not quite as much of a problem with numbers, because they are =
likely=20
> to be managed in blocks. And the numbers are really owned/managed by =
the=20
> SP anyway so they are forced to do something about them.
>=20
> This is also a problem that is not unique to this SP interconnect=20
> mechanism. It is pretty much the same problem in any big company with =
a=20
> lot of sites, even if they have a private network connecting all the =
sites.
>=20
> What you need in this case is a proxy at the company domain level=20
> (cust.sp.net in this case perhaps) that has the roster of all =
usernames=20
> and a mapping to a subdomain - mapping bob@cust.sp.net to=20
> bob@east.cust.sp.net.
> <DM> Yes but some enterprise customers do not want to do this and are =
able to convince the SP to do this.
>=20
> Then, the company has one server registering To:cust.sp.net to provide =

> this mapping, and other servers registering To:east.cust.sp.net, ...
>=20
> 	Thanks,
> 	Paul
>=20
>=20
> Middleton, David (NSN - US/Boca Raton) wrote:
>> Some colleagues have pointed out an aspect that I had not considered =
concerning the domain registration To: user part.  There are cases where =
an Enterprise has multiple PBXs that register and calls should be routed =
to one of the PBXs based on which user is called.  In other words, =
instead of the Enterprise with multiple PBXs appearing as one logical =
PBX, the Enterprise customer wants the service provider to take =
responsibility for distributing the calls to the specific PBX that hosts =
the called user.
>>
>> While this problem could be solved with a local policy to choose one =
of several contacts it is not as clean an approach as being able to =
treat each registration as a separate entity with it's own contact(s).  =
While one could propose that these be treated separately by making =
different sub-domains, the Enterprise most assuredly will not want to =
publish this sub-domain on business cards, etc. when using e-mail style =
addresses.  Besides, it would cause the address to change if the user =
moves from one PBX to another as might happen with a physical move.
>>
>> Another aspect to consider is that the Enterprise user, in this =
scenario, is only reachable via one registered "To: user@host".  So not =
having separate registrations would result in a more complex definition =
and implementation of things like (a.) is there an unexpired binding and =
(b.) picking appropriate contacts by q-value.
>>
>> I think this makes a strong case for having a user part in the To: =
header and treating each unique user@host as a separate registration =E0 =
la RFC 3264.
>>
>> In summary, I support Option 1) as well.
>>
>> - David
>>
>> =20
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of ext Richard Shockey
>> Sent: Wednesday, October 21, 2009 4:42 PM
>> To: dispatch@ietf.org
>> Subject: Re: [dispatch] To-URI in domain-registration
>>
>> +1 to Option 1) as well
>>
>>>  -----Original Message-----
>>>  From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] =
On
>>>  Behalf Of David Hancock
>>>  Sent: Wednesday, October 21, 2009 4:05 PM
>>>  To: Brian Lindsay; Hadriel Kaplan; dispatch@ietf.org
>>>  Subject: Re: [dispatch] To-URI in domain-registration
>>> =20
>>> =20
>>>  Hadriel - I'd prefer option 1) as well.
>>> =20
>>>  David
>>> =20
>>>  > -----Original Message-----
>>>  > From: dispatch-bounces@ietf.org =
[mailto:dispatch-bounces@ietf.org]
>>>  On
>>>  > Behalf Of Brian Lindsay
>>>  > Sent: Wednesday, October 21, 2009 1:53 PM
>>>  > To: Hadriel Kaplan; dispatch@ietf.org
>>>  > Subject: Re: [dispatch] To-URI in domain-registration
>>>  >
>>>  >  Hi Hadriel,
>>>  >
>>>  >     Per my comment yesterday, I'd prefer (1).
>>>  >
>>>  >     Thanks.
>>>  >
>>>  > -----Original Message-----
>>>  > From: dispatch-bounces@ietf.org =
[mailto:dispatch-bounces@ietf.org]
>>>  On
>>>  > Behalf Of Hadriel Kaplan
>>>  > Sent: Wednesday, October 21, 2009 3:43 PM
>>>  > To: dispatch@ietf.org
>>>  > Subject: [dispatch] To-URI in domain-registration
>>>  >
>>>  > Howdy,
>>>  > I'd like to get a consensus call on the question of what the =
To-URI
>>>  > should be for the REGISTER request of a domain registration, =
before
>>>  > finishing the next rev of the draft.  This will affect quite a =
bit
>>>  of
>>>  > text in the draft (e.g., examples), and I think that the decision
>>>  about
>>>  > that will also affect whether we need more than a =
Require:option-tag
>>>  as
>>>  > well.
>>>  >
>>>  > The choices right now are:
>>>  > 1) Mandate the To-URI be in the syntactical form of an AoR,
>>>  including
>>>  > the user@ portion, the same as the From-URI.
>>>  > 	Pros: closer to what current proprietary and other-SDO
>>>  > mechanisms do, on the wire.  Therefore, presumably less of a =
change
>>>  and
>>>  > less barrier to adoption.
>>>  > 2) Mandate the To-URI be a domain only (no user@), while the =
From-
>>>  URI
>>>  > remains the "AoR" of the PBX registering.
>>>  > 	Pros: semantically accurate (you're registering a domain, not an
>>>  > AoR), and may obviate the need for a new header or param to =
indicate
>>>  > this new mechanism is being used.
>>>  >
>>>  > Can people please indicate which they'd prefer?
>>>  >
>>>  > Thanks!
>>>  > -hadriel
>>>  > _______________________________________________
>>>  > dispatch mailing list
>>>  > dispatch@ietf.org
>>>  > https://www.ietf.org/mailman/listinfo/dispatch
>>>  > _______________________________________________
>>>  > dispatch mailing list
>>>  > dispatch@ietf.org
>>>  > https://www.ietf.org/mailman/listinfo/dispatch
>>>  _______________________________________________
>>>  dispatch mailing list
>>>  dispatch@ietf.org
>>>  https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>=20

From pkyzivat@cisco.com  Thu Nov 12 05:46:13 2009
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Middleton, David (NSN - US/Boca Raton) wrote:
> I agree this is possible but having the SP cross-reference domains is not in the spirit of domain registration from my perspective.

But having the SP partition traffic for one domain based on the user 
part of the request *is* in the spirit? IMO it is exactly the same 
thing, or cleaner.

Its a matter of who is responsible for a particular domain, the sp or 
the pbx. IMO you should only take *other* information (such as the 
username) into account if you are responsible for the domain of the 
request. In my example, the sp is responsible for foo.sp.com, but the 
pbxes are responsible for east.foo.sp.com and west.foo.sp.com. The other 
way, both the sp and the pbxes are responsible for foo.sp.com, which is 
problematic to me.

I see from the notes of the ad hoc, that there is some desire to 
restrict the scope to phone numbers. In that case it again is possible 
to give each pbx its own domain name.

	Thanks,
	Paul

> -----Original Message-----
> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Wednesday, November 11, 2009 5:35 PM
> To: Middleton, David (NSN - US/Boca Raton)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] To-URI in domain-registration
> 
> 
> 
> Middleton, David (NSN - US/Boca Raton) wrote:
>> Paul,
>>
>> Sorry for the delay in responding.
>>
>> I think you are on target as far as describing many (perhaps most) SP / PBX interconnect arrangements.  However, the functionality of having the SP route calls per user is a real-world scenario which proves there is merit for having the capability.
> 
> This could still be dealt with by having the pbxes register to separate 
> sub-domains and the SP knowing (via provisioning) which things to route 
> to each.
> 
> So instead of having:
> 
> foo-east-pbx register to east@foo.sp.com
> foo-west-pbx register to west@foo.sp.com
> 
> you could simply have
> 
> foo-east-pbx register to east.foo.sp.com
> foo-west-pbx register to west.foo.sp.com
> 
> Then the question of how requests addressed to xyz@foo.sp.com are mapped 
> to something@east.foo.sp.com or something@west.foo.sp.com is simply out 
> of scope of this document.
> 
> 	Thanks,
> 	Paul
> 
> 
>> Also see inline.
>>
>> David
>>
>>
>>
>> -----Original Message-----
>> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: Friday, October 30, 2009 7:37 PM
>> To: Middleton, David (NSN - US/Boca Raton)
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] To-URI in domain-registration
>>
>> David,
>>
>> IIUC, you are proposing a case such as:
>>
>> - a customer of sp.net has domain cust.sp.net
>> - the customer has two PBX systems, east and west.
>> - you propose that they register, respectively
>>    To: east@cust.sp.net, and
>>    To: west@cust.sp.net.
>> - then numbers assigned to the customer are routed
>>    something like:
>>    1-212-555-88xx -> contact from east@cust.sp.net
>>    1-408-555-88xx -> contact from west@cust.sp.net
>> - also, routing for names be something like:
>>    bob@cust.sp.net   -> contact from east@cust.sp.net
>>    carol@cust.sp.net -> "
>>    ted@cust.sp.net   -> contact from west@cust.sp.net
>>    alice@cust.sp.net -> "
>>    others...
>>
>> Is that right?
>> <DM> Yes
>>
>> You then can't describe that as setting up routing paths for 
>> cust.sp.net. You are instead describing some entirely different routing 
>> algorithm based on the user part.
>> <DM>  Yes, it requires a local policy to determine the location of the users.
>>
>> Subdomains east.cust.sp.net and west.cust.sp.net solve this problem in 
>> large part - at least for the numbers.
>>
>> You clearly don't find that acceptable for the names. If those names 
>> were exposed to end users I wouldn't find it acceptable either. (I want 
>> to be pkyzivat@cisco.com, not pkyzivat@east.cisco.com.)
>> <DM> Agreed
>>
>> The question is whether this is the best way to solve that problem.
>>
>> I think that solution won't be very acceptable to an SP, or to the 
>> customer of that SP. Having the SP provision every one of your users 
>> just isn't going to be popular with anyone. (Unless the SP gets a big 
>> enough per-user fee.)
>> <DM> I agree but if the parties involved reach an acceptable agreement, the technology should support it IMO.
>>
>> Its not quite as much of a problem with numbers, because they are likely 
>> to be managed in blocks. And the numbers are really owned/managed by the 
>> SP anyway so they are forced to do something about them.
>>
>> This is also a problem that is not unique to this SP interconnect 
>> mechanism. It is pretty much the same problem in any big company with a 
>> lot of sites, even if they have a private network connecting all the sites.
>>
>> What you need in this case is a proxy at the company domain level 
>> (cust.sp.net in this case perhaps) that has the roster of all usernames 
>> and a mapping to a subdomain - mapping bob@cust.sp.net to 
>> bob@east.cust.sp.net.
>> <DM> Yes but some enterprise customers do not want to do this and are able to convince the SP to do this.
>>
>> Then, the company has one server registering To:cust.sp.net to provide 
>> this mapping, and other servers registering To:east.cust.sp.net, ...
>>
>> 	Thanks,
>> 	Paul
>>
>>
>> Middleton, David (NSN - US/Boca Raton) wrote:
>>> Some colleagues have pointed out an aspect that I had not considered concerning the domain registration To: user part.  There are cases where an Enterprise has multiple PBXs that register and calls should be routed to one of the PBXs based on which user is called.  In other words, instead of the Enterprise with multiple PBXs appearing as one logical PBX, the Enterprise customer wants the service provider to take responsibility for distributing the calls to the specific PBX that hosts the called user.
>>>
>>> While this problem could be solved with a local policy to choose one of several contacts it is not as clean an approach as being able to treat each registration as a separate entity with it's own contact(s).  While one could propose that these be treated separately by making different sub-domains, the Enterprise most assuredly will not want to publish this sub-domain on business cards, etc. when using e-mail style addresses.  Besides, it would cause the address to change if the user moves from one PBX to another as might happen with a physical move.
>>>
>>> Another aspect to consider is that the Enterprise user, in this scenario, is only reachable via one registered "To: user@host".  So not having separate registrations would result in a more complex definition and implementation of things like (a.) is there an unexpired binding and (b.) picking appropriate contacts by q-value.
>>>
>>> I think this makes a strong case for having a user part in the To: header and treating each unique user@host as a separate registration à la RFC 3264.
>>>
>>> In summary, I support Option 1) as well.
>>>
>>> - David
>>>
>>>  
>>>
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of ext Richard Shockey
>>> Sent: Wednesday, October 21, 2009 4:42 PM
>>> To: dispatch@ietf.org
>>> Subject: Re: [dispatch] To-URI in domain-registration
>>>
>>> +1 to Option 1) as well
>>>
>>>>  -----Original Message-----
>>>>  From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>>>>  Behalf Of David Hancock
>>>>  Sent: Wednesday, October 21, 2009 4:05 PM
>>>>  To: Brian Lindsay; Hadriel Kaplan; dispatch@ietf.org
>>>>  Subject: Re: [dispatch] To-URI in domain-registration
>>>>  
>>>>  
>>>>  Hadriel - I'd prefer option 1) as well.
>>>>  
>>>>  David
>>>>  
>>>>  > -----Original Message-----
>>>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
>>>>  On
>>>>  > Behalf Of Brian Lindsay
>>>>  > Sent: Wednesday, October 21, 2009 1:53 PM
>>>>  > To: Hadriel Kaplan; dispatch@ietf.org
>>>>  > Subject: Re: [dispatch] To-URI in domain-registration
>>>>  >
>>>>  >  Hi Hadriel,
>>>>  >
>>>>  >     Per my comment yesterday, I'd prefer (1).
>>>>  >
>>>>  >     Thanks.
>>>>  >
>>>>  > -----Original Message-----
>>>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
>>>>  On
>>>>  > Behalf Of Hadriel Kaplan
>>>>  > Sent: Wednesday, October 21, 2009 3:43 PM
>>>>  > To: dispatch@ietf.org
>>>>  > Subject: [dispatch] To-URI in domain-registration
>>>>  >
>>>>  > Howdy,
>>>>  > I'd like to get a consensus call on the question of what the To-URI
>>>>  > should be for the REGISTER request of a domain registration, before
>>>>  > finishing the next rev of the draft.  This will affect quite a bit
>>>>  of
>>>>  > text in the draft (e.g., examples), and I think that the decision
>>>>  about
>>>>  > that will also affect whether we need more than a Require:option-tag
>>>>  as
>>>>  > well.
>>>>  >
>>>>  > The choices right now are:
>>>>  > 1) Mandate the To-URI be in the syntactical form of an AoR,
>>>>  including
>>>>  > the user@ portion, the same as the From-URI.
>>>>  > 	Pros: closer to what current proprietary and other-SDO
>>>>  > mechanisms do, on the wire.  Therefore, presumably less of a change
>>>>  and
>>>>  > less barrier to adoption.
>>>>  > 2) Mandate the To-URI be a domain only (no user@), while the From-
>>>>  URI
>>>>  > remains the "AoR" of the PBX registering.
>>>>  > 	Pros: semantically accurate (you're registering a domain, not an
>>>>  > AoR), and may obviate the need for a new header or param to indicate
>>>>  > this new mechanism is being used.
>>>>  >
>>>>  > Can people please indicate which they'd prefer?
>>>>  >
>>>>  > Thanks!
>>>>  > -hadriel
>>>>  > _______________________________________________
>>>>  > dispatch mailing list
>>>>  > dispatch@ietf.org
>>>>  > https://www.ietf.org/mailman/listinfo/dispatch
>>>>  > _______________________________________________
>>>>  > dispatch mailing list
>>>>  > dispatch@ietf.org
>>>>  > https://www.ietf.org/mailman/listinfo/dispatch
>>>>  _______________________________________________
>>>>  dispatch mailing list
>>>>  dispatch@ietf.org
>>>>  https://www.ietf.org/mailman/listinfo/dispatch
>>> _______________________________________________
>>> 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  Thu Nov 12 12:49:52 2009
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To: "'Paul Kyzivat'" <pkyzivat@cisco.com>, "'Middleton, David \(NSN - US/Boca Raton\)'" <david.middleton@nsn.com>
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Subject: Re: [dispatch] To-URI in domain-registration
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To clarify .. in the discussions within the SIPforum there was no =
question
that the scope was restricted to phone numbers.=20

here were only 2-3 participants that did not usually attend IETF =
meetings
and those represented small CLEC's. As a practical matter the overlap
between IETF and the SIPforum discussions was nearly total.=20



>  -----Original Message-----
>  From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>  Behalf Of Paul Kyzivat
>  Sent: Thursday, November 12, 2009 8:47 AM
>  To: Middleton, David (NSN - US/Boca Raton)
>  Cc: dispatch@ietf.org
>  Subject: Re: [dispatch] To-URI in domain-registration
> =20
> =20
> =20
>  Middleton, David (NSN - US/Boca Raton) wrote:
>  > I agree this is possible but having the SP cross-reference domains
>  is not in the spirit of domain registration from my perspective.
> =20
>  But having the SP partition traffic for one domain based on the user
>  part of the request *is* in the spirit? IMO it is exactly the same
>  thing, or cleaner.
> =20
>  Its a matter of who is responsible for a particular domain, the sp or
>  the pbx. IMO you should only take *other* information (such as the
>  username) into account if you are responsible for the domain of the
>  request. In my example, the sp is responsible for foo.sp.com, but the
>  pbxes are responsible for east.foo.sp.com and west.foo.sp.com. The
>  other
>  way, both the sp and the pbxes are responsible for foo.sp.com, which
>  is
>  problematic to me.
> =20
>  I see from the notes of the ad hoc, that there is some desire to
>  restrict the scope to phone numbers. In that case it again is =
possible
>  to give each pbx its own domain name.
> =20
>  	Thanks,
>  	Paul
> =20
>  > -----Original Message-----
>  > From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
>  > Sent: Wednesday, November 11, 2009 5:35 PM
>  > To: Middleton, David (NSN - US/Boca Raton)
>  > Cc: dispatch@ietf.org
>  > Subject: Re: [dispatch] To-URI in domain-registration
>  >
>  >
>  >
>  > Middleton, David (NSN - US/Boca Raton) wrote:
>  >> Paul,
>  >>
>  >> Sorry for the delay in responding.
>  >>
>  >> I think you are on target as far as describing many (perhaps most)
>  SP / PBX interconnect arrangements.  However, the functionality of
>  having the SP route calls per user is a real-world scenario which
>  proves there is merit for having the capability.
>  >
>  > This could still be dealt with by having the pbxes register to
>  separate
>  > sub-domains and the SP knowing (via provisioning) which things to
>  route
>  > to each.
>  >
>  > So instead of having:
>  >
>  > foo-east-pbx register to east@foo.sp.com
>  > foo-west-pbx register to west@foo.sp.com
>  >
>  > you could simply have
>  >
>  > foo-east-pbx register to east.foo.sp.com
>  > foo-west-pbx register to west.foo.sp.com
>  >
>  > Then the question of how requests addressed to xyz@foo.sp.com are
>  mapped
>  > to something@east.foo.sp.com or something@west.foo.sp.com is simply
>  out
>  > of scope of this document.
>  >
>  > 	Thanks,
>  > 	Paul
>  >
>  >
>  >> Also see inline.
>  >>
>  >> David
>  >>
>  >>
>  >>
>  >> -----Original Message-----
>  >> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
>  >> Sent: Friday, October 30, 2009 7:37 PM
>  >> To: Middleton, David (NSN - US/Boca Raton)
>  >> Cc: dispatch@ietf.org
>  >> Subject: Re: [dispatch] To-URI in domain-registration
>  >>
>  >> David,
>  >>
>  >> IIUC, you are proposing a case such as:
>  >>
>  >> - a customer of sp.net has domain cust.sp.net
>  >> - the customer has two PBX systems, east and west.
>  >> - you propose that they register, respectively
>  >>    To: east@cust.sp.net, and
>  >>    To: west@cust.sp.net.
>  >> - then numbers assigned to the customer are routed
>  >>    something like:
>  >>    1-212-555-88xx -> contact from east@cust.sp.net
>  >>    1-408-555-88xx -> contact from west@cust.sp.net
>  >> - also, routing for names be something like:
>  >>    bob@cust.sp.net   -> contact from east@cust.sp.net
>  >>    carol@cust.sp.net -> "
>  >>    ted@cust.sp.net   -> contact from west@cust.sp.net
>  >>    alice@cust.sp.net -> "
>  >>    others...
>  >>
>  >> Is that right?
>  >> <DM> Yes
>  >>
>  >> You then can't describe that as setting up routing paths for
>  >> cust.sp.net. You are instead describing some entirely different
>  routing
>  >> algorithm based on the user part.
>  >> <DM>  Yes, it requires a local policy to determine the location of
>  the users.
>  >>
>  >> Subdomains east.cust.sp.net and west.cust.sp.net solve this =
problem
>  in
>  >> large part - at least for the numbers.
>  >>
>  >> You clearly don't find that acceptable for the names. If those
>  names
>  >> were exposed to end users I wouldn't find it acceptable either. (I
>  want
>  >> to be pkyzivat@cisco.com, not pkyzivat@east.cisco.com.)
>  >> <DM> Agreed
>  >>
>  >> The question is whether this is the best way to solve that =
problem.
>  >>
>  >> I think that solution won't be very acceptable to an SP, or to the
>  >> customer of that SP. Having the SP provision every one of your
>  users
>  >> just isn't going to be popular with anyone. (Unless the SP gets a
>  big
>  >> enough per-user fee.)
>  >> <DM> I agree but if the parties involved reach an acceptable
>  agreement, the technology should support it IMO.
>  >>
>  >> Its not quite as much of a problem with numbers, because they are
>  likely
>  >> to be managed in blocks. And the numbers are really owned/managed
>  by the
>  >> SP anyway so they are forced to do something about them.
>  >>
>  >> This is also a problem that is not unique to this SP interconnect
>  >> mechanism. It is pretty much the same problem in any big company
>  with a
>  >> lot of sites, even if they have a private network connecting all
>  the sites.
>  >>
>  >> What you need in this case is a proxy at the company domain level
>  >> (cust.sp.net in this case perhaps) that has the roster of all
>  usernames
>  >> and a mapping to a subdomain - mapping bob@cust.sp.net to
>  >> bob@east.cust.sp.net.
>  >> <DM> Yes but some enterprise customers do not want to do this and
>  are able to convince the SP to do this.
>  >>
>  >> Then, the company has one server registering To:cust.sp.net to
>  provide
>  >> this mapping, and other servers registering To:east.cust.sp.net,
>  ...
>  >>
>  >> 	Thanks,
>  >> 	Paul
>  >>
>  >>
>  >> Middleton, David (NSN - US/Boca Raton) wrote:
>  >>> Some colleagues have pointed out an aspect that I had not
>  considered concerning the domain registration To: user part.  There
>  are cases where an Enterprise has multiple PBXs that register and
>  calls should be routed to one of the PBXs based on which user is
>  called.  In other words, instead of the Enterprise with multiple PBXs
>  appearing as one logical PBX, the Enterprise customer wants the
>  service provider to take responsibility for distributing the calls to
>  the specific PBX that hosts the called user.
>  >>>
>  >>> While this problem could be solved with a local policy to choose
>  one of several contacts it is not as clean an approach as being able
>  to treat each registration as a separate entity with it's own
>  contact(s).  While one could propose that these be treated separately
>  by making different sub-domains, the Enterprise most assuredly will
>  not want to publish this sub-domain on business cards, etc. when =
using
>  e-mail style addresses.  Besides, it would cause the address to =
change
>  if the user moves from one PBX to another as might happen with a
>  physical move.
>  >>>
>  >>> Another aspect to consider is that the Enterprise user, in this
>  scenario, is only reachable via one registered "To: user@host".  So
>  not having separate registrations would result in a more complex
>  definition and implementation of things like (a.) is there an
>  unexpired binding and (b.) picking appropriate contacts by q-value.
>  >>>
>  >>> I think this makes a strong case for having a user part in the =
To:
>  header and treating each unique user@host as a separate registration =
=E0
>  la RFC 3264.
>  >>>
>  >>> In summary, I support Option 1) as well.
>  >>>
>  >>> - David
>  >>>
>  >>>
>  >>>
>  >>> -----Original Message-----
>  >>> From: dispatch-bounces@ietf.org =
[mailto:dispatch-bounces@ietf.org]
>  On Behalf Of ext Richard Shockey
>  >>> Sent: Wednesday, October 21, 2009 4:42 PM
>  >>> To: dispatch@ietf.org
>  >>> Subject: Re: [dispatch] To-URI in domain-registration
>  >>>
>  >>> +1 to Option 1) as well
>  >>>
>  >>>>  -----Original Message-----
>  >>>>  From: dispatch-bounces@ietf.org [mailto:dispatch-
>  bounces@ietf.org] On
>  >>>>  Behalf Of David Hancock
>  >>>>  Sent: Wednesday, October 21, 2009 4:05 PM
>  >>>>  To: Brian Lindsay; Hadriel Kaplan; dispatch@ietf.org
>  >>>>  Subject: Re: [dispatch] To-URI in domain-registration
>  >>>>
>  >>>>
>  >>>>  Hadriel - I'd prefer option 1) as well.
>  >>>>
>  >>>>  David
>  >>>>
>  >>>>  > -----Original Message-----
>  >>>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-
>  bounces@ietf.org]
>  >>>>  On
>  >>>>  > Behalf Of Brian Lindsay
>  >>>>  > Sent: Wednesday, October 21, 2009 1:53 PM
>  >>>>  > To: Hadriel Kaplan; dispatch@ietf.org
>  >>>>  > Subject: Re: [dispatch] To-URI in domain-registration
>  >>>>  >
>  >>>>  >  Hi Hadriel,
>  >>>>  >
>  >>>>  >     Per my comment yesterday, I'd prefer (1).
>  >>>>  >
>  >>>>  >     Thanks.
>  >>>>  >
>  >>>>  > -----Original Message-----
>  >>>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-
>  bounces@ietf.org]
>  >>>>  On
>  >>>>  > Behalf Of Hadriel Kaplan
>  >>>>  > Sent: Wednesday, October 21, 2009 3:43 PM
>  >>>>  > To: dispatch@ietf.org
>  >>>>  > Subject: [dispatch] To-URI in domain-registration
>  >>>>  >
>  >>>>  > Howdy,
>  >>>>  > I'd like to get a consensus call on the question of what the
>  To-URI
>  >>>>  > should be for the REGISTER request of a domain registration,
>  before
>  >>>>  > finishing the next rev of the draft.  This will affect quite =
a
>  bit
>  >>>>  of
>  >>>>  > text in the draft (e.g., examples), and I think that the
>  decision
>  >>>>  about
>  >>>>  > that will also affect whether we need more than a
>  Require:option-tag
>  >>>>  as
>  >>>>  > well.
>  >>>>  >
>  >>>>  > The choices right now are:
>  >>>>  > 1) Mandate the To-URI be in the syntactical form of an AoR,
>  >>>>  including
>  >>>>  > the user@ portion, the same as the From-URI.
>  >>>>  > 	Pros: closer to what current proprietary and other-SDO
>  >>>>  > mechanisms do, on the wire.  Therefore, presumably less of a
>  change
>  >>>>  and
>  >>>>  > less barrier to adoption.
>  >>>>  > 2) Mandate the To-URI be a domain only (no user@), while the
>  From-
>  >>>>  URI
>  >>>>  > remains the "AoR" of the PBX registering.
>  >>>>  > 	Pros: semantically accurate (you're registering a domain,
>  not an
>  >>>>  > AoR), and may obviate the need for a new header or param to
>  indicate
>  >>>>  > this new mechanism is being used.
>  >>>>  >
>  >>>>  > Can people please indicate which they'd prefer?
>  >>>>  >
>  >>>>  > Thanks!
>  >>>>  > -hadriel
>  >>>>  > _______________________________________________
>  >>>>  > dispatch mailing list
>  >>>>  > dispatch@ietf.org
>  >>>>  > https://www.ietf.org/mailman/listinfo/dispatch
>  >>>>  > _______________________________________________
>  >>>>  > dispatch mailing list
>  >>>>  > dispatch@ietf.org
>  >>>>  > https://www.ietf.org/mailman/listinfo/dispatch
>  >>>>  _______________________________________________
>  >>>>  dispatch mailing list
>  >>>>  dispatch@ietf.org
>  >>>>  https://www.ietf.org/mailman/listinfo/dispatch
>  >>> _______________________________________________
>  >>> dispatch mailing list
>  >>> dispatch@ietf.org
>  >>> https://www.ietf.org/mailman/listinfo/dispatch
>  >>> _______________________________________________
>  >>> dispatch mailing list
>  >>> dispatch@ietf.org
>  >>> https://www.ietf.org/mailman/listinfo/dispatch
>  >>>
>  >
>  _______________________________________________
>  dispatch mailing list
>  dispatch@ietf.org
>  https://www.ietf.org/mailman/listinfo/dispatch


From john.elwell@siemens-enterprise.com  Thu Nov 12 13:50:32 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Richard Shockey <richard@shockey.us>, "'Paul Kyzivat'" <pkyzivat@cisco.com>, "'Middleton, David (NSN - US/Boca Raton)'" <david.middleton@nsn.com>
Date: Thu, 12 Nov 2009 22:50:11 +0100
Thread-Topic: Phone numbers and domain registration (was RE: [dispatch] To-URI in domain-registration)
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Subject: [dispatch] Phone numbers and domain registration (was RE: To-URI in domain-registration)
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During the DISPATCH ad hoc session there was indeed a suggestion to limit t=
he mechanism to domains employing E.164-based SIP URIs. However, this would=
 leave an issue with contact URIs, which would presumably not be constraine=
d to being based on telephone numbers. It must be possible to route an out-=
of-dialog request to a contact URI issued by the SIP-PBX.

John

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Richard Shockey
> Sent: 12 November 2009 20:50
> To: 'Paul Kyzivat'; 'Middleton, David (NSN - US/Boca Raton)'
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] To-URI in domain-registration
>=20
> To clarify .. in the discussions within the SIPforum there=20
> was no question
> that the scope was restricted to phone numbers.=20
>=20
> here were only 2-3 participants that did not usually attend=20
> IETF meetings
> and those represented small CLEC's. As a practical matter the overlap
> between IETF and the SIPforum discussions was nearly total.=20
>=20
>=20
>=20
> >  -----Original Message-----
> >  From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On
> >  Behalf Of Paul Kyzivat
> >  Sent: Thursday, November 12, 2009 8:47 AM
> >  To: Middleton, David (NSN - US/Boca Raton)
> >  Cc: dispatch@ietf.org
> >  Subject: Re: [dispatch] To-URI in domain-registration
> > =20
> > =20
> > =20
> >  Middleton, David (NSN - US/Boca Raton) wrote:
> >  > I agree this is possible but having the SP=20
> cross-reference domains
> >  is not in the spirit of domain registration from my perspective.
> > =20
> >  But having the SP partition traffic for one domain based=20
> on the user
> >  part of the request *is* in the spirit? IMO it is exactly the same
> >  thing, or cleaner.
> > =20
> >  Its a matter of who is responsible for a particular=20
> domain, the sp or
> >  the pbx. IMO you should only take *other* information (such as the
> >  username) into account if you are responsible for the domain of the
> >  request. In my example, the sp is responsible for=20
> foo.sp.com, but the
> >  pbxes are responsible for east.foo.sp.com and west.foo.sp.com. The
> >  other
> >  way, both the sp and the pbxes are responsible for=20
> foo.sp.com, which
> >  is
> >  problematic to me.
> > =20
> >  I see from the notes of the ad hoc, that there is some desire to
> >  restrict the scope to phone numbers. In that case it again=20
> is possible
> >  to give each pbx its own domain name.
> > =20
> >  	Thanks,
> >  	Paul
> > =20
> >  > -----Original Message-----
> >  > From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >  > Sent: Wednesday, November 11, 2009 5:35 PM
> >  > To: Middleton, David (NSN - US/Boca Raton)
> >  > Cc: dispatch@ietf.org
> >  > Subject: Re: [dispatch] To-URI in domain-registration
> >  >
> >  >
> >  >
> >  > Middleton, David (NSN - US/Boca Raton) wrote:
> >  >> Paul,
> >  >>
> >  >> Sorry for the delay in responding.
> >  >>
> >  >> I think you are on target as far as describing many=20
> (perhaps most)
> >  SP / PBX interconnect arrangements.  However, the functionality of
> >  having the SP route calls per user is a real-world scenario which
> >  proves there is merit for having the capability.
> >  >
> >  > This could still be dealt with by having the pbxes register to
> >  separate
> >  > sub-domains and the SP knowing (via provisioning) which things to
> >  route
> >  > to each.
> >  >
> >  > So instead of having:
> >  >
> >  > foo-east-pbx register to east@foo.sp.com
> >  > foo-west-pbx register to west@foo.sp.com
> >  >
> >  > you could simply have
> >  >
> >  > foo-east-pbx register to east.foo.sp.com
> >  > foo-west-pbx register to west.foo.sp.com
> >  >
> >  > Then the question of how requests addressed to xyz@foo.sp.com are
> >  mapped
> >  > to something@east.foo.sp.com or=20
> something@west.foo.sp.com is simply
> >  out
> >  > of scope of this document.
> >  >
> >  > 	Thanks,
> >  > 	Paul
> >  >
> >  >
> >  >> Also see inline.
> >  >>
> >  >> David
> >  >>
> >  >>
> >  >>
> >  >> -----Original Message-----
> >  >> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >  >> Sent: Friday, October 30, 2009 7:37 PM
> >  >> To: Middleton, David (NSN - US/Boca Raton)
> >  >> Cc: dispatch@ietf.org
> >  >> Subject: Re: [dispatch] To-URI in domain-registration
> >  >>
> >  >> David,
> >  >>
> >  >> IIUC, you are proposing a case such as:
> >  >>
> >  >> - a customer of sp.net has domain cust.sp.net
> >  >> - the customer has two PBX systems, east and west.
> >  >> - you propose that they register, respectively
> >  >>    To: east@cust.sp.net, and
> >  >>    To: west@cust.sp.net.
> >  >> - then numbers assigned to the customer are routed
> >  >>    something like:
> >  >>    1-212-555-88xx -> contact from east@cust.sp.net
> >  >>    1-408-555-88xx -> contact from west@cust.sp.net
> >  >> - also, routing for names be something like:
> >  >>    bob@cust.sp.net   -> contact from east@cust.sp.net
> >  >>    carol@cust.sp.net -> "
> >  >>    ted@cust.sp.net   -> contact from west@cust.sp.net
> >  >>    alice@cust.sp.net -> "
> >  >>    others...
> >  >>
> >  >> Is that right?
> >  >> <DM> Yes
> >  >>
> >  >> You then can't describe that as setting up routing paths for
> >  >> cust.sp.net. You are instead describing some entirely different
> >  routing
> >  >> algorithm based on the user part.
> >  >> <DM>  Yes, it requires a local policy to determine the=20
> location of
> >  the users.
> >  >>
> >  >> Subdomains east.cust.sp.net and west.cust.sp.net solve=20
> this problem
> >  in
> >  >> large part - at least for the numbers.
> >  >>
> >  >> You clearly don't find that acceptable for the names. If those
> >  names
> >  >> were exposed to end users I wouldn't find it acceptable=20
> either. (I
> >  want
> >  >> to be pkyzivat@cisco.com, not pkyzivat@east.cisco.com.)
> >  >> <DM> Agreed
> >  >>
> >  >> The question is whether this is the best way to solve=20
> that problem.
> >  >>
> >  >> I think that solution won't be very acceptable to an=20
> SP, or to the
> >  >> customer of that SP. Having the SP provision every one of your
> >  users
> >  >> just isn't going to be popular with anyone. (Unless the=20
> SP gets a
> >  big
> >  >> enough per-user fee.)
> >  >> <DM> I agree but if the parties involved reach an acceptable
> >  agreement, the technology should support it IMO.
> >  >>
> >  >> Its not quite as much of a problem with numbers,=20
> because they are
> >  likely
> >  >> to be managed in blocks. And the numbers are really=20
> owned/managed
> >  by the
> >  >> SP anyway so they are forced to do something about them.
> >  >>
> >  >> This is also a problem that is not unique to this SP=20
> interconnect
> >  >> mechanism. It is pretty much the same problem in any big company
> >  with a
> >  >> lot of sites, even if they have a private network connecting all
> >  the sites.
> >  >>
> >  >> What you need in this case is a proxy at the company=20
> domain level
> >  >> (cust.sp.net in this case perhaps) that has the roster of all
> >  usernames
> >  >> and a mapping to a subdomain - mapping bob@cust.sp.net to
> >  >> bob@east.cust.sp.net.
> >  >> <DM> Yes but some enterprise customers do not want to=20
> do this and
> >  are able to convince the SP to do this.
> >  >>
> >  >> Then, the company has one server registering To:cust.sp.net to
> >  provide
> >  >> this mapping, and other servers registering To:east.cust.sp.net,
> >  ...
> >  >>
> >  >> 	Thanks,
> >  >> 	Paul
> >  >>
> >  >>
> >  >> Middleton, David (NSN - US/Boca Raton) wrote:
> >  >>> Some colleagues have pointed out an aspect that I had not
> >  considered concerning the domain registration To: user part.  There
> >  are cases where an Enterprise has multiple PBXs that register and
> >  calls should be routed to one of the PBXs based on which user is
> >  called.  In other words, instead of the Enterprise with=20
> multiple PBXs
> >  appearing as one logical PBX, the Enterprise customer wants the
> >  service provider to take responsibility for distributing=20
> the calls to
> >  the specific PBX that hosts the called user.
> >  >>>
> >  >>> While this problem could be solved with a local policy=20
> to choose
> >  one of several contacts it is not as clean an approach as=20
> being able
> >  to treat each registration as a separate entity with it's own
> >  contact(s).  While one could propose that these be treated=20
> separately
> >  by making different sub-domains, the Enterprise most assuredly will
> >  not want to publish this sub-domain on business cards,=20
> etc. when using
> >  e-mail style addresses.  Besides, it would cause the=20
> address to change
> >  if the user moves from one PBX to another as might happen with a
> >  physical move.
> >  >>>
> >  >>> Another aspect to consider is that the Enterprise user, in this
> >  scenario, is only reachable via one registered "To: user@host".  So
> >  not having separate registrations would result in a more complex
> >  definition and implementation of things like (a.) is there an
> >  unexpired binding and (b.) picking appropriate contacts by q-value.
> >  >>>
> >  >>> I think this makes a strong case for having a user=20
> part in the To:
> >  header and treating each unique user@host as a separate=20
> registration =E0
> >  la RFC 3264.
> >  >>>
> >  >>> In summary, I support Option 1) as well.
> >  >>>
> >  >>> - David
> >  >>>
> >  >>>
> >  >>>
> >  >>> -----Original Message-----
> >  >>> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org]
> >  On Behalf Of ext Richard Shockey
> >  >>> Sent: Wednesday, October 21, 2009 4:42 PM
> >  >>> To: dispatch@ietf.org
> >  >>> Subject: Re: [dispatch] To-URI in domain-registration
> >  >>>
> >  >>> +1 to Option 1) as well
> >  >>>
> >  >>>>  -----Original Message-----
> >  >>>>  From: dispatch-bounces@ietf.org [mailto:dispatch-
> >  bounces@ietf.org] On
> >  >>>>  Behalf Of David Hancock
> >  >>>>  Sent: Wednesday, October 21, 2009 4:05 PM
> >  >>>>  To: Brian Lindsay; Hadriel Kaplan; dispatch@ietf.org
> >  >>>>  Subject: Re: [dispatch] To-URI in domain-registration
> >  >>>>
> >  >>>>
> >  >>>>  Hadriel - I'd prefer option 1) as well.
> >  >>>>
> >  >>>>  David
> >  >>>>
> >  >>>>  > -----Original Message-----
> >  >>>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-
> >  bounces@ietf.org]
> >  >>>>  On
> >  >>>>  > Behalf Of Brian Lindsay
> >  >>>>  > Sent: Wednesday, October 21, 2009 1:53 PM
> >  >>>>  > To: Hadriel Kaplan; dispatch@ietf.org
> >  >>>>  > Subject: Re: [dispatch] To-URI in domain-registration
> >  >>>>  >
> >  >>>>  >  Hi Hadriel,
> >  >>>>  >
> >  >>>>  >     Per my comment yesterday, I'd prefer (1).
> >  >>>>  >
> >  >>>>  >     Thanks.
> >  >>>>  >
> >  >>>>  > -----Original Message-----
> >  >>>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-
> >  bounces@ietf.org]
> >  >>>>  On
> >  >>>>  > Behalf Of Hadriel Kaplan
> >  >>>>  > Sent: Wednesday, October 21, 2009 3:43 PM
> >  >>>>  > To: dispatch@ietf.org
> >  >>>>  > Subject: [dispatch] To-URI in domain-registration
> >  >>>>  >
> >  >>>>  > Howdy,
> >  >>>>  > I'd like to get a consensus call on the question=20
> of what the
> >  To-URI
> >  >>>>  > should be for the REGISTER request of a domain=20
> registration,
> >  before
> >  >>>>  > finishing the next rev of the draft.  This will=20
> affect quite a
> >  bit
> >  >>>>  of
> >  >>>>  > text in the draft (e.g., examples), and I think that the
> >  decision
> >  >>>>  about
> >  >>>>  > that will also affect whether we need more than a
> >  Require:option-tag
> >  >>>>  as
> >  >>>>  > well.
> >  >>>>  >
> >  >>>>  > The choices right now are:
> >  >>>>  > 1) Mandate the To-URI be in the syntactical form of an AoR,
> >  >>>>  including
> >  >>>>  > the user@ portion, the same as the From-URI.
> >  >>>>  > 	Pros: closer to what current proprietary and other-SDO
> >  >>>>  > mechanisms do, on the wire.  Therefore, presumably=20
> less of a
> >  change
> >  >>>>  and
> >  >>>>  > less barrier to adoption.
> >  >>>>  > 2) Mandate the To-URI be a domain only (no user@),=20
> while the
> >  From-
> >  >>>>  URI
> >  >>>>  > remains the "AoR" of the PBX registering.
> >  >>>>  > 	Pros: semantically accurate (you're registering=20
> a domain,
> >  not an
> >  >>>>  > AoR), and may obviate the need for a new header or param to
> >  indicate
> >  >>>>  > this new mechanism is being used.
> >  >>>>  >
> >  >>>>  > Can people please indicate which they'd prefer?
> >  >>>>  >
> >  >>>>  > Thanks!
> >  >>>>  > -hadriel
> >  >>>>  > _______________________________________________
> >  >>>>  > dispatch mailing list
> >  >>>>  > dispatch@ietf.org
> >  >>>>  > https://www.ietf.org/mailman/listinfo/dispatch
> >  >>>>  > _______________________________________________
> >  >>>>  > dispatch mailing list
> >  >>>>  > dispatch@ietf.org
> >  >>>>  > https://www.ietf.org/mailman/listinfo/dispatch
> >  >>>>  _______________________________________________
> >  >>>>  dispatch mailing list
> >  >>>>  dispatch@ietf.org
> >  >>>>  https://www.ietf.org/mailman/listinfo/dispatch
> >  >>> _______________________________________________
> >  >>> dispatch mailing list
> >  >>> dispatch@ietf.org
> >  >>> https://www.ietf.org/mailman/listinfo/dispatch
> >  >>> _______________________________________________
> >  >>> dispatch mailing list
> >  >>> dispatch@ietf.org
> >  >>> https://www.ietf.org/mailman/listinfo/dispatch
> >  >>>
> >  >
> >  _______________________________________________
> >  dispatch mailing list
> >  dispatch@ietf.org
> >  https://www.ietf.org/mailman/listinfo/dispatch
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From eburger@standardstrack.com  Thu Nov 12 14:16:41 2009
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Likewise, I do not think the Note in the statement below is accurate.
> Adam Roach:  Are we really proposing to write the use of REGISTER in =20=

> the charter?  Answer: Yes.  Adam stated that he was not prepared to =20=

> have that discussion today because he thought it would be discussed =20=

> in the WG.  This feels like =93let=92s ratify this draft=94 rather =
than =20
> =93let=92s work on this problem=94.
>
>
> Spencer Dawkins: the intent is not to ratify existing draft.
>
> Note:  In spite of Spencer=92s statement, the ensuing discussion made =20=

> it very clear that for at least some of the authors, the intent was =20=

> in fact very much to ratify this draft, or something very near it.  =20=

> Apparently the initial charter draft did not mention REGISTER and it =20=

> was included because SIP Forum participants explicitly asked for it =20=

> to be added.
>
>
> Eric: the subject of whether or not REGISTER should be included in =20
> the charter has been discussed on the list, but there is not consensus
>


The reason of including REGISTER in the charter was not to ratify the =20=

draft word-for-word, but to satisfy the conditions of no dynamic DNS =20
capability on the part of either the service provider or user. I.e., =20
limit the solution space so we did not spend months evaluating an =20
approach with zero chance of deployment.  That said, consensus was the =20=

charter limitation was overly restricted and to remove that limitation =20=

(restricting the solution space to REGIST) from the draft charter.


On Nov 12, 2009, at 5:26 PM, Spencer Dawkins wrote:

> Hi, Shida and Jim,
>
> Thanks for taking notes and publishing them quickly.
>
> What *I* thought happened (very close to the end of the meeting), =20
> was that
>
> - Robert expressed his concern that SIP Forum participation and IETF =20=

> participation didn't overlap, and this was a problem because that's =20=

> the IESG's experience with inter-SDO coordination is that nothing =20
> good happens unless there's sufficient overlap,
>
> - Spencer said something like "we had about 12 (and John said 14) =20
> participants at the last SIPconnect face-to-face, and about 7 were =20
> regular IETF attendees, is that enough overlap?", and
>
> - Robert said he didn't think so.
>
> Others can, of course, confirm (and the audio will probably be =20
> available in a small number of days), but it's going to be real =20
> important to tease a way forward on this, and I don't want the notes =20=

> to give the impression that Robert was convinced when I asked my =20
> question.
>
> Thanks,
>
> Spencer
> ----- Original Message -----
> From: Eric Burger
> To: dispatch@ietf.org
> Sent: Thursday, November 12, 2009 5:14 PM
> Subject: [dispatch] DREGS Ad Hoc minutes
>
> DREGS Adhoc
>
> Thursday 12 November 2009, 11:30 =96 1:00
>
> Chairs: Eric Burger and Mary Barnes
>
> Scribes: Shida Schubert and Jim McEachern
>
> John Elwell =96 Charter Discussion.
>
> http://www.ietf.org/proceedings/09nov/slides/dispatch-6.ppt
>
> Jon Peterson reacted to the John=92s statement that =93decided we =
needed =20
> a new WG=94 and said that he had suggested it perhaps should be in =20
> DRINKS.  Was concerned about the presumption of what needs to be =20
> done to address the open issue.
>
> Cullen:  the purpose of dispatch is to recommend what should be =20
> done.  So we could recommend it go to DRINKS or wherever.  Let=92s see =
=20
> what we recommend.
>
> Clarified that the proposal was that this should be looked at, not =20
> that a WG was needed.
>
> Trying to standardize a uniform way for PBXs to register their =20
> domain.  Today everyone does it their own way, even though they are =20=

> all very similar.
>
> Proposing to use REGISTER with an added header for requesting domain =20=

> registration.  They are proposing this approach because it builds on =20=

> what is already widely deployed.
>
> Adam Roach:  Are we really proposing to write the use of REGISTER in =20=

> the charter?  Answer: Yes.  Adam stated that he was not prepared to =20=

> have that discussion today because he thought it would be discussed =20=

> in the WG.  This feels like =93let=92s ratify this draft=94 rather =
than =20
> =93let=92s work on this problem=94.
>
> Spencer Dawkins: the intent is not to ratify existing draft.
>
> Note:  In spite of Spencer=92s statement, the ensuing discussion made =20=

> it very clear that for at least some of the authors, the intent was =20=

> in fact very much to ratify this draft, or something very near it.  =20=

> Apparently the initial charter draft did not mention REGISTER and it =20=

> was included because SIP Forum participants explicitly asked for it =20=

> to be added.
>
> Eric: the subject of whether or not REGISTER should be included in =20
> the charter has been discussed on the list, but there is not consensus
>
> Someone stated that REGISTER was proposed because =93that is what is =20=

> implemented=94 and we need to align with those implementations to be =20=

> relevant.  Jon countered that earlier it had been stated that the =20
> entire reason for this is that there are multiple ways this is =20
> implemented and that we need a standard to get interoperability.  So =20=

> if it is being done multiple ways today, the argument that REGISTER =20=

> needs to be in the charter to align with deployed equipment, simply =20=

> does not fly.
>
> Cable Labs supported the use of REGISTER because that is what they =20
> use in their specifications.
>
> Jim McEachern & Adam Roach both pointed out that even if you remove =20=

> the word REGISTER from the charter, it does not mean that the =20
> eventual solution will have REGISTER.  The bulk of the objections =20
> (dare I say the allergic reaction?) is to the fact that the solution =20=

> (REGISTER) is being specified in the charter, when we should be =20
> simply defining the problem we are trying to solve.
>
> Anwar Siddiqi Asked about the definition of small =96 medium business, =
=20
> and said that this will not apply to large enterprises since they =20
> will never give their register information to the SP.  Therefore why =20=

> not make it optional.
>
> Markus Isomaki, Nokia:  What does this imply about the addresses of =20=

> the terminals behind the PBX?  Answer: UA registers with the PBX, =20
> but this work deals with the PBX registering with the SP.  It is =20
> designed to help the SP work out where to send requests to =20
> =93example.com=94.
>
> Jon Peterson:  Charter scope question.  Is the charter focused on =20
> the specific problem on the slide, or the more general problem? =20
> (Note: I think this was referring to the bullets specifying =93a =20
> header field for requesting domain registration=94 and =93a SIP option =
=20
> tag to detect non-support of this mechanism=94.)   Answer: The more =20=

> general one.  (Note:  I think this meant the general problem of PBXs =20=

> registering=85)
>
> Jon Peterson:  The only way this charter will get approved is by =20
> taking REGISTER out of the draft.  Lots of nods to that.
>
> Spencer:  If this looks like tweaking existing implementations, it =20
> will get more interest than if it looked like a new method.  =20
> Preference is to be up front about this.
>
> Cullen (channeling Hadriel):  If Hadriel were here he would insist =20
> it must be REGISTER or it will never be deployed.
>
> Cullen: would the possible solution space now include dynamic dns?  =20=

> The only reason to specify a solution in the charter is to =20
> explicitly exclude alternate classes of solution =96 isn=92t it?
>
> Chris Stanley =96 cable labs.  We need to deploy this for businesses =20=

> and need to move quickly.  (And REGISTER is the mechanism we use.)
>
> Adam Roach:  how far are we going to go down this path of specifying =20=

> the answer in the charter?  If we allow it for REGISTER for this WG, =20=

> what other things are we going to do it for?
>
> Cullen as AD:  He was the one who earlier encouraged them to be very =20=

> specific to see if they could get consensus on the approach to =20
> shorten the process, even though he suspected the proposal would get =20=

> the reaction that it is currently getting.  However, if the group =20
> cannot get consensus to specify the solution in the charter, then =20
> the only approach is to remove it.  This is clearly the case.
>
> Hum:  Should we leave the charter exactly as it is?  Result: =20
> Moderate hum
>
> Hum:  Should we leave the charter largely as is,  but remove the =20
> solution (REGISTER) from the charter.  Result: Strong hum.
>
> Discussed a vote on =93Is there a critical mass in the IETF to work on =
=20
> this problem?=94, but instead decided on the following wording.
>
> Hum:  Does anyone think that we don=92t have the critical mass and the =
=20
> expertise to tackle this.  Result:  no one objected.  So consensus =20
> for this.
>
> Hum: is the IETF the place to work on this?  Answer: yes
>
> Note:  The following happened after the meeting had officially =20
> closed, but is included here because it pointed toward a possible =20
> way forward.
>
> Jon: Discussion of the precedent setting risks of just adopting an =20
> existing solution.  One way forward may be to think about this as a =20=

> telephone number problem rather than a domain problem, which would =20
> make it a much easier problem.  If we can scope it that way then it =20=

> will make it a much quicker problem to solve.
>
> Alan Johnston:  All the SIP Forum discussion is in fact about =20
> telephone numbers, so restricting it to telephone numbers is what we =20=

> are looking for
>
> Cullen:  Recognize that we do need to find ways to work faster.  =93If =
=20
> you send me a charter by midnight tonight, I will have it on the =20
> agenda for the next IESG meeting=94
>
> Meeting officially closed.
>
> Since we had time, we had an open discussion of how to make progress=85
>
> Discussion of what would be realistic dates for this?  The =20
> discussion indicated that March and June might be more believable =20
> dates.
>
> Cullen:  Recognize that we do need to find ways to work faster.  =93If =
=20
> you send me a charter by midnight tonight, I will have it on the =20
> agenda for the next IESG meeting=94
>
> Robert Sparks:  Recipe for success in SDOs cooperating is having a =20
> significant overlap of common participants working on the problem.  =20=

> He is concerned that the overlap is not enough.
>
> Spencer:  Last SIP Forum meeting had 12 people and 7 of them were =20
> also in the IETF.  Is that enough?
>
> Jon: Discussion of the precedent setting risks of just adopting an =20
> existing solution.  One way forward may be to think about this as a =20=

> telephone number problem rather than a domain problem, then it is a =20=

> much easier problem.  If we can scope it that way then it will make =20=

> it a much easier problem to solve.
>
> Alan Johnston:  All the SIP Forum discussion is in fact about =20
> telephone numbers, so restricting it to telephone numbers would be =20
> acceptable.
>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Phone numbers and domain registration (was RE: To-URI in domain-registration)
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Elwell, John wrote:
> During the DISPATCH ad hoc session there was indeed a suggestion to limit the mechanism to domains employing E.164-based SIP URIs. However, this would leave an issue with contact URIs, which would presumably not be constrained to being based on telephone numbers. It must be possible to route an out-of-dialog request to a contact URI issued by the SIP-PBX.

Hmm, that's true. And it sheds a different light on the question about 
whether the To: of the register can have a username as well as domain name.

The argument for that has been that then the SP can steer requests to 
different pbxes in the same domain.

I can see how that could work for phone numbers - static configuration 
of those to different "usernames@domain".

I can *possibly* imagine that working for email style URIs too, though 
it would require the SP to provision all the usernames in the domain 
that the PBXes share.

But I really *can't* imagine how that can work for contact addresses 
using the shared domain and some arbitrary username part.

	Thanks,
	Paul

> John
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Richard Shockey
>> Sent: 12 November 2009 20:50
>> To: 'Paul Kyzivat'; 'Middleton, David (NSN - US/Boca Raton)'
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] To-URI in domain-registration
>>
>> To clarify .. in the discussions within the SIPforum there 
>> was no question
>> that the scope was restricted to phone numbers. 
>>
>> here were only 2-3 participants that did not usually attend 
>> IETF meetings
>> and those represented small CLEC's. As a practical matter the overlap
>> between IETF and the SIPforum discussions was nearly total. 
>>
>>
>>
>>>  -----Original Message-----
>>>  From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On
>>>  Behalf Of Paul Kyzivat
>>>  Sent: Thursday, November 12, 2009 8:47 AM
>>>  To: Middleton, David (NSN - US/Boca Raton)
>>>  Cc: dispatch@ietf.org
>>>  Subject: Re: [dispatch] To-URI in domain-registration
>>>  
>>>  
>>>  
>>>  Middleton, David (NSN - US/Boca Raton) wrote:
>>>  > I agree this is possible but having the SP 
>> cross-reference domains
>>>  is not in the spirit of domain registration from my perspective.
>>>  
>>>  But having the SP partition traffic for one domain based 
>> on the user
>>>  part of the request *is* in the spirit? IMO it is exactly the same
>>>  thing, or cleaner.
>>>  
>>>  Its a matter of who is responsible for a particular 
>> domain, the sp or
>>>  the pbx. IMO you should only take *other* information (such as the
>>>  username) into account if you are responsible for the domain of the
>>>  request. In my example, the sp is responsible for 
>> foo.sp.com, but the
>>>  pbxes are responsible for east.foo.sp.com and west.foo.sp.com. The
>>>  other
>>>  way, both the sp and the pbxes are responsible for 
>> foo.sp.com, which
>>>  is
>>>  problematic to me.
>>>  
>>>  I see from the notes of the ad hoc, that there is some desire to
>>>  restrict the scope to phone numbers. In that case it again 
>> is possible
>>>  to give each pbx its own domain name.
>>>  
>>>  	Thanks,
>>>  	Paul
>>>  
>>>  > -----Original Message-----
>>>  > From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>  > Sent: Wednesday, November 11, 2009 5:35 PM
>>>  > To: Middleton, David (NSN - US/Boca Raton)
>>>  > Cc: dispatch@ietf.org
>>>  > Subject: Re: [dispatch] To-URI in domain-registration
>>>  >
>>>  >
>>>  >
>>>  > Middleton, David (NSN - US/Boca Raton) wrote:
>>>  >> Paul,
>>>  >>
>>>  >> Sorry for the delay in responding.
>>>  >>
>>>  >> I think you are on target as far as describing many 
>> (perhaps most)
>>>  SP / PBX interconnect arrangements.  However, the functionality of
>>>  having the SP route calls per user is a real-world scenario which
>>>  proves there is merit for having the capability.
>>>  >
>>>  > This could still be dealt with by having the pbxes register to
>>>  separate
>>>  > sub-domains and the SP knowing (via provisioning) which things to
>>>  route
>>>  > to each.
>>>  >
>>>  > So instead of having:
>>>  >
>>>  > foo-east-pbx register to east@foo.sp.com
>>>  > foo-west-pbx register to west@foo.sp.com
>>>  >
>>>  > you could simply have
>>>  >
>>>  > foo-east-pbx register to east.foo.sp.com
>>>  > foo-west-pbx register to west.foo.sp.com
>>>  >
>>>  > Then the question of how requests addressed to xyz@foo.sp.com are
>>>  mapped
>>>  > to something@east.foo.sp.com or 
>> something@west.foo.sp.com is simply
>>>  out
>>>  > of scope of this document.
>>>  >
>>>  > 	Thanks,
>>>  > 	Paul
>>>  >
>>>  >
>>>  >> Also see inline.
>>>  >>
>>>  >> David
>>>  >>
>>>  >>
>>>  >>
>>>  >> -----Original Message-----
>>>  >> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>  >> Sent: Friday, October 30, 2009 7:37 PM
>>>  >> To: Middleton, David (NSN - US/Boca Raton)
>>>  >> Cc: dispatch@ietf.org
>>>  >> Subject: Re: [dispatch] To-URI in domain-registration
>>>  >>
>>>  >> David,
>>>  >>
>>>  >> IIUC, you are proposing a case such as:
>>>  >>
>>>  >> - a customer of sp.net has domain cust.sp.net
>>>  >> - the customer has two PBX systems, east and west.
>>>  >> - you propose that they register, respectively
>>>  >>    To: east@cust.sp.net, and
>>>  >>    To: west@cust.sp.net.
>>>  >> - then numbers assigned to the customer are routed
>>>  >>    something like:
>>>  >>    1-212-555-88xx -> contact from east@cust.sp.net
>>>  >>    1-408-555-88xx -> contact from west@cust.sp.net
>>>  >> - also, routing for names be something like:
>>>  >>    bob@cust.sp.net   -> contact from east@cust.sp.net
>>>  >>    carol@cust.sp.net -> "
>>>  >>    ted@cust.sp.net   -> contact from west@cust.sp.net
>>>  >>    alice@cust.sp.net -> "
>>>  >>    others...
>>>  >>
>>>  >> Is that right?
>>>  >> <DM> Yes
>>>  >>
>>>  >> You then can't describe that as setting up routing paths for
>>>  >> cust.sp.net. You are instead describing some entirely different
>>>  routing
>>>  >> algorithm based on the user part.
>>>  >> <DM>  Yes, it requires a local policy to determine the 
>> location of
>>>  the users.
>>>  >>
>>>  >> Subdomains east.cust.sp.net and west.cust.sp.net solve 
>> this problem
>>>  in
>>>  >> large part - at least for the numbers.
>>>  >>
>>>  >> You clearly don't find that acceptable for the names. If those
>>>  names
>>>  >> were exposed to end users I wouldn't find it acceptable 
>> either. (I
>>>  want
>>>  >> to be pkyzivat@cisco.com, not pkyzivat@east.cisco.com.)
>>>  >> <DM> Agreed
>>>  >>
>>>  >> The question is whether this is the best way to solve 
>> that problem.
>>>  >>
>>>  >> I think that solution won't be very acceptable to an 
>> SP, or to the
>>>  >> customer of that SP. Having the SP provision every one of your
>>>  users
>>>  >> just isn't going to be popular with anyone. (Unless the 
>> SP gets a
>>>  big
>>>  >> enough per-user fee.)
>>>  >> <DM> I agree but if the parties involved reach an acceptable
>>>  agreement, the technology should support it IMO.
>>>  >>
>>>  >> Its not quite as much of a problem with numbers, 
>> because they are
>>>  likely
>>>  >> to be managed in blocks. And the numbers are really 
>> owned/managed
>>>  by the
>>>  >> SP anyway so they are forced to do something about them.
>>>  >>
>>>  >> This is also a problem that is not unique to this SP 
>> interconnect
>>>  >> mechanism. It is pretty much the same problem in any big company
>>>  with a
>>>  >> lot of sites, even if they have a private network connecting all
>>>  the sites.
>>>  >>
>>>  >> What you need in this case is a proxy at the company 
>> domain level
>>>  >> (cust.sp.net in this case perhaps) that has the roster of all
>>>  usernames
>>>  >> and a mapping to a subdomain - mapping bob@cust.sp.net to
>>>  >> bob@east.cust.sp.net.
>>>  >> <DM> Yes but some enterprise customers do not want to 
>> do this and
>>>  are able to convince the SP to do this.
>>>  >>
>>>  >> Then, the company has one server registering To:cust.sp.net to
>>>  provide
>>>  >> this mapping, and other servers registering To:east.cust.sp.net,
>>>  ...
>>>  >>
>>>  >> 	Thanks,
>>>  >> 	Paul
>>>  >>
>>>  >>
>>>  >> Middleton, David (NSN - US/Boca Raton) wrote:
>>>  >>> Some colleagues have pointed out an aspect that I had not
>>>  considered concerning the domain registration To: user part.  There
>>>  are cases where an Enterprise has multiple PBXs that register and
>>>  calls should be routed to one of the PBXs based on which user is
>>>  called.  In other words, instead of the Enterprise with 
>> multiple PBXs
>>>  appearing as one logical PBX, the Enterprise customer wants the
>>>  service provider to take responsibility for distributing 
>> the calls to
>>>  the specific PBX that hosts the called user.
>>>  >>>
>>>  >>> While this problem could be solved with a local policy 
>> to choose
>>>  one of several contacts it is not as clean an approach as 
>> being able
>>>  to treat each registration as a separate entity with it's own
>>>  contact(s).  While one could propose that these be treated 
>> separately
>>>  by making different sub-domains, the Enterprise most assuredly will
>>>  not want to publish this sub-domain on business cards, 
>> etc. when using
>>>  e-mail style addresses.  Besides, it would cause the 
>> address to change
>>>  if the user moves from one PBX to another as might happen with a
>>>  physical move.
>>>  >>>
>>>  >>> Another aspect to consider is that the Enterprise user, in this
>>>  scenario, is only reachable via one registered "To: user@host".  So
>>>  not having separate registrations would result in a more complex
>>>  definition and implementation of things like (a.) is there an
>>>  unexpired binding and (b.) picking appropriate contacts by q-value.
>>>  >>>
>>>  >>> I think this makes a strong case for having a user 
>> part in the To:
>>>  header and treating each unique user@host as a separate 
>> registration à
>>>  la RFC 3264.
>>>  >>>
>>>  >>> In summary, I support Option 1) as well.
>>>  >>>
>>>  >>> - David
>>>  >>>
>>>  >>>
>>>  >>>
>>>  >>> -----Original Message-----
>>>  >>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org]
>>>  On Behalf Of ext Richard Shockey
>>>  >>> Sent: Wednesday, October 21, 2009 4:42 PM
>>>  >>> To: dispatch@ietf.org
>>>  >>> Subject: Re: [dispatch] To-URI in domain-registration
>>>  >>>
>>>  >>> +1 to Option 1) as well
>>>  >>>
>>>  >>>>  -----Original Message-----
>>>  >>>>  From: dispatch-bounces@ietf.org [mailto:dispatch-
>>>  bounces@ietf.org] On
>>>  >>>>  Behalf Of David Hancock
>>>  >>>>  Sent: Wednesday, October 21, 2009 4:05 PM
>>>  >>>>  To: Brian Lindsay; Hadriel Kaplan; dispatch@ietf.org
>>>  >>>>  Subject: Re: [dispatch] To-URI in domain-registration
>>>  >>>>
>>>  >>>>
>>>  >>>>  Hadriel - I'd prefer option 1) as well.
>>>  >>>>
>>>  >>>>  David
>>>  >>>>
>>>  >>>>  > -----Original Message-----
>>>  >>>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-
>>>  bounces@ietf.org]
>>>  >>>>  On
>>>  >>>>  > Behalf Of Brian Lindsay
>>>  >>>>  > Sent: Wednesday, October 21, 2009 1:53 PM
>>>  >>>>  > To: Hadriel Kaplan; dispatch@ietf.org
>>>  >>>>  > Subject: Re: [dispatch] To-URI in domain-registration
>>>  >>>>  >
>>>  >>>>  >  Hi Hadriel,
>>>  >>>>  >
>>>  >>>>  >     Per my comment yesterday, I'd prefer (1).
>>>  >>>>  >
>>>  >>>>  >     Thanks.
>>>  >>>>  >
>>>  >>>>  > -----Original Message-----
>>>  >>>>  > From: dispatch-bounces@ietf.org [mailto:dispatch-
>>>  bounces@ietf.org]
>>>  >>>>  On
>>>  >>>>  > Behalf Of Hadriel Kaplan
>>>  >>>>  > Sent: Wednesday, October 21, 2009 3:43 PM
>>>  >>>>  > To: dispatch@ietf.org
>>>  >>>>  > Subject: [dispatch] To-URI in domain-registration
>>>  >>>>  >
>>>  >>>>  > Howdy,
>>>  >>>>  > I'd like to get a consensus call on the question 
>> of what the
>>>  To-URI
>>>  >>>>  > should be for the REGISTER request of a domain 
>> registration,
>>>  before
>>>  >>>>  > finishing the next rev of the draft.  This will 
>> affect quite a
>>>  bit
>>>  >>>>  of
>>>  >>>>  > text in the draft (e.g., examples), and I think that the
>>>  decision
>>>  >>>>  about
>>>  >>>>  > that will also affect whether we need more than a
>>>  Require:option-tag
>>>  >>>>  as
>>>  >>>>  > well.
>>>  >>>>  >
>>>  >>>>  > The choices right now are:
>>>  >>>>  > 1) Mandate the To-URI be in the syntactical form of an AoR,
>>>  >>>>  including
>>>  >>>>  > the user@ portion, the same as the From-URI.
>>>  >>>>  > 	Pros: closer to what current proprietary and other-SDO
>>>  >>>>  > mechanisms do, on the wire.  Therefore, presumably 
>> less of a
>>>  change
>>>  >>>>  and
>>>  >>>>  > less barrier to adoption.
>>>  >>>>  > 2) Mandate the To-URI be a domain only (no user@), 
>> while the
>>>  From-
>>>  >>>>  URI
>>>  >>>>  > remains the "AoR" of the PBX registering.
>>>  >>>>  > 	Pros: semantically accurate (you're registering 
>> a domain,
>>>  not an
>>>  >>>>  > AoR), and may obviate the need for a new header or param to
>>>  indicate
>>>  >>>>  > this new mechanism is being used.
>>>  >>>>  >
>>>  >>>>  > Can people please indicate which they'd prefer?
>>>  >>>>  >
>>>  >>>>  > Thanks!
>>>  >>>>  > -hadriel
>>>  >>>>  > _______________________________________________
>>>  >>>>  > dispatch mailing list
>>>  >>>>  > dispatch@ietf.org
>>>  >>>>  > https://www.ietf.org/mailman/listinfo/dispatch
>>>  >>>>  > _______________________________________________
>>>  >>>>  > dispatch mailing list
>>>  >>>>  > dispatch@ietf.org
>>>  >>>>  > https://www.ietf.org/mailman/listinfo/dispatch
>>>  >>>>  _______________________________________________
>>>  >>>>  dispatch mailing list
>>>  >>>>  dispatch@ietf.org
>>>  >>>>  https://www.ietf.org/mailman/listinfo/dispatch
>>>  >>> _______________________________________________
>>>  >>> dispatch mailing list
>>>  >>> dispatch@ietf.org
>>>  >>> https://www.ietf.org/mailman/listinfo/dispatch
>>>  >>> _______________________________________________
>>>  >>> dispatch mailing list
>>>  >>> dispatch@ietf.org
>>>  >>> https://www.ietf.org/mailman/listinfo/dispatch
>>>  >>>
>>>  >
>>>  _______________________________________________
>>>  dispatch mailing list
>>>  dispatch@ietf.org
>>>  https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>

From mary.barnes@nortel.com  Thu Nov 12 14:32:10 2009
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The minutes on the server have already been updated to remove the =
"Note:" .


Mary.

-----Original Message-----
From: dispatch-bounces@ietf.org on behalf of Eric Burger
Sent: Thu 11/12/2009 4:17 PM
To: dispatch@ietf.org
Subject: Re: [dispatch] DREGS Ad Hoc minutes
=20
Likewise, I do not think the Note in the statement below is accurate.
> Adam Roach:  Are we really proposing to write the use of REGISTER in =20
> the charter?  Answer: Yes.  Adam stated that he was not prepared to =20
> have that discussion today because he thought it would be discussed =20
> in the WG.  This feels like "let's ratify this draft" rather than =20
> "let's work on this problem".
>
>
> Spencer Dawkins: the intent is not to ratify existing draft.
>
> Note:  In spite of Spencer's statement, the ensuing discussion made =20
> it very clear that for at least some of the authors, the intent was =20
> in fact very much to ratify this draft, or something very near it.  =20
> Apparently the initial charter draft did not mention REGISTER and it =20
> was included because SIP Forum participants explicitly asked for it =20
> to be added.
>
>
> Eric: the subject of whether or not REGISTER should be included in =20
> the charter has been discussed on the list, but there is not consensus
>


The reason of including REGISTER in the charter was not to ratify the =20
draft word-for-word, but to satisfy the conditions of no dynamic DNS =20
capability on the part of either the service provider or user. I.e., =20
limit the solution space so we did not spend months evaluating an =20
approach with zero chance of deployment.  That said, consensus was the =20
charter limitation was overly restricted and to remove that limitation =20
(restricting the solution space to REGIST) from the draft charter.


On Nov 12, 2009, at 5:26 PM, Spencer Dawkins wrote:

> Hi, Shida and Jim,
>
> Thanks for taking notes and publishing them quickly.
>
> What *I* thought happened (very close to the end of the meeting), =20
> was that
>
> - Robert expressed his concern that SIP Forum participation and IETF =20
> participation didn't overlap, and this was a problem because that's =20
> the IESG's experience with inter-SDO coordination is that nothing =20
> good happens unless there's sufficient overlap,
>
> - Spencer said something like "we had about 12 (and John said 14) =20
> participants at the last SIPconnect face-to-face, and about 7 were =20
> regular IETF attendees, is that enough overlap?", and
>
> - Robert said he didn't think so.
>
> Others can, of course, confirm (and the audio will probably be =20
> available in a small number of days), but it's going to be real =20
> important to tease a way forward on this, and I don't want the notes =20
> to give the impression that Robert was convinced when I asked my =20
> question.
>
> Thanks,
>
> Spencer
> ----- Original Message -----
> From: Eric Burger
> To: dispatch@ietf.org
> Sent: Thursday, November 12, 2009 5:14 PM
> Subject: [dispatch] DREGS Ad Hoc minutes
>
> DREGS Adhoc
>
> Thursday 12 November 2009, 11:30 - 1:00
>
> Chairs: Eric Burger and Mary Barnes
>
> Scribes: Shida Schubert and Jim McEachern
>
> John Elwell - Charter Discussion.
>
> http://www.ietf.org/proceedings/09nov/slides/dispatch-6.ppt
>
> Jon Peterson reacted to the John's statement that "decided we needed =20
> a new WG" and said that he had suggested it perhaps should be in =20
> DRINKS.  Was concerned about the presumption of what needs to be =20
> done to address the open issue.
>
> Cullen:  the purpose of dispatch is to recommend what should be =20
> done.  So we could recommend it go to DRINKS or wherever.  Let's see =20
> what we recommend.
>
> Clarified that the proposal was that this should be looked at, not =20
> that a WG was needed.
>
> Trying to standardize a uniform way for PBXs to register their =20
> domain.  Today everyone does it their own way, even though they are =20
> all very similar.
>
> Proposing to use REGISTER with an added header for requesting domain =20
> registration.  They are proposing this approach because it builds on =20
> what is already widely deployed.
>
> Adam Roach:  Are we really proposing to write the use of REGISTER in =20
> the charter?  Answer: Yes.  Adam stated that he was not prepared to =20
> have that discussion today because he thought it would be discussed =20
> in the WG.  This feels like "let's ratify this draft" rather than =20
> "let's work on this problem".
>
> Spencer Dawkins: the intent is not to ratify existing draft.
>
> Note:  In spite of Spencer's statement, the ensuing discussion made =20
> it very clear that for at least some of the authors, the intent was =20
> in fact very much to ratify this draft, or something very near it.  =20
> Apparently the initial charter draft did not mention REGISTER and it =20
> was included because SIP Forum participants explicitly asked for it =20
> to be added.
>
> Eric: the subject of whether or not REGISTER should be included in =20
> the charter has been discussed on the list, but there is not consensus
>
> Someone stated that REGISTER was proposed because "that is what is =20
> implemented" and we need to align with those implementations to be =20
> relevant.  Jon countered that earlier it had been stated that the =20
> entire reason for this is that there are multiple ways this is =20
> implemented and that we need a standard to get interoperability.  So =20
> if it is being done multiple ways today, the argument that REGISTER =20
> needs to be in the charter to align with deployed equipment, simply =20
> does not fly.
>
> Cable Labs supported the use of REGISTER because that is what they =20
> use in their specifications.
>
> Jim McEachern & Adam Roach both pointed out that even if you remove =20
> the word REGISTER from the charter, it does not mean that the =20
> eventual solution will have REGISTER.  The bulk of the objections =20
> (dare I say the allergic reaction?) is to the fact that the solution =20
> (REGISTER) is being specified in the charter, when we should be =20
> simply defining the problem we are trying to solve.
>
> Anwar Siddiqi Asked about the definition of small - medium business, =20
> and said that this will not apply to large enterprises since they =20
> will never give their register information to the SP.  Therefore why =20
> not make it optional.
>
> Markus Isomaki, Nokia:  What does this imply about the addresses of =20
> the terminals behind the PBX?  Answer: UA registers with the PBX, =20
> but this work deals with the PBX registering with the SP.  It is =20
> designed to help the SP work out where to send requests to =20
> "example.com".
>
> Jon Peterson:  Charter scope question.  Is the charter focused on =20
> the specific problem on the slide, or the more general problem? =20
> (Note: I think this was referring to the bullets specifying "a =20
> header field for requesting domain registration" and "a SIP option =20
> tag to detect non-support of this mechanism".)   Answer: The more =20
> general one.  (Note:  I think this meant the general problem of PBXs =20
> registering.)
>
> Jon Peterson:  The only way this charter will get approved is by =20
> taking REGISTER out of the draft.  Lots of nods to that.
>
> Spencer:  If this looks like tweaking existing implementations, it =20
> will get more interest than if it looked like a new method.  =20
> Preference is to be up front about this.
>
> Cullen (channeling Hadriel):  If Hadriel were here he would insist =20
> it must be REGISTER or it will never be deployed.
>
> Cullen: would the possible solution space now include dynamic dns?  =20
> The only reason to specify a solution in the charter is to =20
> explicitly exclude alternate classes of solution - isn't it?
>
> Chris Stanley - cable labs.  We need to deploy this for businesses =20
> and need to move quickly.  (And REGISTER is the mechanism we use.)
>
> Adam Roach:  how far are we going to go down this path of specifying =20
> the answer in the charter?  If we allow it for REGISTER for this WG, =20
> what other things are we going to do it for?
>
> Cullen as AD:  He was the one who earlier encouraged them to be very =20
> specific to see if they could get consensus on the approach to =20
> shorten the process, even though he suspected the proposal would get =20
> the reaction that it is currently getting.  However, if the group =20
> cannot get consensus to specify the solution in the charter, then =20
> the only approach is to remove it.  This is clearly the case.
>
> Hum:  Should we leave the charter exactly as it is?  Result: =20
> Moderate hum
>
> Hum:  Should we leave the charter largely as is,  but remove the =20
> solution (REGISTER) from the charter.  Result: Strong hum.
>
> Discussed a vote on "Is there a critical mass in the IETF to work on =20
> this problem?", but instead decided on the following wording.
>
> Hum:  Does anyone think that we don't have the critical mass and the =20
> expertise to tackle this.  Result:  no one objected.  So consensus =20
> for this.
>
> Hum: is the IETF the place to work on this?  Answer: yes
>
> Note:  The following happened after the meeting had officially =20
> closed, but is included here because it pointed toward a possible =20
> way forward.
>
> Jon: Discussion of the precedent setting risks of just adopting an =20
> existing solution.  One way forward may be to think about this as a =20
> telephone number problem rather than a domain problem, which would =20
> make it a much easier problem.  If we can scope it that way then it =20
> will make it a much quicker problem to solve.
>
> Alan Johnston:  All the SIP Forum discussion is in fact about =20
> telephone numbers, so restricting it to telephone numbers is what we =20
> are looking for
>
> Cullen:  Recognize that we do need to find ways to work faster.  "If =20
> you send me a charter by midnight tonight, I will have it on the =20
> agenda for the next IESG meeting"
>
> Meeting officially closed.
>
> Since we had time, we had an open discussion of how to make progress.
>
> Discussion of what would be realistic dates for this?  The =20
> discussion indicated that March and June might be more believable =20
> dates.
>
> Cullen:  Recognize that we do need to find ways to work faster.  "If =20
> you send me a charter by midnight tonight, I will have it on the =20
> agenda for the next IESG meeting"
>
> Robert Sparks:  Recipe for success in SDOs cooperating is having a =20
> significant overlap of common participants working on the problem.  =20
> He is concerned that the overlap is not enough.
>
> Spencer:  Last SIP Forum meeting had 12 people and 7 of them were =20
> also in the IETF.  Is that enough?
>
> Jon: Discussion of the precedent setting risks of just adopting an =20
> existing solution.  One way forward may be to think about this as a =20
> telephone number problem rather than a domain problem, then it is a =20
> much easier problem.  If we can scope it that way then it will make =20
> it a much easier problem to solve.
>
> Alan Johnston:  All the SIP Forum discussion is in fact about =20
> telephone numbers, so restricting it to telephone numbers would be =20
> acceptable.
>
>
>
> _______________________________________________
> 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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<P><FONT SIZE=3D2>The minutes on the server have already been updated to =
remove the &quot;Note:&quot; .<BR>
<BR>
<BR>
Mary.<BR>
<BR>
-----Original Message-----<BR>
From: dispatch-bounces@ietf.org on behalf of Eric Burger<BR>
Sent: Thu 11/12/2009 4:17 PM<BR>
To: dispatch@ietf.org<BR>
Subject: Re: [dispatch] DREGS Ad Hoc minutes<BR>
<BR>
Likewise, I do not think the Note in the statement below is =
accurate.<BR>
&gt; Adam Roach:&nbsp; Are we really proposing to write the use of =
REGISTER in&nbsp;<BR>
&gt; the charter?&nbsp; Answer: Yes.&nbsp; Adam stated that he was not =
prepared to&nbsp;<BR>
&gt; have that discussion today because he thought it would be =
discussed&nbsp;<BR>
&gt; in the WG.&nbsp; This feels like &quot;let's ratify this =
draft&quot; rather than&nbsp;<BR>
&gt; &quot;let's work on this problem&quot;.<BR>
&gt;<BR>
&gt;<BR>
&gt; Spencer Dawkins: the intent is not to ratify existing draft.<BR>
&gt;<BR>
&gt; Note:&nbsp; In spite of Spencer's statement, the ensuing discussion =
made&nbsp;<BR>
&gt; it very clear that for at least some of the authors, the intent =
was&nbsp;<BR>
&gt; in fact very much to ratify this draft, or something very near =
it.&nbsp;&nbsp;<BR>
&gt; Apparently the initial charter draft did not mention REGISTER and =
it&nbsp;<BR>
&gt; was included because SIP Forum participants explicitly asked for =
it&nbsp;<BR>
&gt; to be added.<BR>
&gt;<BR>
&gt;<BR>
&gt; Eric: the subject of whether or not REGISTER should be included =
in&nbsp;<BR>
&gt; the charter has been discussed on the list, but there is not =
consensus<BR>
&gt;<BR>
<BR>
<BR>
The reason of including REGISTER in the charter was not to ratify =
the&nbsp;<BR>
draft word-for-word, but to satisfy the conditions of no dynamic =
DNS&nbsp;<BR>
capability on the part of either the service provider or user. =
I.e.,&nbsp;<BR>
limit the solution space so we did not spend months evaluating =
an&nbsp;<BR>
approach with zero chance of deployment.&nbsp; That said, consensus was =
the&nbsp;<BR>
charter limitation was overly restricted and to remove that =
limitation&nbsp;<BR>
(restricting the solution space to REGIST) from the draft charter.<BR>
<BR>
<BR>
On Nov 12, 2009, at 5:26 PM, Spencer Dawkins wrote:<BR>
<BR>
&gt; Hi, Shida and Jim,<BR>
&gt;<BR>
&gt; Thanks for taking notes and publishing them quickly.<BR>
&gt;<BR>
&gt; What *I* thought happened (very close to the end of the =
meeting),&nbsp;<BR>
&gt; was that<BR>
&gt;<BR>
&gt; - Robert expressed his concern that SIP Forum participation and =
IETF&nbsp;<BR>
&gt; participation didn't overlap, and this was a problem because =
that's&nbsp;<BR>
&gt; the IESG's experience with inter-SDO coordination is that =
nothing&nbsp;<BR>
&gt; good happens unless there's sufficient overlap,<BR>
&gt;<BR>
&gt; - Spencer said something like &quot;we had about 12 (and John said =
14)&nbsp;<BR>
&gt; participants at the last SIPconnect face-to-face, and about 7 =
were&nbsp;<BR>
&gt; regular IETF attendees, is that enough overlap?&quot;, and<BR>
&gt;<BR>
&gt; - Robert said he didn't think so.<BR>
&gt;<BR>
&gt; Others can, of course, confirm (and the audio will probably =
be&nbsp;<BR>
&gt; available in a small number of days), but it's going to be =
real&nbsp;<BR>
&gt; important to tease a way forward on this, and I don't want the =
notes&nbsp;<BR>
&gt; to give the impression that Robert was convinced when I asked =
my&nbsp;<BR>
&gt; question.<BR>
&gt;<BR>
&gt; Thanks,<BR>
&gt;<BR>
&gt; Spencer<BR>
&gt; ----- Original Message -----<BR>
&gt; From: Eric Burger<BR>
&gt; To: dispatch@ietf.org<BR>
&gt; Sent: Thursday, November 12, 2009 5:14 PM<BR>
&gt; Subject: [dispatch] DREGS Ad Hoc minutes<BR>
&gt;<BR>
&gt; DREGS Adhoc<BR>
&gt;<BR>
&gt; Thursday 12 November 2009, 11:30 - 1:00<BR>
&gt;<BR>
&gt; Chairs: Eric Burger and Mary Barnes<BR>
&gt;<BR>
&gt; Scribes: Shida Schubert and Jim McEachern<BR>
&gt;<BR>
&gt; John Elwell - Charter Discussion.<BR>
&gt;<BR>
&gt; <A =
HREF=3D"http://www.ietf.org/proceedings/09nov/slides/dispatch-6.ppt">http=
://www.ietf.org/proceedings/09nov/slides/dispatch-6.ppt</A><BR>
&gt;<BR>
&gt; Jon Peterson reacted to the John's statement that &quot;decided we =
needed&nbsp;<BR>
&gt; a new WG&quot; and said that he had suggested it perhaps should be =
in&nbsp;<BR>
&gt; DRINKS.&nbsp; Was concerned about the presumption of what needs to =
be&nbsp;<BR>
&gt; done to address the open issue.<BR>
&gt;<BR>
&gt; Cullen:&nbsp; the purpose of dispatch is to recommend what should =
be&nbsp;<BR>
&gt; done.&nbsp; So we could recommend it go to DRINKS or =
wherever.&nbsp; Let's see&nbsp;<BR>
&gt; what we recommend.<BR>
&gt;<BR>
&gt; Clarified that the proposal was that this should be looked at, =
not&nbsp;<BR>
&gt; that a WG was needed.<BR>
&gt;<BR>
&gt; Trying to standardize a uniform way for PBXs to register =
their&nbsp;<BR>
&gt; domain.&nbsp; Today everyone does it their own way, even though =
they are&nbsp;<BR>
&gt; all very similar.<BR>
&gt;<BR>
&gt; Proposing to use REGISTER with an added header for requesting =
domain&nbsp;<BR>
&gt; registration.&nbsp; They are proposing this approach because it =
builds on&nbsp;<BR>
&gt; what is already widely deployed.<BR>
&gt;<BR>
&gt; Adam Roach:&nbsp; Are we really proposing to write the use of =
REGISTER in&nbsp;<BR>
&gt; the charter?&nbsp; Answer: Yes.&nbsp; Adam stated that he was not =
prepared to&nbsp;<BR>
&gt; have that discussion today because he thought it would be =
discussed&nbsp;<BR>
&gt; in the WG.&nbsp; This feels like &quot;let's ratify this =
draft&quot; rather than&nbsp;<BR>
&gt; &quot;let's work on this problem&quot;.<BR>
&gt;<BR>
&gt; Spencer Dawkins: the intent is not to ratify existing draft.<BR>
&gt;<BR>
&gt; Note:&nbsp; In spite of Spencer's statement, the ensuing discussion =
made&nbsp;<BR>
&gt; it very clear that for at least some of the authors, the intent =
was&nbsp;<BR>
&gt; in fact very much to ratify this draft, or something very near =
it.&nbsp;&nbsp;<BR>
&gt; Apparently the initial charter draft did not mention REGISTER and =
it&nbsp;<BR>
&gt; was included because SIP Forum participants explicitly asked for =
it&nbsp;<BR>
&gt; to be added.<BR>
&gt;<BR>
&gt; Eric: the subject of whether or not REGISTER should be included =
in&nbsp;<BR>
&gt; the charter has been discussed on the list, but there is not =
consensus<BR>
&gt;<BR>
&gt; Someone stated that REGISTER was proposed because &quot;that is =
what is&nbsp;<BR>
&gt; implemented&quot; and we need to align with those implementations =
to be&nbsp;<BR>
&gt; relevant.&nbsp; Jon countered that earlier it had been stated that =
the&nbsp;<BR>
&gt; entire reason for this is that there are multiple ways this =
is&nbsp;<BR>
&gt; implemented and that we need a standard to get =
interoperability.&nbsp; So&nbsp;<BR>
&gt; if it is being done multiple ways today, the argument that =
REGISTER&nbsp;<BR>
&gt; needs to be in the charter to align with deployed equipment, =
simply&nbsp;<BR>
&gt; does not fly.<BR>
&gt;<BR>
&gt; Cable Labs supported the use of REGISTER because that is what =
they&nbsp;<BR>
&gt; use in their specifications.<BR>
&gt;<BR>
&gt; Jim McEachern &amp; Adam Roach both pointed out that even if you =
remove&nbsp;<BR>
&gt; the word REGISTER from the charter, it does not mean that =
the&nbsp;<BR>
&gt; eventual solution will have REGISTER.&nbsp; The bulk of the =
objections&nbsp;<BR>
&gt; (dare I say the allergic reaction?) is to the fact that the =
solution&nbsp;<BR>
&gt; (REGISTER) is being specified in the charter, when we should =
be&nbsp;<BR>
&gt; simply defining the problem we are trying to solve.<BR>
&gt;<BR>
&gt; Anwar Siddiqi Asked about the definition of small - medium =
business,&nbsp;<BR>
&gt; and said that this will not apply to large enterprises since =
they&nbsp;<BR>
&gt; will never give their register information to the SP.&nbsp; =
Therefore why&nbsp;<BR>
&gt; not make it optional.<BR>
&gt;<BR>
&gt; Markus Isomaki, Nokia:&nbsp; What does this imply about the =
addresses of&nbsp;<BR>
&gt; the terminals behind the PBX?&nbsp; Answer: UA registers with the =
PBX,&nbsp;<BR>
&gt; but this work deals with the PBX registering with the SP.&nbsp; It =
is&nbsp;<BR>
&gt; designed to help the SP work out where to send requests =
to&nbsp;<BR>
&gt; &quot;example.com&quot;.<BR>
&gt;<BR>
&gt; Jon Peterson:&nbsp; Charter scope question.&nbsp; Is the charter =
focused on&nbsp;<BR>
&gt; the specific problem on the slide, or the more general =
problem?&nbsp;<BR>
&gt; (Note: I think this was referring to the bullets specifying =
&quot;a&nbsp;<BR>
&gt; header field for requesting domain registration&quot; and &quot;a =
SIP option&nbsp;<BR>
&gt; tag to detect non-support of this mechanism&quot;.)&nbsp;&nbsp; =
Answer: The more&nbsp;<BR>
&gt; general one.&nbsp; (Note:&nbsp; I think this meant the general =
problem of PBXs&nbsp;<BR>
&gt; registering.)<BR>
&gt;<BR>
&gt; Jon Peterson:&nbsp; The only way this charter will get approved is =
by&nbsp;<BR>
&gt; taking REGISTER out of the draft.&nbsp; Lots of nods to that.<BR>
&gt;<BR>
&gt; Spencer:&nbsp; If this looks like tweaking existing =
implementations, it&nbsp;<BR>
&gt; will get more interest than if it looked like a new =
method.&nbsp;&nbsp;<BR>
&gt; Preference is to be up front about this.<BR>
&gt;<BR>
&gt; Cullen (channeling Hadriel):&nbsp; If Hadriel were here he would =
insist&nbsp;<BR>
&gt; it must be REGISTER or it will never be deployed.<BR>
&gt;<BR>
&gt; Cullen: would the possible solution space now include dynamic =
dns?&nbsp;&nbsp;<BR>
&gt; The only reason to specify a solution in the charter is =
to&nbsp;<BR>
&gt; explicitly exclude alternate classes of solution - isn't it?<BR>
&gt;<BR>
&gt; Chris Stanley - cable labs.&nbsp; We need to deploy this for =
businesses&nbsp;<BR>
&gt; and need to move quickly.&nbsp; (And REGISTER is the mechanism we =
use.)<BR>
&gt;<BR>
&gt; Adam Roach:&nbsp; how far are we going to go down this path of =
specifying&nbsp;<BR>
&gt; the answer in the charter?&nbsp; If we allow it for REGISTER for =
this WG,&nbsp;<BR>
&gt; what other things are we going to do it for?<BR>
&gt;<BR>
&gt; Cullen as AD:&nbsp; He was the one who earlier encouraged them to =
be very&nbsp;<BR>
&gt; specific to see if they could get consensus on the approach =
to&nbsp;<BR>
&gt; shorten the process, even though he suspected the proposal would =
get&nbsp;<BR>
&gt; the reaction that it is currently getting.&nbsp; However, if the =
group&nbsp;<BR>
&gt; cannot get consensus to specify the solution in the charter, =
then&nbsp;<BR>
&gt; the only approach is to remove it.&nbsp; This is clearly the =
case.<BR>
&gt;<BR>
&gt; Hum:&nbsp; Should we leave the charter exactly as it is?&nbsp; =
Result:&nbsp;<BR>
&gt; Moderate hum<BR>
&gt;<BR>
&gt; Hum:&nbsp; Should we leave the charter largely as is,&nbsp; but =
remove the&nbsp;<BR>
&gt; solution (REGISTER) from the charter.&nbsp; Result: Strong hum.<BR>
&gt;<BR>
&gt; Discussed a vote on &quot;Is there a critical mass in the IETF to =
work on&nbsp;<BR>
&gt; this problem?&quot;, but instead decided on the following =
wording.<BR>
&gt;<BR>
&gt; Hum:&nbsp; Does anyone think that we don't have the critical mass =
and the&nbsp;<BR>
&gt; expertise to tackle this.&nbsp; Result:&nbsp; no one =
objected.&nbsp; So consensus&nbsp;<BR>
&gt; for this.<BR>
&gt;<BR>
&gt; Hum: is the IETF the place to work on this?&nbsp; Answer: yes<BR>
&gt;<BR>
&gt; Note:&nbsp; The following happened after the meeting had =
officially&nbsp;<BR>
&gt; closed, but is included here because it pointed toward a =
possible&nbsp;<BR>
&gt; way forward.<BR>
&gt;<BR>
&gt; Jon: Discussion of the precedent setting risks of just adopting =
an&nbsp;<BR>
&gt; existing solution.&nbsp; One way forward may be to think about this =
as a&nbsp;<BR>
&gt; telephone number problem rather than a domain problem, which =
would&nbsp;<BR>
&gt; make it a much easier problem.&nbsp; If we can scope it that way =
then it&nbsp;<BR>
&gt; will make it a much quicker problem to solve.<BR>
&gt;<BR>
&gt; Alan Johnston:&nbsp; All the SIP Forum discussion is in fact =
about&nbsp;<BR>
&gt; telephone numbers, so restricting it to telephone numbers is what =
we&nbsp;<BR>
&gt; are looking for<BR>
&gt;<BR>
&gt; Cullen:&nbsp; Recognize that we do need to find ways to work =
faster.&nbsp; &quot;If&nbsp;<BR>
&gt; you send me a charter by midnight tonight, I will have it on =
the&nbsp;<BR>
&gt; agenda for the next IESG meeting&quot;<BR>
&gt;<BR>
&gt; Meeting officially closed.<BR>
&gt;<BR>
&gt; Since we had time, we had an open discussion of how to make =
progress.<BR>
&gt;<BR>
&gt; Discussion of what would be realistic dates for this?&nbsp; =
The&nbsp;<BR>
&gt; discussion indicated that March and June might be more =
believable&nbsp;<BR>
&gt; dates.<BR>
&gt;<BR>
&gt; Cullen:&nbsp; Recognize that we do need to find ways to work =
faster.&nbsp; &quot;If&nbsp;<BR>
&gt; you send me a charter by midnight tonight, I will have it on =
the&nbsp;<BR>
&gt; agenda for the next IESG meeting&quot;<BR>
&gt;<BR>
&gt; Robert Sparks:&nbsp; Recipe for success in SDOs cooperating is =
having a&nbsp;<BR>
&gt; significant overlap of common participants working on the =
problem.&nbsp;&nbsp;<BR>
&gt; He is concerned that the overlap is not enough.<BR>
&gt;<BR>
&gt; Spencer:&nbsp; Last SIP Forum meeting had 12 people and 7 of them =
were&nbsp;<BR>
&gt; also in the IETF.&nbsp; Is that enough?<BR>
&gt;<BR>
&gt; Jon: Discussion of the precedent setting risks of just adopting =
an&nbsp;<BR>
&gt; existing solution.&nbsp; One way forward may be to think about this =
as a&nbsp;<BR>
&gt; telephone number problem rather than a domain problem, then it is =
a&nbsp;<BR>
&gt; much easier problem.&nbsp; If we can scope it that way then it will =
make&nbsp;<BR>
&gt; it a much easier problem to solve.<BR>
&gt;<BR>
&gt; Alan Johnston:&nbsp; All the SIP Forum discussion is in fact =
about&nbsp;<BR>
&gt; telephone numbers, so restricting it to telephone numbers would =
be&nbsp;<BR>
&gt; acceptable.<BR>
&gt;<BR>
&gt;<BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; dispatch mailing list<BR>
&gt; dispatch@ietf.org<BR>
&gt; <A =
HREF=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>
<BR>
_______________________________________________<BR>
dispatch mailing list<BR>
dispatch@ietf.org<BR>
<A =
HREF=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>
<BR>
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From jiri@iptel.org  Mon Nov 16 11:31:06 2009
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Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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I think it is worth mentioning and doublechecking a problem we touched
in this conversation, which is SIP servers modifying SDP to facilitate
NAT traversal. I think Jon P. has said this was unacceptable use-case not
good enough as motivation for work related to it, because it exposes
such calls to MiTM attacks. Is that right, Jon?

-jiri

Mary Barnes wrote:
> 
> Hi all,
> 
> I have uploaded Jim McEachern's notes from the Identity Adhoc yesterday:
> http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/Identity-AdHoc-Notes-jmce.doc
> 
> Regards,
> Mary.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From jon.peterson@neustar.biz  Mon Nov 16 12:45:35 2009
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Date: Mon, 16 Nov 2009 12:45:28 -0800
From: Jon Peterson <jon.peterson@neustar.biz>
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I was not arguing against NAT traversal, actually. From the start of the 
SIP identity work, we considered support for NAT traversal as a 
requirement. Of course, the authors of RFC4474 assumed that something 
roughly like ICE would take over this role, and obviously the deployment 
situation is bit more complicated (we similarly hoped that GRUU would 
make it unnecessary for intermediaries to modify Contact headers, and so 
on). Regardless, I still believe we need to have a reasonable story for 
how an identity mechanism will accommodate NAT traversal.

To rehash arguments that have circulated in this discussion many times, 
one must distinguish the NAT traversal case from the "media steering" 
requirement that has motivated much of the re-examination of identity. 
RFC4474 signatures are applied not by the user agent (at least not in 
the typical case), but rather by a server operated by the user agent's 
administrative domain. The administrative domain can more or less modify 
signaling arbitrarily before adding an Identity header per RFC4474. 
Thus, some sort of intermediary that modifies SIP signaling for the sake 
of NAT traversal can work with RFC4474, provided that the intermediary 
acts before the Identity header is added.

While it can therefore be argued that RFC4474 does not rule out 
intermediary-based NAT traversal entirely, it does protect against any 
intermediary altering SDP once a request has left the originating 
administrative domain with an Identity header. The "media steering" use 
cases that challenge this design are predicated on an intermediary with 
no necessary relationship to the originating or terminating 
administrative domain unilaterally modifying SDP in transit, usually in 
order to force layer 3 routing of media to pass through one or more 
anchor points in a network - perhaps for the sake of drawing the traffic 
across a managed circuit with carefully-managed quality, or to 
facilitate lawful intercept, or what have you. It is that requirement 
which I have consistently argued is equivalent to exposing this 
mechanism to a man-in-the-middle attack. If we removed enough 
protections in the design of RFC4474 to enable media steering by "good" 
intermediaries, we are effectively also enabling any malicious entities 
to change those aspects of SIP requests as well.

I do not find the requirement for NAT traversal unacceptable, but I do 
consider the requirement for media steering highly dubious at best.

Jon Peterson
NeuStar, Inc

Jiri Kuthan wrote:
> I think it is worth mentioning and doublechecking a problem we touched
> in this conversation, which is SIP servers modifying SDP to facilitate
> NAT traversal. I think Jon P. has said this was unacceptable use-case not
> good enough as motivation for work related to it, because it exposes
> such calls to MiTM attacks. Is that right, Jon?
>
> -jiri
>
> Mary Barnes wrote:
>>
>> Hi all,
>>
>> I have uploaded Jim McEachern's notes from the Identity Adhoc yesterday:
>> http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/Identity-AdHoc-Notes-jmce.doc 
>>
>>
>> Regards,
>> Mary.
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch


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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Jon Peterson <jon.peterson@neustar.biz>, Jiri Kuthan <jiri@iptel.org>
Date: Mon, 16 Nov 2009 13:20:37 -0800
Thread-Topic: [dispatch] Identity Adhoc - Nov 9th: Notes available
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>From the start of the SIP identity work, we considered support for NAT tra=
versal as a
requirement.

In hindsight, this assumption does not include the case when NAT traversal =
for all
application layer protocols is taken care of by HIP, in which case SIP deve=
lopers do
not have to worry about it:
"Basic HIP Extensions for Traversal of Network Address Translators"
http://tools.ietf.org/html/draft-ietf-hip-nat-traversal-09

HIP developers do the NAT traversal work and the problem discussed below do=
 not arise.
HIP uses ICE as well.

Is there a way to convince the SIP standards community of the benefits of H=
IP?
Notable exception:
ftp://ftp.rfc-editor.org/in-notes/internet-drafts/draft-ietf-hip-bone-03.tx=
t

Thanks, Henry


On 11/16/09 2:45 PM, "Jon Peterson" <jon.peterson@neustar.biz> wrote:



I was not arguing against NAT traversal, actually. From the start of the
SIP identity work, we considered support for NAT traversal as a
requirement. Of course, the authors of RFC4474 assumed that something
roughly like ICE would take over this role, and obviously the deployment
situation is bit more complicated (we similarly hoped that GRUU would
make it unnecessary for intermediaries to modify Contact headers, and so
on). Regardless, I still believe we need to have a reasonable story for
how an identity mechanism will accommodate NAT traversal.

To rehash arguments that have circulated in this discussion many times,
one must distinguish the NAT traversal case from the "media steering"
requirement that has motivated much of the re-examination of identity.
RFC4474 signatures are applied not by the user agent (at least not in
the typical case), but rather by a server operated by the user agent's
administrative domain. The administrative domain can more or less modify
signaling arbitrarily before adding an Identity header per RFC4474.
Thus, some sort of intermediary that modifies SIP signaling for the sake
of NAT traversal can work with RFC4474, provided that the intermediary
acts before the Identity header is added.

While it can therefore be argued that RFC4474 does not rule out
intermediary-based NAT traversal entirely, it does protect against any
intermediary altering SDP once a request has left the originating
administrative domain with an Identity header. The "media steering" use
cases that challenge this design are predicated on an intermediary with
no necessary relationship to the originating or terminating
administrative domain unilaterally modifying SDP in transit, usually in
order to force layer 3 routing of media to pass through one or more
anchor points in a network - perhaps for the sake of drawing the traffic
across a managed circuit with carefully-managed quality, or to
facilitate lawful intercept, or what have you. It is that requirement
which I have consistently argued is equivalent to exposing this
mechanism to a man-in-the-middle attack. If we removed enough
protections in the design of RFC4474 to enable media steering by "good"
intermediaries, we are effectively also enabling any malicious entities
to change those aspects of SIP requests as well.

I do not find the requirement for NAT traversal unacceptable, but I do
consider the requirement for media steering highly dubious at best.

Jon Peterson
NeuStar, Inc

Jiri Kuthan wrote:
> I think it is worth mentioning and doublechecking a problem we touched
> in this conversation, which is SIP servers modifying SDP to facilitate
> NAT traversal. I think Jon P. has said this was unacceptable use-case not
> good enough as motivation for work related to it, because it exposes
> such calls to MiTM attacks. Is that right, Jon?
>
> -jiri
>
> Mary Barnes wrote:
>>
>> Hi all,
>>
>> I have uploaded Jim McEachern's notes from the Identity Adhoc yesterday:
>> http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/Identity-AdHoc-=
Notes-jmce.doc
>>
>>
>> Regards,
>> Mary.
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> 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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<HTML>
<HEAD>
<TITLE>Re: [dispatch] Identity Adhoc - Nov 9th: Notes available</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
13pt'>&gt;From the start of the SIP identity work, we considered support fo=
r NAT traversal as a<BR>
requirement. <BR>
<BR>
In hindsight, this assumption does not include the case when NAT traversal =
for all <BR>
application layer protocols is taken care of by HIP, in which case SIP deve=
lopers do <BR>
not have to worry about it: <BR>
&#8220;</SPAN><FONT SIZE=3D"2"><SPAN STYLE=3D'font-size:12pt'>Basic HIP Ext=
ensions for Traversal of Network Address Translators&#8221;<BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:13pt'><a href=3D"http://tools.ietf.o=
rg/html/draft-ietf-hip-nat-traversal-09">http://tools.ietf.org/html/draft-i=
etf-hip-nat-traversal-09</a><BR>
<BR>
HIP developers do the NAT traversal work and the problem discussed below do=
 not arise. <BR>
HIP uses ICE as well.<BR>
<BR>
Is there a way to convince the SIP standards community of the benefits of H=
IP?<BR>
Notable exception:<BR>
<a href=3D"ftp://ftp.rfc-editor.org/in-notes/internet-drafts/draft-ietf-hip=
-bone-03.txt">ftp://ftp.rfc-editor.org/in-notes/internet-drafts/draft-ietf-=
hip-bone-03.txt</a><BR>
<BR>
Thanks, Henry<BR>
<BR>
<BR>
On 11/16/09 2:45 PM, &quot;Jon Peterson&quot; &lt;<a href=3D"jon.peterson@n=
eustar.biz">jon.peterson@neustar.biz</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:13pt'><BR>
<BR>
I was not arguing against NAT traversal, actually. From the start of the<BR=
>
SIP identity work, we considered support for NAT traversal as a<BR>
requirement. Of course, the authors of RFC4474 assumed that something<BR>
roughly like ICE would take over this role, and obviously the deployment<BR=
>
situation is bit more complicated (we similarly hoped that GRUU would<BR>
make it unnecessary for intermediaries to modify Contact headers, and so<BR=
>
on). Regardless, I still believe we need to have a reasonable story for<BR>
how an identity mechanism will accommodate NAT traversal.<BR>
<BR>
To rehash arguments that have circulated in this discussion many times,<BR>
one must distinguish the NAT traversal case from the &quot;media steering&q=
uot;<BR>
requirement that has motivated much of the re-examination of identity.<BR>
RFC4474 signatures are applied not by the user agent (at least not in<BR>
the typical case), but rather by a server operated by the user agent's<BR>
administrative domain. The administrative domain can more or less modify<BR=
>
signaling arbitrarily before adding an Identity header per RFC4474.<BR>
Thus, some sort of intermediary that modifies SIP signaling for the sake<BR=
>
of NAT traversal can work with RFC4474, provided that the intermediary<BR>
acts before the Identity header is added.<BR>
<BR>
While it can therefore be argued that RFC4474 does not rule out<BR>
intermediary-based NAT traversal entirely, it does protect against any<BR>
intermediary altering SDP once a request has left the originating<BR>
administrative domain with an Identity header. The &quot;media steering&quo=
t; use<BR>
cases that challenge this design are predicated on an intermediary with<BR>
no necessary relationship to the originating or terminating<BR>
administrative domain unilaterally modifying SDP in transit, usually in<BR>
order to force layer 3 routing of media to pass through one or more<BR>
anchor points in a network - perhaps for the sake of drawing the traffic<BR=
>
across a managed circuit with carefully-managed quality, or to<BR>
facilitate lawful intercept, or what have you. It is that requirement<BR>
which I have consistently argued is equivalent to exposing this<BR>
mechanism to a man-in-the-middle attack. If we removed enough<BR>
protections in the design of RFC4474 to enable media steering by &quot;good=
&quot;<BR>
intermediaries, we are effectively also enabling any malicious entities<BR>
to change those aspects of SIP requests as well.<BR>
<BR>
I do not find the requirement for NAT traversal unacceptable, but I do<BR>
consider the requirement for media steering highly dubious at best.<BR>
<BR>
Jon Peterson<BR>
NeuStar, Inc<BR>
<BR>
Jiri Kuthan wrote:<BR>
&gt; I think it is worth mentioning and doublechecking a problem we touched=
<BR>
&gt; in this conversation, which is SIP servers modifying SDP to facilitate=
<BR>
&gt; NAT traversal. I think Jon P. has said this was unacceptable use-case =
not<BR>
&gt; good enough as motivation for work related to it, because it exposes<B=
R>
&gt; such calls to MiTM attacks. Is that right, Jon?<BR>
&gt;<BR>
&gt; -jiri<BR>
&gt;<BR>
&gt; Mary Barnes wrote:<BR>
&gt;&gt;<BR>
&gt;&gt; Hi all,<BR>
&gt;&gt;<BR>
&gt;&gt; I have uploaded Jim McEachern's notes from the Identity Adhoc yest=
erday:<BR>
&gt;&gt; <a href=3D"http://www.softarmor.com/dispatch/IETF76/Meeting%20note=
s/Identity-AdHoc-Notes-jmce.doc">http://www.softarmor.com/dispatch/IETF76/M=
eeting%20notes/Identity-AdHoc-Notes-jmce.doc</a><BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;&gt; Regards,<BR>
&gt;&gt; Mary.<BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;&gt; ------------------------------------------------------------------=
------<BR>
&gt;&gt;<BR>
&gt;&gt; _______________________________________________<BR>
&gt;&gt; dispatch mailing list<BR>
&gt;&gt; <a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https:/=
/www.ietf.org/mailman/listinfo/dispatch</a><BR>
<BR>
_______________________________________________<BR>
dispatch mailing 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>
</BODY>
</HTML>


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To: Jon Peterson <jon.peterson@neustar.biz>
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Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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So, in summary: SDP (and other) modification should only be done on a =
UA's behalf  by its administrative domain, otherwise, it's unsafe.

Seems reasonable to me.

We seem to recycle the same arguments over and over again....

On Nov 16, 2009, at 22:45 , Jon Peterson wrote:

>=20
> I was not arguing against NAT traversal, actually. =46rom the start of =
the SIP identity work, we considered support for NAT traversal as a =
requirement. Of course, the authors of RFC4474 assumed that something =
roughly like ICE would take over this role, and obviously the deployment =
situation is bit more complicated (we similarly hoped that GRUU would =
make it unnecessary for intermediaries to modify Contact headers, and so =
on). Regardless, I still believe we need to have a reasonable story for =
how an identity mechanism will accommodate NAT traversal.
>=20
> To rehash arguments that have circulated in this discussion many =
times, one must distinguish the NAT traversal case from the "media =
steering" requirement that has motivated much of the re-examination of =
identity. RFC4474 signatures are applied not by the user agent (at least =
not in the typical case), but rather by a server operated by the user =
agent's administrative domain. The administrative domain can more or =
less modify signaling arbitrarily before adding an Identity header per =
RFC4474. Thus, some sort of intermediary that modifies SIP signaling for =
the sake of NAT traversal can work with RFC4474, provided that the =
intermediary acts before the Identity header is added.
>=20
> While it can therefore be argued that RFC4474 does not rule out =
intermediary-based NAT traversal entirely, it does protect against any =
intermediary altering SDP once a request has left the originating =
administrative domain with an Identity header. The "media steering" use =
cases that challenge this design are predicated on an intermediary with =
no necessary relationship to the originating or terminating =
administrative domain unilaterally modifying SDP in transit, usually in =
order to force layer 3 routing of media to pass through one or more =
anchor points in a network - perhaps for the sake of drawing the traffic =
across a managed circuit with carefully-managed quality, or to =
facilitate lawful intercept, or what have you. It is that requirement =
which I have consistently argued is equivalent to exposing this =
mechanism to a man-in-the-middle attack. If we removed enough =
protections in the design of RFC4474 to enable media steering by "good" =
intermediaries, we are effectively also enabling any malicious entities =
to change those aspects of SIP requests as well.
>=20
> I do not find the requirement for NAT traversal unacceptable, but I do =
consider the requirement for media steering highly dubious at best.
>=20
> Jon Peterson
> NeuStar, Inc
>=20
> Jiri Kuthan wrote:
>> I think it is worth mentioning and doublechecking a problem we =
touched
>> in this conversation, which is SIP servers modifying SDP to =
facilitate
>> NAT traversal. I think Jon P. has said this was unacceptable use-case =
not
>> good enough as motivation for work related to it, because it exposes
>> such calls to MiTM attacks. Is that right, Jon?
>>=20
>> -jiri
>>=20
>> Mary Barnes wrote:
>>>=20
>>> Hi all,
>>>=20
>>> I have uploaded Jim McEachern's notes from the Identity Adhoc =
yesterday:
>>> =
http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/Identity-AdHoc-No=
tes-jmce.doc=20
>>>=20
>>> Regards,
>>> Mary.
>>>=20
>>>=20
>>> =
------------------------------------------------------------------------
>>>=20
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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Jon Peterson wrote:
> 
> I was not arguing against NAT traversal, actually. From the start of the 
> SIP identity work, we considered support for NAT traversal as a 
> requirement. Of course, the authors of RFC4474 assumed that something 
> roughly like ICE would take over this role, and obviously the deployment 
> situation is bit more complicated (we similarly hoped that GRUU would 
> make it unnecessary for intermediaries to modify Contact headers, and so 
> on). Regardless, I still believe we need to have a reasonable story for 
> how an identity mechanism will accommodate NAT traversal.
> 
> To rehash arguments that have circulated in this discussion many times, 
> one must distinguish the NAT traversal case from the "media steering" 
> requirement that has motivated much of the re-examination of identity. 
> RFC4474 signatures are applied not by the user agent (at least not in 
> the typical case), but rather by a server operated by the user agent's 
> administrative domain. The administrative domain can more or less modify 
> signaling arbitrarily before adding an Identity header per RFC4474. 
> Thus, some sort of intermediary that modifies SIP signaling for the sake 
> of NAT traversal can work with RFC4474, provided that the intermediary 
> acts before the Identity header is added.
> 
> While it can therefore be argued that RFC4474 does not rule out 
> intermediary-based NAT traversal entirely, it does protect against any 
> intermediary altering SDP once a request has left the originating 
> administrative domain with an Identity header. 
> The "media steering" use 
> cases that challenge this design are predicated on an intermediary with 
> no necessary relationship to the originating or terminating 
> administrative domain unilaterally modifying SDP in transit, usually in 
> order to force layer 3 routing of media to pass through one or more 
> anchor points in a network - perhaps for the sake of drawing the traffic 
> across a managed circuit with carefully-managed quality, or to 
> facilitate lawful intercept, or what have you. It is that requirement 
> which I have consistently argued is equivalent to exposing this 
> mechanism to a man-in-the-middle attack.

how is that different from a MiTM attacking SDP on the path from UA
to a 4474-enable proxy?

>   If we removed enough
> protections in the design of RFC4474 to enable media steering by "good" 
> intermediaries, we are effectively also enabling any malicious entities 
> to change those aspects of SIP requests as well.

I'm wondering if  RFC4474 can prevent MiTM as it is not  guaranteed to provide
end-to-end protection in the typical case.

> 
> I do not find the requirement for NAT traversal unacceptable, but I do 
> consider the requirement for media steering highly dubious at best.

Some while ago I would have thought that in this typical case there
are two administrative zones under respective control with a pipe in
between. Then it shall be manageable to put a SDP-manipulating closer
to UA than 4474 processing. In the meantime I think it became hard to
find a SIP path which in between two 4474 entities has nothing that
rewrites SDP.

-jiri


> 
> Jon Peterson
> NeuStar, Inc
> 
> Jiri Kuthan wrote:
>> I think it is worth mentioning and doublechecking a problem we touched
>> in this conversation, which is SIP servers modifying SDP to facilitate
>> NAT traversal. I think Jon P. has said this was unacceptable use-case not
>> good enough as motivation for work related to it, because it exposes
>> such calls to MiTM attacks. Is that right, Jon?
>>
>> -jiri
>>
>> Mary Barnes wrote:
>>>
>>> Hi all,
>>>
>>> I have uploaded Jim McEachern's notes from the Identity Adhoc yesterday:
>>> http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/Identity-AdHoc-Notes-jmce.doc 
>>>
>>>
>>> Regards,
>>> Mary.
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
> 

From jiri@iptel.org  Tue Nov 17 00:27:38 2009
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From: Jiri Kuthan <jiri@iptel.org>
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Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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Francois Audet wrote:
> So, in summary: SDP (and other) modification should only be done on a UA's behalf  by its administrative domain, otherwise, it's unsafe.


Hi Francois,

I'm not quite yet sure what administrative domain means?

The picture I'm having in my mind is using say iptel.org service (located
in Central Europe in a facility administered by iptel.org's administrator)
from an IETF meeting (administered by the IETF team) to my family
(located within what a German ISP administers).

-jiri

> 
> Seems reasonable to me.
> 
> We seem to recycle the same arguments over and over again....
> 
> On Nov 16, 2009, at 22:45 , Jon Peterson wrote:
> 
>> I was not arguing against NAT traversal, actually. From the start of the SIP identity work, we considered support for NAT traversal as a requirement. Of course, the authors of RFC4474 assumed that something roughly like ICE would take over this role, and obviously the deployment situation is bit more complicated (we similarly hoped that GRUU would make it unnecessary for intermediaries to modify Contact headers, and so on). Regardless, I still believe we need to have a reasonable story for how an identity mechanism will accommodate NAT traversal.
>>
>> To rehash arguments that have circulated in this discussion many times, one must distinguish the NAT traversal case from the "media steering" requirement that has motivated much of the re-examination of identity. RFC4474 signatures are applied not by the user agent (at least not in the typical case), but rather by a server operated by the user agent's administrative domain. The administrative domain can more or less modify signaling arbitrarily before adding an Identity header per RFC4474. Thus, some sort of intermediary that modifies SIP signaling for the sake of NAT traversal can work with RFC4474, provided that the intermediary acts before the Identity header is added.
>>
>> While it can therefore be argued that RFC4474 does not rule out intermediary-based NAT traversal entirely, it does protect against any intermediary altering SDP once a request has left the originating administrative domain with an Identity header. The "media steering" use cases that challenge this design are predicated on an intermediary with no necessary relationship to the originating or terminating administrative domain unilaterally modifying SDP in transit, usually in order to force layer 3 routing of media to pass through one or more anchor points in a network - perhaps for the sake of drawing the traffic across a managed circuit with carefully-managed quality, or to facilitate lawful intercept, or what have you. It is that requirement which I have consistently argued is equivalent to exposing this mechanism to a man-in-the-middle attack. If we removed enough protections in the design of RFC4474 to enable media steering by "good" intermediaries, we are effectively als
o enabling any malicious entities to change those aspects of SIP requests as well.
>>
>> I do not find the requirement for NAT traversal unacceptable, but I do consider the requirement for media steering highly dubious at best.
>>
>> Jon Peterson
>> NeuStar, Inc
>>
>> Jiri Kuthan wrote:
>>> I think it is worth mentioning and doublechecking a problem we touched
>>> in this conversation, which is SIP servers modifying SDP to facilitate
>>> NAT traversal. I think Jon P. has said this was unacceptable use-case not
>>> good enough as motivation for work related to it, because it exposes
>>> such calls to MiTM attacks. Is that right, Jon?
>>>
>>> -jiri
>>>
>>> Mary Barnes wrote:
>>>> Hi all,
>>>>
>>>> I have uploaded Jim McEachern's notes from the Identity Adhoc yesterday:
>>>> http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/Identity-AdHoc-Notes-jmce.doc 
>>>>
>>>> Regards,
>>>> Mary.
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> _______________________________________________
>>>> 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 jon.peterson@neustar.biz  Tue Nov 17 04:20:36 2009
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>> The "media steering" use cases that challenge this design are 
>> predicated on an intermediary with no necessary relationship to the 
>> originating or terminating administrative domain unilaterally 
>> modifying SDP in transit, usually in order to force layer 3 routing 
>> of media to pass through one or more anchor points in a network - 
>> perhaps for the sake of drawing the traffic across a managed circuit 
>> with carefully-managed quality, or to facilitate lawful intercept, or 
>> what have you. It is that requirement which I have consistently 
>> argued is equivalent to exposing this mechanism to a 
>> man-in-the-middle attack.
> how is that different from a MiTM attacking SDP on the path from UA
> to a 4474-enable proxy?
I'm not sure how to answer your question except to say that it is a 
fundamental difference in the protection scope of the mechanism. The 
path between the user agent and the RFC4474-enabled proxy must be 
secured in some fashion, it's true, and there are a number of ways that 
security might be provided that can either allow for the presence of 
other intermediaries in the administrative domain or not, depending on 
the SIP deployment of the domain (RFC4474 does talk about this a bit). 
If those protections aren't in place, then of course any adversary can 
alter the signaling arbitrarily with fear of detection. Once an Identity 
header is in place, however, any subsequent modifications to the 
signaling that could lead to impersonation (in so far as the threat of 
impersonation is defined in the draft) are supposed to show up as 
violations. That's the protection that the Identity mechanism affords.

So the difference is that in one case, there's supposed to be security, 
and in the other case there isn't.
>>   If we removed enough
>> protections in the design of RFC4474 to enable media steering by 
>> "good" intermediaries, we are effectively also enabling any malicious 
>> entities to change those aspects of SIP requests as well.
> I'm wondering if  RFC4474 can prevent MiTM as it is not  guaranteed to 
> provide
> end-to-end protection in the typical case.
No mechanism can "prevent" a man-in-the-middle, it can merely detect it. 
RFC4474 provides its protections within its defined scope, which is  
from an authentication service in the administrative domain to the 
terminating user agent. If the path between the originating user agent 
and the authentication service is insecure and vulnerable, as discussed 
above, then yes, RFC4474 doesn't provide end-to-end protection.
>>
>> I do not find the requirement for NAT traversal unacceptable, but I 
>> do consider the requirement for media steering highly dubious at best.
> Some while ago I would have thought that in this typical case there
> are two administrative zones under respective control with a pipe in
> between. Then it shall be manageable to put a SDP-manipulating closer
> to UA than 4474 processing. In the meantime I think it became hard to
> find a SIP path which in between two 4474 entities has nothing that
> rewrites SDP.
You are not the first one to make this observation, but I don't think 
this information is very useful in drafting the sorts of requirements in 
Mr. Elwell's document. The stated requirement for "media steering" 
assumes something much more specific than simply wanting to rewrite SDP. 
An intermediary that wants to rewrite SDP may have one of several 
relationships with the originating and terminating user agents or 
administrative domains. The "media steering" use cases invariably assume 
an entity acting unilaterally, without necessarily the knowledge or 
consent of either side of the transaction - it acts by virtue of being 
in the path of signaling, even if only as an ALG. If we design a 
security mechanism that allows an entity to act with that sort of 
impunity, it may facilitate some few beneficial services but it will 
also facilitate a variety of attacks. I believe we can do better - we 
can find ways to allow intermediaries to perform the services they'd 
like to perform without acting unilaterally. Certainly at the time 
RFC4474 was under development, we believed that mechanisms like 
session-policy would eventually provide a way for intermediaries to 
force user agents to conform with intermediary policy, including session 
dependent policy cases for SDP. If we're going to dedicate design energy 
to solving problems in this neighborhood, one could argue that solving 
for these building blocks first (allowing intermediaries to modify 
signaling in a collaborative manner rather than unilaterally) will make 
the path to an identity solution much, much easier. From a requirements 
perspective, this is much closer to what I'd like to see expressed than 
something like "media steering," which basically equates to reducing the 
protection space.

Jon Peterson
NeuStar, Inc.

> -jiri
>
>


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Subject: [dispatch] FW: I-D Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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FYI to the general SIP community. 

There have been substantial field reports of difficulties with T.30 FAX is
SIP networks and the SIP Forum was asked to help facilitate discussions
among a group of technical experts in fax order to provider recommendations
to either the IETF and or ITU SG16 on possible approaches to the problem.

Fax is a essential realtime communications service that is not going away
and making fax work properly needs to be addressed.

-----Original Message-----
From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org]
On Behalf Of Internet-Drafts@ietf.org
Sent: Tuesday, November 17, 2009 3:15 AM
To: i-d-announce@ietf.org
Subject: I-D Action:draft-jones-sip-forum-fax-problem-statement-00.txt 

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

	Title           : SIP Forum - Fax Over IP Task Group Problem
Statement
	Author(s)       : X. Chen, et al.
	Filename        : draft-jones-sip-forum-fax-problem-statement-00.txt
	Pages           : 13
	Date            : 2009-11-17

This memo is published for informational purposes to document the issues
identified by the SIP Forum with respect to the transmission of facsimile
signaling messages and fax page data over Internet Protocol (IP) networks.
Further, it is the intent of this memo to alert the IETF to the formation of
a Fax Over IP Task Group within the SIP Forum chartered to investigate and
address identified issues as they relate to the deployment of fax services
in Session Initiation Protocol (SIP) networks.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-jones-sip-forum-fax-problem-statem
ent-00.txt

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

Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.


From jiri@iptel.org  Tue Nov 17 09:57:41 2009
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Jon Peterson wrote:
> 
>>> The "media steering" use cases that challenge this design are 
>>> predicated on an intermediary with no necessary relationship to the 
>>> originating or terminating administrative domain unilaterally 
>>> modifying SDP in transit, usually in order to force layer 3 routing 
>>> of media to pass through one or more anchor points in a network - 
>>> perhaps for the sake of drawing the traffic across a managed circuit 
>>> with carefully-managed quality, or to facilitate lawful intercept, or 
>>> what have you. It is that requirement which I have consistently 
>>> argued is equivalent to exposing this mechanism to a 
>>> man-in-the-middle attack.
>> how is that different from a MiTM attacking SDP on the path from UA
>> to a 4474-enable proxy?


What is all if it then good for if we don't have guarantee for e2e
integrity protectino and have quite likely (Though not lovely) chance
that ALGs will break even identity?

To me having a partial protection of SDP along the SIP path sounds
remarkably similar to the concept of partial pregnancy.


> I'm not sure how to answer your question except to say that it is a 
> fundamental difference in the protection scope of the mechanism. The 
> path between the user agent and the RFC4474-enabled proxy must be 
> secured in some fashion, it's true, and there are a number of ways that 
> security might be provided that can either allow for the presence of 
> other intermediaries in the administrative domain or not, depending on 
> the SIP deployment of the domain (RFC4474 does talk about this a bit). 
> If those protections aren't in place, then of course any adversary can 
> alter the signaling arbitrarily with fear of detection. 

Even if there are other technologies (let's assume for a while that TLS
can be relied upon along the whole path) that can cover the whole path,
what's integrity in 4474 then good for?

> Once an Identity
> header is in place, however, any subsequent modifications to the 
> signaling that could lead to impersonation (in so far as the threat of 
> impersonation is defined in the draft) are supposed to show up as 
> violations. That's the protection that the Identity mechanism affords.

I understand the benefit. The point is that Identity appears meaningful
even without that (to the point I know who is calling) and the cost is
that it breaks whenever there is an ALG in the path, which happens quite
frequently.


> 
> So the difference is that in one case, there's supposed to be security, 

I'm worried that covering SDP along a piece of the path is not eliminating
MiTM attacks and therefore doesn't provide better security.

> and in the other case there isn't.
>>>   If we removed enough
>>> protections in the design of RFC4474 to enable media steering by 
>>> "good" intermediaries, we are effectively also enabling any malicious 
>>> entities to change those aspects of SIP requests as well.
>> I'm wondering if  RFC4474 can prevent MiTM as it is not  guaranteed to 
>> provide
>> end-to-end protection in the typical case.
> No mechanism can "prevent" a man-in-the-middle, it can merely detect it. 
> RFC4474 provides its protections within its defined scope, which is  
> from an authentication service in the administrative domain to the 
> terminating user agent. If the path between the originating user agent 
> and the authentication service is insecure and vulnerable, as discussed 
> above, then yes, RFC4474 doesn't provide end-to-end protection.


That's my problem -- we only have partial integrity protection  and for
this limited benefit  we don't get over ALGs.

>>>
>>> I do not find the requirement for NAT traversal unacceptable, but I 
>>> do consider the requirement for media steering highly dubious at best.
>> Some while ago I would have thought that in this typical case there
>> are two administrative zones under respective control with a pipe in
>> between. Then it shall be manageable to put a SDP-manipulating closer
>> to UA than 4474 processing. In the meantime I think it became hard to
>> find a SIP path which in between two 4474 entities has nothing that
>> rewrites SDP.
> You are not the first one to make this observation, but I don't think 
> this information is very useful in drafting the sorts of requirements in 
> Mr. Elwell's document. The stated requirement for "media steering" 
> assumes something much more specific than simply wanting to rewrite SDP. 
> An intermediary that wants to rewrite SDP may have one of several 
> relationships with the originating and terminating user agents or 
> administrative domains. The "media steering" use cases invariably assume 
> an entity acting unilaterally, without necessarily the knowledge or 
> consent of either side of the transaction - it acts by virtue of being 
> in the path of signaling, even if only as an ALG. If we design a 
> security mechanism that allows an entity to act with that sort of 
> impunity, it may facilitate some few beneficial services but it will 
> also facilitate a variety of attacks.

I'm afraid the attacks are there anyhow. Use of STUN, for example, exposes
the whole architecture to MiTM attack even if there is SDP protection
end-to-end in combination with hop-by-hop TLS encryption. TLS implementations
are for some reason quite rare. In other words if there is somone in position
to rewrite packets as sent from a UA, SDP protection between two proxy servers
will not stop him.


>  I believe we can do better - we
> can find ways to allow intermediaries to perform the services they'd 
> like to perform without acting unilaterally. Certainly at the time 
> RFC4474 was under development, we believed that mechanisms like 
> session-policy would eventually provide a way for intermediaries to 
> force user agents to conform with intermediary policy, including session 
> dependent policy cases for SDP. If we're going to dedicate design energy 
> to solving problems in this neighborhood, one could argue that solving 
> for these building blocks first (allowing intermediaries to modify 
> signaling in a collaborative manner rather than unilaterally) will make 
> the path to an identity solution much, much easier. From a requirements 
> perspective, this is much closer to what I'd like to see expressed than 
> something like "media steering," which basically equates to reducing the 
> protection space.

The problem I see is it reduces protection space but it does not reduce
the problem space. With TLS we could be getting protection along the whole
path on a hop-by-hop basis so the residual security risk is that
a legitimate SIP element on the path between 4474 issuer and verifier
changes SDP.

Why do you think that any negotiation consensus-producing facility would
fly better than adoption of the policy framework?

Or do you rather think along the lines of setting the user expectations?
(a la a new error code "18x hang up now and upgrade to nat-free IPv6
or media encryption, otherwise you will be in the middle of a call that
may be intercepted by parties with legitimitate as well as illegitimate
interest to do so" with obligation that all evil parties send this sort
of message...?)

OR advisory SDP protection which makes recepients alert of a change but
doesn't yet mandate the protocol stack to throw broken traffic away?

or a BCP that creates a VPN tunnel, producing security and consensus to
use some public address for public trAFFIC?

I really don't know what your suggestion above means.

The point is really as with NAT's IP address rewriting, there is SDP
rewriting and assuming in protocol design that these things don't
exist will probably do bigger harm (such as unusability of 4474)
then accepting it as matter of fact.

-jiri

> 
> Jon Peterson
> NeuStar, Inc.
> 
>> -jiri
>>
>>
> 

From dwing@cisco.com  Tue Nov 17 12:28:58 2009
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From: "Dan Wing" <dwing@cisco.com>
To: "'Jon Peterson'" <jon.peterson@neustar.biz>, "'Jiri Kuthan'" <jiri@iptel.org>
References: <F592E36A5C943E4E91F25880D05AD1140CC8E1F3@zrc2hxm0.corp.nortel.com><4B01A877.4010306@iptel.org> <4B01B9E8.7050507@neustar.biz>
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Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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> -----Original Message-----
> From: dispatch-bounces@ietf.org 
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Jon Peterson
> Sent: Monday, November 16, 2009 12:45 PM
> To: Jiri Kuthan
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
> 
> 
> I was not arguing against NAT traversal, actually. From the 
> start of the 
> SIP identity work, we considered support for NAT traversal as a 
> requirement. Of course, the authors of RFC4474 assumed that something 
> roughly like ICE would take over this role, and obviously the 
> deployment 
> situation is bit more complicated (we similarly hoped that GRUU would 
> make it unnecessary for intermediaries to modify Contact 
> headers, and so 
> on). Regardless, I still believe we need to have a reasonable 
> story for 
> how an identity mechanism will accommodate NAT traversal.
> 
> To rehash arguments that have circulated in this discussion 
> many times, 
> one must distinguish the NAT traversal case from the "media steering" 
> requirement that has motivated much of the re-examination of 
> identity. 
> RFC4474 signatures are applied not by the user agent (at least not in 
> the typical case), but rather by a server operated by the 
> user agent's 
> administrative domain. The administrative domain can more or 
> less modify 
> signaling arbitrarily before adding an Identity header per RFC4474. 
> Thus, some sort of intermediary that modifies SIP signaling 
> for the sake 
> of NAT traversal can work with RFC4474, provided that the 
> intermediary 
> acts before the Identity header is added.
> 
> While it can therefore be argued that RFC4474 does not rule out 
> intermediary-based NAT traversal entirely, it does protect 
> against any 
> intermediary altering SDP once a request has left the originating 
> administrative domain with an Identity header. The "media 
> steering" use 
> cases that challenge this design are predicated on an 
> intermediary with 
> no necessary relationship to the originating or terminating 
> administrative domain unilaterally modifying SDP in transit, 
> usually in 
> order to force layer 3 routing of media to pass through one or more 
> anchor points in a network - perhaps for the sake of drawing 
> the traffic 
> across a managed circuit with carefully-managed quality, or to 
> facilitate lawful intercept, or what have you. It is that requirement 
> which I have consistently argued is equivalent to exposing this 
> mechanism to a man-in-the-middle attack. If we removed enough 
> protections in the design of RFC4474 to enable media steering 
> by "good" 
> intermediaries, we are effectively also enabling any 
> malicious entities 
> to change those aspects of SIP requests as well.
> 
> I do not find the requirement for NAT traversal unacceptable, 
> but I do 
> consider the requirement for media steering highly dubious at best.

"Highly dubious" = nobody really does it?

-d


> Jon Peterson
> NeuStar, Inc
> 
> Jiri Kuthan wrote:
> > I think it is worth mentioning and doublechecking a problem 
> we touched
> > in this conversation, which is SIP servers modifying SDP to 
> facilitate
> > NAT traversal. I think Jon P. has said this was 
> unacceptable use-case not
> > good enough as motivation for work related to it, because it exposes
> > such calls to MiTM attacks. Is that right, Jon?
> >
> > -jiri
> >
> > Mary Barnes wrote:
> >>
> >> Hi all,
> >>
> >> I have uploaded Jim McEachern's notes from the Identity 
> Adhoc yesterday:
> >> 
> http://www.softarmor.com/dispatch/IETF76/Meeting%20notes/Ident
> ity-AdHoc-Notes-jmce.doc 
> >>
> >>
> >> Regards,
> >> Mary.
> >>
> >>
> >> 
> --------------------------------------------------------------
> ----------
> >>
> >> _______________________________________________
> >> 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 victor.pascual.avila@gmail.com  Tue Nov 17 13:09:00 2009
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On Tue, Nov 17, 2009 at 9:28 PM, Dan Wing <dwing@cisco.com> wrote:
>> I do not find the requirement for NAT traversal unacceptable,
>> but I do
>> consider the requirement for media steering highly dubious at best.
>
> "Highly dubious" =3D nobody really does it?

Network administrators have actually found it easier to rewrite SDPs
than to use other existing mechanisms.

Cheers,
--=20
Victor Pascual =C3=81vila

From dean.willis@softarmor.com  Tue Nov 17 13:09:01 2009
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Dan Wing wrote:
>Jon said:
>>I do not find the requirement for NAT traversal unacceptable, 
>>but I do 
>>consider the requirement for media steering highly dubious at best.
> 
> 
> "Highly dubious" = nobody really does it?

People do highly dubious stuff all the time; they have unprotected sex, 
they jump out of perfectly good airplanes, they vote for Obama, they try 
for a single-handed Atlantic crossing in a 6 meter sailboat with no 
preparation, they try to implement RAI specifications from the RFC 
without parallel testing against existing systems, and so on.

The question is: Should we take the human urge to engage in highly 
dubious practies like traffic steering into account when we write an 
identity spec? If so, at what level of  dubiousness (or dubyaness, for 
the left-leaning) do we draw the line?

--
dean

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Dean Willis wrote:
> Dan Wing wrote:
>> Jon said:
>>> I do not find the requirement for NAT traversal unacceptable, but I 
>>> do consider the requirement for media steering highly dubious at best.
>>
>>
>> "Highly dubious" = nobody really does it?
> 
> People do highly dubious stuff all the time; they have unprotected sex, 
> they jump out of perfectly good airplanes, they vote for Obama, they try 
> for a single-handed Atlantic crossing in a 6 meter sailboat with no 
> preparation, they try to implement RAI specifications from the RFC 
> without parallel testing against existing systems, and so on.
> 
> The question is: Should we take the human urge to engage in highly 
> dubious practies like traffic steering into account when we write an 
> identity spec? If so, at what level of  dubiousness (or dubyaness, for 
> the left-leaning) do we draw the line?

I'd been thinking the dubyaness was just a bad dream.
Thank you for disabusing me of that.

	Paul

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From: "Dan Wing" <dwing@cisco.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>
References: <F592E36A5C943E4E91F25880D05AD1140CC8E1F3@zrc2hxm0.corp.nortel.com><4B01A877.4010306@iptel.org>	<4B01B9E8.7050507@neustar.biz> <048d01ca67c4$96503410$c2f0200a@cisco.com> <4B0310E1.9070200@softarmor.com>
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Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com] 
> Sent: Tuesday, November 17, 2009 1:09 PM
> To: Dan Wing
> Cc: 'Jon Peterson'; 'Jiri Kuthan'; dispatch@ietf.org
> Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
> 
> Dan Wing wrote:
> >Jon said:
> >>I do not find the requirement for NAT traversal unacceptable, 
> >>but I do 
> >>consider the requirement for media steering highly dubious at best.
> > 
> > 
> > "Highly dubious" = nobody really does it?
> 
> People do highly dubious stuff all the time; they have 
> unprotected sex, 
> they jump out of perfectly good airplanes, they vote for 
> Obama, they try 
> for a single-handed Atlantic crossing in a 6 meter sailboat with no 
> preparation, they try to implement RAI specifications from the RFC 
> without parallel testing against existing systems, and so on.
> 
> The question is: Should we take the human urge to engage in highly 
> dubious practies like traffic steering into account when we write an 
> identity spec? If so, at what level of  dubiousness (or 
> dubyaness, for the left-leaning) do we draw the line?

Said another way:  pragmatic versus academic.

-d


From HKaplan@acmepacket.com  Wed Nov 18 11:12:42 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Jon Peterson <jon.peterson@neustar.biz>, Jiri Kuthan <jiri@iptel.org>
Date: Wed, 18 Nov 2009 14:11:49 -0500
Thread-Topic: [dispatch] Identity Adhoc - Nov 9th: Notes available
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Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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Forget the NAT traversal piece, or even media steering, and just tell me wh=
at signing the IP Address/port in the SDP buys you.  Not the whole SDP, jus=
t the IP/ports.  Imagine if the media IP/ports were in a SIP header, and te=
ll me why we MUST sign that header to get "SIP Identity".

My claim is signing those IP/ports is neither necessary nor sufficient for =
the purposes of SIP Identity nor media identity.  It's an emperor with no c=
lothes, except we end up getting the bill for the clothing.

It is not sufficient because it does not prove the authenticity and identit=
y of the media.  And I don't just mean in an academic sense of "prove".  No=
t only can the media IP/ports still be spoofed and even intercepted - but n=
othing prevents me from just sending media from another address/port to get=
 you to listen to my advertisement, instructions to call my number to "veri=
fy your credit card", or whatever.

Only a media-plane mechanism will be sufficient for media identity, such as=
 DTLS-SRTP.

It is not necessary because, given a mechanism such as DTLS-SRTP (which *is=
* necessary for the property of media identity), signing the IP/ports is su=
perfluous.  It's making the mistake of confusing IP with an identity, vs. a=
 routing identifier. (ie, the argument HIP makes)

-hadriel

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Jon Peterson
> Sent: Monday, November 16, 2009 3:45 PM
>=20
> To rehash arguments that have circulated in this discussion many times,
> one must distinguish the NAT traversal case from the "media steering"
> requirement that has motivated much of the re-examination of identity.
> RFC4474 signatures are applied not by the user agent (at least not in
> the typical case), but rather by a server operated by the user agent's
> administrative domain. The administrative domain can more or less modify
> signaling arbitrarily before adding an Identity header per RFC4474.
> Thus, some sort of intermediary that modifies SIP signaling for the sake
> of NAT traversal can work with RFC4474, provided that the intermediary
> acts before the Identity header is added.
>=20
> While it can therefore be argued that RFC4474 does not rule out
> intermediary-based NAT traversal entirely, it does protect against any
> intermediary altering SDP once a request has left the originating
> administrative domain with an Identity header. The "media steering" use
> cases that challenge this design are predicated on an intermediary with
> no necessary relationship to the originating or terminating
> administrative domain unilaterally modifying SDP in transit, usually in
> order to force layer 3 routing of media to pass through one or more
> anchor points in a network - perhaps for the sake of drawing the traffic
> across a managed circuit with carefully-managed quality, or to
> facilitate lawful intercept, or what have you. It is that requirement
> which I have consistently argued is equivalent to exposing this
> mechanism to a man-in-the-middle attack. If we removed enough
> protections in the design of RFC4474 to enable media steering by "good"
> intermediaries, we are effectively also enabling any malicious entities
> to change those aspects of SIP requests as well.

From bernard_aboba@hotmail.com  Wed Nov 18 13:05:56 2009
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Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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Dean said:

 

"People do highly dubious stuff all the time; they have unprotected sex, 

they jump out of perfectly good airplanes, they vote for Obama, they try 

for a single-handed Atlantic crossing in a 6 meter sailboat with no 

preparation, they try to implement RAI specifications from the RFC 

without parallel testing against existing systems, and so on.

 

The question is: Should we take the human urge to engage in highly 

dubious practies like traffic steering into account when we write an 

identity spec? If so, at what level of  dubiousness (or dubyaness, for 

the left-leaning) do we draw the line?"

 

[BA]   A key question to answer is whether SIP Identity is for general
purpose use or not:

 

a)      If it is not for general purpose use, then the analysis (and
applicability statement)

can focus solely on those use cases for which it is appropriate, and the

"dubious" cases can be noted as out of scope.   

 

b)      If the intent is to provide a facility for general purpose use, then
enabling

use in commonly encountered situations is a requirement, regardless of how

"dubious" they might be.  

 

 


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<p class=3DMsoNormal>Dean said:<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>&quot;People do highly dubious stuff all the time; =
they have
unprotected sex, <o:p></o:p></p>

<p class=3DMsoNormal>they jump out of perfectly good airplanes, they =
vote for
Obama, they try <o:p></o:p></p>

<p class=3DMsoNormal>for a single-handed Atlantic crossing in a 6 meter =
sailboat
with no <o:p></o:p></p>

<p class=3DMsoNormal>preparation, they try to implement RAI =
specifications from
the RFC <o:p></o:p></p>

<p class=3DMsoNormal>without parallel testing against existing systems, =
and so
on.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The question is: Should we take the human urge to =
engage in
highly <o:p></o:p></p>

<p class=3DMsoNormal>dubious practies like traffic steering into account =
when we
write an <o:p></o:p></p>

<p class=3DMsoNormal>identity spec? If so, at what level of&nbsp; =
dubiousness (or
dubyaness, for <o:p></o:p></p>

<p class=3DMsoNormal>the left-leaning) do we draw the =
line?&quot;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>[BA] &nbsp;&nbsp;A key question to answer is =
whether SIP
Identity is for general purpose use or not:<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>a)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>If it is not for general purpose use, then the =
analysis
(and applicability statement)<o:p></o:p></p>

<p class=3DMsoNormal>can focus solely on those use cases for which it is
appropriate, and the<o:p></o:p></p>

<p class=3DMsoNormal>&quot;dubious&quot; cases can be noted as out of
scope.&nbsp; &nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>b)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>If the intent is to provide a facility for =
general
purpose use, then enabling<o:p></o:p></p>

<p class=3DMsoNormal>use in commonly encountered situations is a =
requirement,
regardless of how<o:p></o:p></p>

<p class=3DMsoNormal>&quot;dubious&quot; they might be.&nbsp; =
<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

</html>

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From fluffy@cisco.com  Wed Nov 18 21:31:17 2009
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From: Cullen Jennings <fluffy@cisco.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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If we all believe the only thing SBC rewrite is the IP and port of the  
media, I'm sure we can solve this in no time flat. Unfortunately, when  
a solution to solve that problem have been presented, it has become  
clear that SBC change much more than that.

Cullen <in my individual contributor role>

On Nov 18, 2009, at 12:11 PM, Hadriel Kaplan wrote:

>
> Forget the NAT traversal piece, or even media steering, and just  
> tell me what signing the IP Address/port in the SDP buys you.  Not  
> the whole SDP, just the IP/ports.  Imagine if the media IP/ports  
> were in a SIP header, and tell me why we MUST sign that header to  
> get "SIP Identity".
>
> My claim is signing those IP/ports is neither necessary nor  
> sufficient for the purposes of SIP Identity nor media identity.   
> It's an emperor with no clothes, except we end up getting the bill  
> for the clothing.
>
> It is not sufficient because it does not prove the authenticity and  
> identity of the media.  And I don't just mean in an academic sense  
> of "prove".  Not only can the media IP/ports still be spoofed and  
> even intercepted - but nothing prevents me from just sending media  
> from another address/port to get you to listen to my advertisement,  
> instructions to call my number to "verify your credit card", or  
> whatever.
>
> Only a media-plane mechanism will be sufficient for media identity,  
> such as DTLS-SRTP.
>
> It is not necessary because, given a mechanism such as DTLS-SRTP  
> (which *is* necessary for the property of media identity), signing  
> the IP/ports is superfluous.  It's making the mistake of confusing  
> IP with an identity, vs. a routing identifier. (ie, the argument HIP  
> makes)
>
> -hadriel
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Jon Peterson
>> Sent: Monday, November 16, 2009 3:45 PM
>>
>> To rehash arguments that have circulated in this discussion many  
>> times,
>> one must distinguish the NAT traversal case from the "media steering"
>> requirement that has motivated much of the re-examination of  
>> identity.
>> RFC4474 signatures are applied not by the user agent (at least not in
>> the typical case), but rather by a server operated by the user  
>> agent's
>> administrative domain. The administrative domain can more or less  
>> modify
>> signaling arbitrarily before adding an Identity header per RFC4474.
>> Thus, some sort of intermediary that modifies SIP signaling for the  
>> sake
>> of NAT traversal can work with RFC4474, provided that the  
>> intermediary
>> acts before the Identity header is added.
>>
>> While it can therefore be argued that RFC4474 does not rule out
>> intermediary-based NAT traversal entirely, it does protect against  
>> any
>> intermediary altering SDP once a request has left the originating
>> administrative domain with an Identity header. The "media steering"  
>> use
>> cases that challenge this design are predicated on an intermediary  
>> with
>> no necessary relationship to the originating or terminating
>> administrative domain unilaterally modifying SDP in transit,  
>> usually in
>> order to force layer 3 routing of media to pass through one or more
>> anchor points in a network - perhaps for the sake of drawing the  
>> traffic
>> across a managed circuit with carefully-managed quality, or to
>> facilitate lawful intercept, or what have you. It is that requirement
>> which I have consistently argued is equivalent to exposing this
>> mechanism to a man-in-the-middle attack. If we removed enough
>> protections in the design of RFC4474 to enable media steering by  
>> "good"
>> intermediaries, we are effectively also enabling any malicious  
>> entities
>> to change those aspects of SIP requests as well.
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From HKaplan@acmepacket.com  Thu Nov 19 06:51:42 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Cullen Jennings <fluffy@cisco.com>
Date: Thu, 19 Nov 2009 09:51:35 -0500
Thread-Topic: [dispatch] Identity Adhoc - Nov 9th: Notes available
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Sure, but I'm taking it one bite-size issue at a time, starting with the bi=
ggest bite. :)=20

Some of the things SBC's change *should* fail a 4474-type signature I think=
.  And some of them are either not that commonly done, or are done in the o=
riginating or terminating domain anyway where changing them doesn't matter =
for the mechanism (e.g., transcoding).

-hadriel

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Thursday, November 19, 2009 12:31 AM
> To: Hadriel Kaplan
> Cc: Jon Peterson; Jiri Kuthan; dispatch@ietf.org
> Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
>=20
>=20
> If we all believe the only thing SBC rewrite is the IP and port of the
> media, I'm sure we can solve this in no time flat. Unfortunately, when
> a solution to solve that problem have been presented, it has become
> clear that SBC change much more than that.
>=20
> Cullen <in my individual contributor role>
>=20
> On Nov 18, 2009, at 12:11 PM, Hadriel Kaplan wrote:
>=20
> >
> > Forget the NAT traversal piece, or even media steering, and just
> > tell me what signing the IP Address/port in the SDP buys you.  Not
> > the whole SDP, just the IP/ports.  Imagine if the media IP/ports
> > were in a SIP header, and tell me why we MUST sign that header to
> > get "SIP Identity".
> >
> > My claim is signing those IP/ports is neither necessary nor
> > sufficient for the purposes of SIP Identity nor media identity.
> > It's an emperor with no clothes, except we end up getting the bill
> > for the clothing.
> >
> > It is not sufficient because it does not prove the authenticity and
> > identity of the media.  And I don't just mean in an academic sense
> > of "prove".  Not only can the media IP/ports still be spoofed and
> > even intercepted - but nothing prevents me from just sending media
> > from another address/port to get you to listen to my advertisement,
> > instructions to call my number to "verify your credit card", or
> > whatever.
> >
> > Only a media-plane mechanism will be sufficient for media identity,
> > such as DTLS-SRTP.
> >
> > It is not necessary because, given a mechanism such as DTLS-SRTP
> > (which *is* necessary for the property of media identity), signing
> > the IP/ports is superfluous.  It's making the mistake of confusing
> > IP with an identity, vs. a routing identifier. (ie, the argument HIP
> > makes)
> >
> > -hadriel
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> >> Behalf Of Jon Peterson
> >> Sent: Monday, November 16, 2009 3:45 PM
> >>
> >> To rehash arguments that have circulated in this discussion many
> >> times,
> >> one must distinguish the NAT traversal case from the "media steering"
> >> requirement that has motivated much of the re-examination of
> >> identity.
> >> RFC4474 signatures are applied not by the user agent (at least not in
> >> the typical case), but rather by a server operated by the user
> >> agent's
> >> administrative domain. The administrative domain can more or less
> >> modify
> >> signaling arbitrarily before adding an Identity header per RFC4474.
> >> Thus, some sort of intermediary that modifies SIP signaling for the
> >> sake
> >> of NAT traversal can work with RFC4474, provided that the
> >> intermediary
> >> acts before the Identity header is added.
> >>
> >> While it can therefore be argued that RFC4474 does not rule out
> >> intermediary-based NAT traversal entirely, it does protect against
> >> any
> >> intermediary altering SDP once a request has left the originating
> >> administrative domain with an Identity header. The "media steering"
> >> use
> >> cases that challenge this design are predicated on an intermediary
> >> with
> >> no necessary relationship to the originating or terminating
> >> administrative domain unilaterally modifying SDP in transit,
> >> usually in
> >> order to force layer 3 routing of media to pass through one or more
> >> anchor points in a network - perhaps for the sake of drawing the
> >> traffic
> >> across a managed circuit with carefully-managed quality, or to
> >> facilitate lawful intercept, or what have you. It is that requirement
> >> which I have consistently argued is equivalent to exposing this
> >> mechanism to a man-in-the-middle attack. If we removed enough
> >> protections in the design of RFC4474 to enable media steering by
> >> "good"
> >> intermediaries, we are effectively also enabling any malicious
> >> entities
> >> to change those aspects of SIP requests as well.
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch


From fluffy@cisco.com  Fri Nov 20 12:14:50 2009
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Subject: [dispatch] MARTINI  Charter based on DREGs ad-hoc
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I plan to send the following charter to the IESG on monday unless =
someone catch significant problems with it. The IESG would discuss it on =
the Dec 3 call and would decide if it was ready for IETF LC. At that =
point it would be sent for a 2 week IETF Last Call where everyone could =
comment on it and then come back to the IESG for approval on the Dec 17 =
IESG call.=20

Cullen <RAI AD>

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

Multiple AoR reachabiliTy InformatioN Indication (MARTINI)
=20
Proposed Chair(s):
Spencer Dawkins <spencer@wonderhamster.org>
Bernard Aboba <bernard_aboba@hotmail.com>

Real-time Applications and Infrastructure Area Director(s):
Robert Sparks <rjsparks@nostrum.com>
Cullen Jennings <fluffy@cisco.com>

Real-time Applications and Infrastructure Area Advisor:
Cullen Jennings <fluffy@cisco.com>

Technical Advisor(s):

Mailing Lists:
General Discussion: martini@ietf.org
To Subscribe: martini-request@ietf.org
In Body: subscribe
Archive: http://www.ietf.org/mail-archive/web/martini/index.html

Description of Working Group

The MARTINI working group is chartered to specify a means by which an =
entity
authoritative for SIP URIs can dynamically register reachability =
information for
multiple AoRs with a service provider.

The SIP protocol [RFC 3261 and its extensions] supports multiple means =
of
obtaining the connection information necessary to deliver out-of-dialog =
SIP
requests to their intended targets.  When a SIP Proxy needs to send a =
request to
a target address-of-record (AoR) within its domain, it can use a =
location
service to obtain the registered contact URI, together with any =
associated path
information [RFC 3327], and build a route set to reach the target UA(s). =
 The
SIP REGISTER method can be used to register contact URIs and path
information. SIP-outbound [RFC 5626] enhances this mechanism to cater =
for UAs
behind NATs and firewalls. When a SIP UA or Proxy needs to send a =
request to a
target for which it is not authoritative, the UA/Proxy can use RFC3263
procedures for using DNS to resolve the next-hop connection information.

Since a single SSP may support multiple thousands of such SMB SIP-PBX's, =
it is
impractical and cost-prohibitive to manually provision their IP =
addresses in
every SIP node along paths to those SIP-PBXs.  Furthermore, IP addresses =
can be
dynamically assigned and therefore can potentially change relatively =
frequently.

Today dynamic reachability mechanisms are commonly used, based on the =
SIP
REGISTER method.  Effectively a single REGISTER request registers the =
AoR of the
SIP-PBX, so that any out-of-dialog request targeted at a SIP URI for =
which the
SIP-PBX is authoritative can be delivered from the SSP to the SIP-PBX. =
However,
implementations of this mechanism vary in details, leading to =
interoperability
issues between SIP-PBXs and SSPs, and the need for equipment to support
different variants.

The task of this working group is to to standardize a multiple AoR =
registration
mechanism for SIP that can be widely deployed by service providers at =
large
scale. The solution will support AoRs with E.164 addresses at a minimum,
although support for a broader class of AoRs may be included.

The solution will utilize existing SIP mechanisms to the extent =
possible,
although it is anticipated that small protocol extensions are likely to =
be
required, and hence a standards track (rather than BCP) deliverable is
expected. The solution will accommodate existing SIP extensions relating =
to
registration (e.g., Path, Service-Route [RFC 3608] and SIP-outbound) by =
ensuring
that they are not precluded from use in the context of multiple AoR
registrations. The solution will incorporate a compatibility mechanism =
to ensure
a deterministic outcome when interworking with entities that do not =
support
multiple AoR registration.

The working group will work coordinate with SIP Forum and other industry =
groups
on requirements and other IETF working groups including DRINKS and =
SIPCORE.

Goals and Milestones
Dec 2010     Draft solicitation
Jan 2010     Interim meeting
Feb 2010     First Working Group Last Call
Mar 2010     Second Working Group Last Call
Apr 2010     Multiple AoR Registration specification to IESG (PS)



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From: Jiri Kuthan <jiri@iptel.org>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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Cullen Jennings wrote:
> 
> If we all believe the only thing SBC rewrite is the IP and port of the 
> media, I'm sure we can solve this in no time flat. 

I think there are certainly many who believe this is a reasonable thing
to do ....

> Unfortunately, when a
> solution to solve that problem have been presented, it has become clear 
> that SBC change much more than that.

... and it would be probably more efficient than trying to overload the work
with other interesting features.

what are the next steps then?

-jiri


> 
> Cullen <in my individual contributor role>
> 
> On Nov 18, 2009, at 12:11 PM, Hadriel Kaplan wrote:
> 
>>
>> Forget the NAT traversal piece, or even media steering, and just tell 
>> me what signing the IP Address/port in the SDP buys you.  Not the 
>> whole SDP, just the IP/ports.  Imagine if the media IP/ports were in a 
>> SIP header, and tell me why we MUST sign that header to get "SIP 
>> Identity".
>>
>> My claim is signing those IP/ports is neither necessary nor sufficient 
>> for the purposes of SIP Identity nor media identity.  It's an emperor 
>> with no clothes, except we end up getting the bill for the clothing.
>>
>> It is not sufficient because it does not prove the authenticity and 
>> identity of the media.  And I don't just mean in an academic sense of 
>> "prove".  Not only can the media IP/ports still be spoofed and even 
>> intercepted - but nothing prevents me from just sending media from 
>> another address/port to get you to listen to my advertisement, 
>> instructions to call my number to "verify your credit card", or whatever.
>>
>> Only a media-plane mechanism will be sufficient for media identity, 
>> such as DTLS-SRTP.
>>
>> It is not necessary because, given a mechanism such as DTLS-SRTP 
>> (which *is* necessary for the property of media identity), signing the 
>> IP/ports is superfluous.  It's making the mistake of confusing IP with 
>> an identity, vs. a routing identifier. (ie, the argument HIP makes)
>>
>> -hadriel
>>
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>>> Behalf Of Jon Peterson
>>> Sent: Monday, November 16, 2009 3:45 PM
>>>
>>> To rehash arguments that have circulated in this discussion many times,
>>> one must distinguish the NAT traversal case from the "media steering"
>>> requirement that has motivated much of the re-examination of identity.
>>> RFC4474 signatures are applied not by the user agent (at least not in
>>> the typical case), but rather by a server operated by the user agent's
>>> administrative domain. The administrative domain can more or less modify
>>> signaling arbitrarily before adding an Identity header per RFC4474.
>>> Thus, some sort of intermediary that modifies SIP signaling for the sake
>>> of NAT traversal can work with RFC4474, provided that the intermediary
>>> acts before the Identity header is added.
>>>
>>> While it can therefore be argued that RFC4474 does not rule out
>>> intermediary-based NAT traversal entirely, it does protect against any
>>> intermediary altering SDP once a request has left the originating
>>> administrative domain with an Identity header. The "media steering" use
>>> cases that challenge this design are predicated on an intermediary with
>>> no necessary relationship to the originating or terminating
>>> administrative domain unilaterally modifying SDP in transit, usually in
>>> order to force layer 3 routing of media to pass through one or more
>>> anchor points in a network - perhaps for the sake of drawing the traffic
>>> across a managed circuit with carefully-managed quality, or to
>>> facilitate lawful intercept, or what have you. It is that requirement
>>> which I have consistently argued is equivalent to exposing this
>>> mechanism to a man-in-the-middle attack. If we removed enough
>>> protections in the design of RFC4474 to enable media steering by "good"
>>> intermediaries, we are effectively also enabling any malicious entities
>>> to change those aspects of SIP requests as well.
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
> 

From john.elwell@siemens-enterprise.com  Mon Nov 23 00:42:42 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Cullen Jennings <fluffy@cisco.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Mon, 23 Nov 2009 09:42:33 +0100
Thread-Topic: [dispatch] MARTINI  Charter based on DREGs ad-hoc
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Thanks, Cullen and others, for this initiative. Compared with DREGS, this s=
eems to address the concerns raised during the ad hoc in Hiroshima. There d=
oes seem to be some text missing - see below:=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Cullen Jennings
> Sent: 20 November 2009 20:15
> To: dispatch@ietf.org
> Subject: [dispatch] MARTINI Charter based on DREGs ad-hoc
>=20
>=20
> I plan to send the following charter to the IESG on monday=20
> unless someone catch significant problems with it. The IESG=20
> would discuss it on the Dec 3 call and would decide if it was=20
> ready for IETF LC. At that point it would be sent for a 2=20
> week IETF Last Call where everyone could comment on it and=20
> then come back to the IESG for approval on the Dec 17 IESG call.=20
>=20
> Cullen <RAI AD>
>=20
> ------------
>=20
> Multiple AoR reachabiliTy InformatioN Indication (MARTINI)
> =20
> Proposed Chair(s):
> Spencer Dawkins <spencer@wonderhamster.org>
> Bernard Aboba <bernard_aboba@hotmail.com>
>=20
> Real-time Applications and Infrastructure Area Director(s):
> Robert Sparks <rjsparks@nostrum.com>
> Cullen Jennings <fluffy@cisco.com>
>=20
> Real-time Applications and Infrastructure Area Advisor:
> Cullen Jennings <fluffy@cisco.com>
>=20
> Technical Advisor(s):
>=20
> Mailing Lists:
> General Discussion: martini@ietf.org
> To Subscribe: martini-request@ietf.org
> In Body: subscribe
> Archive: http://www.ietf.org/mail-archive/web/martini/index.html
>=20
> Description of Working Group
>=20
> The MARTINI working group is chartered to specify a means by=20
> which an entity
> authoritative for SIP URIs can dynamically register=20
> reachability information for
> multiple AoRs with a service provider.
>=20
> The SIP protocol [RFC 3261 and its extensions] supports=20
> multiple means of
> obtaining the connection information necessary to deliver=20
> out-of-dialog SIP
> requests to their intended targets.  When a SIP Proxy needs=20
> to send a request to
> a target address-of-record (AoR) within its domain, it can=20
> use a location
> service to obtain the registered contact URI, together with=20
> any associated path
> information [RFC 3327], and build a route set to reach the=20
> target UA(s).  The
> SIP REGISTER method can be used to register contact URIs and path
> information. SIP-outbound [RFC 5626] enhances this mechanism=20
> to cater for UAs
> behind NATs and firewalls. When a SIP UA or Proxy needs to=20
> send a request to a
> target for which it is not authoritative, the UA/Proxy can use RFC3263
> procedures for using DNS to resolve the next-hop connection=20
> information.
>
[JRE] There seems to be something missing prior to the text below. We need =
something to describe the problem, and we also need to introduce the terms =
"SSP" and "SIP-PBX". I propose we add a paragraph:

"Practice has shown that existing mechanisms are not always sufficient for =
supporting SIP-PBXs for small/medium businesses (SMB). A SIP-PBX might serv=
e tens or hundreds of AoRs by acting as the registrar/proxy. UAs register w=
ith the SIP-PBX on behalf of the AoRs concerned, and inbound requests to th=
ose AoRs need to be delivered to the SIP-PBX for onward delivery to registe=
red UAs. Where the SIP-PBX uses a SIP Service Provider (SSP) for external t=
raffic, the SIP-PBX needs to be reachable from the SSP for inbound requests=
 to the AoRs concerned."

John

=20
> Since a single SSP may support multiple thousands of such SMB=20
> SIP-PBX's, it is
> impractical and cost-prohibitive to manually provision their=20
> IP addresses in
> every SIP node along paths to those SIP-PBXs.  Furthermore,=20
> IP addresses can be
> dynamically assigned and therefore can potentially change=20
> relatively frequently.
>=20
> Today dynamic reachability mechanisms are commonly used,=20
> based on the SIP
> REGISTER method.  Effectively a single REGISTER request=20
> registers the AoR of the
> SIP-PBX, so that any out-of-dialog request targeted at a SIP=20
> URI for which the
> SIP-PBX is authoritative can be delivered from the SSP to the=20
> SIP-PBX. However,
> implementations of this mechanism vary in details, leading to=20
> interoperability
> issues between SIP-PBXs and SSPs, and the need for equipment=20
> to support
> different variants.
>=20
> The task of this working group is to to standardize a=20
> multiple AoR registration
> mechanism for SIP that can be widely deployed by service=20
> providers at large
> scale. The solution will support AoRs with E.164 addresses at=20
> a minimum,
> although support for a broader class of AoRs may be included.
>=20
> The solution will utilize existing SIP mechanisms to the=20
> extent possible,
> although it is anticipated that small protocol extensions are=20
> likely to be
> required, and hence a standards track (rather than BCP) deliverable is
> expected. The solution will accommodate existing SIP=20
> extensions relating to
> registration (e.g., Path, Service-Route [RFC 3608] and=20
> SIP-outbound) by ensuring
> that they are not precluded from use in the context of multiple AoR
> registrations. The solution will incorporate a compatibility=20
> mechanism to ensure
> a deterministic outcome when interworking with entities that=20
> do not support
> multiple AoR registration.
>=20
> The working group will work coordinate with SIP Forum and=20
> other industry groups
> on requirements and other IETF working groups including=20
> DRINKS and SIPCORE.
>=20
> Goals and Milestones
> Dec 2010     Draft solicitation
> Jan 2010     Interim meeting
> Feb 2010     First Working Group Last Call
> Mar 2010     Second Working Group Last Call
> Apr 2010     Multiple AoR Registration specification to IESG (PS)
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Victor Pascual Avila <victor.pascual.avila@gmail.com>, Dan Wing <dwing@cisco.com>
Date: Mon, 23 Nov 2009 10:21:09 +0100
Thread-Topic: [dispatch] Identity Adhoc - Nov 9th: Notes available
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References: <F592E36A5C943E4E91F25880D05AD1140CC8E1F3@zrc2hxm0.corp.nortel.com> <4B01A877.4010306@iptel.org> <4B01B9E8.7050507@neustar.biz> <048d01ca67c4$96503410$c2f0200a@cisco.com> <618e24240911171308s316be105k79baaa974ab17963@mail.gmail.com>
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Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
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=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Victor Pascual Avila
> Sent: 17 November 2009 21:09
> To: Dan Wing
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Identity Adhoc - Nov 9th: Notes available
>=20
> On Tue, Nov 17, 2009 at 9:28 PM, Dan Wing <dwing@cisco.com> wrote:
> >> I do not find the requirement for NAT traversal unacceptable,
> >> but I do
> >> consider the requirement for media steering highly dubious at best.
> >
> > "Highly dubious" =3D nobody really does it?
>=20
> Network administrators have actually found it easier to rewrite SDPs
> than to use other existing mechanisms.
[JRE] This is an important point. Those who stand to benefit from e2e (or e=
nd-domain-to-end-domain) authenticated identification are the end users and=
 their domains. Intermediate domains do not benefit. Therefore any mechanis=
m that is harder to deploy at intermediate domains without benefiting those=
 intermediate domains will not be adopted.

John


>=20
> Cheers,
> --=20
> Victor Pascual =C1vila
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
Date: Tue, 24 Nov 2009 21:25:05 +0100
Thread-Topic: SIP Forum UA-config presentation
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=20
Members of the SIP Forum UA-config Task Group tried to arrange a presentati=
on of this work during the Hiroshima IETF, but failed to obtain a room (and=
 the late informal offer for Friday 08.00 was clearly inconvenient for most=
 participants).

Therefore we would be prepared to give the presentation by Webex some time =
during the coming weeks, if there is sufficient interest. The work has prog=
ressed (and changed direction somewhat) since we last talked to DISPATCH (a=
t IETF#74), and in fact it is very nearly ready for Last Call.=20

The UA configuration Task Group's home page is here:
http://www.sipforum.org/content/view/311/253/

The presentation can be found here:
http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,3=
21/Itemid,75/

and the latest draft can be found here:
http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,3=
76/Itemid,75/

If you are interested in a presentation and discussion via Webex, please le=
t me know and suggest some dates (during week beginning 30th November, 7th =
or 14th December).

In any case, please do look at the presentation and draft, and feel free to=
 join our weekly call (Mondays, 16.00 GMT) and/or join the mailing list.

John=

From bernard_aboba@hotmail.com  Tue Nov 24 12:35:27 2009
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Subject: [dispatch] Draft MARTINI charter with corrections incorporated
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Here is an updated version of the MARTINI charter with proposed corrections
incorporated:

 

=========================================================================

Multiple AoR reachabiliTy INformatIon (MARTINI)

 

Proposed Chair(s):

Spencer Dawkins <spencer at wonderhamster.org>

Bernard Aboba <bernard_aboba at hotmail.com>

 

Real-time Applications and Infrastructure Area Director(s):

Robert Sparks <rjsparks at nostrum.com>

Cullen Jennings <fluffy at cisco.com>

 

Real-time Applications and Infrastructure Area Advisor:

Cullen Jennings <fluffy at cisco.com>

 

Technical Advisor(s):

 

Mailing Lists:

General Discussion: martini at ietf.org

To Subscribe: martini-request at ietf.org

In Body: subscribe

Archive: http://www.ietf.org/mail-archive/web/martini/index.html

 

Description of Working Group

 

The MARTINI working group is chartered to specify a means by 

which an entity authoritative for SIP URIs can dynamically register 

reachability information for multiple AoRs with a service provider.

 

The SIP protocol [RFC 3261 and its extensions] supports 

multiple means of obtaining the connection information necessary 

to deliver out-of-dialog SIP requests to their intended targets.  

When a SIP Proxy needs  to send a request to a target address-of-record 

(AoR) within its domain, it can use a location service to obtain the 

registered contact URI, together with any associated path

information [RFC 3327], and build a route set to reach the 

target UA(s).  The SIP REGISTER method can be used to register contact 

URIs and path information. SIP-outbound [RFC 5626] enhances this mechanism 

to cater for UAs behind NATs and firewalls. When a SIP UA or Proxy needs to 

send a request to a target for which it is not authoritative, the UA/Proxy 

can use RFC 3263 procedures for using DNS to resolve the next-hop connection


information.

 

Practice has shown that existing mechanisms are not always sufficient for 

supporting SIP-PBXs for small/medium businesses (SMB). A SIP-PBX might serve


tens or hundreds of AoRs by acting as the registrar/proxy. UAs register with


the SIP-PBX on behalf of the AoRs concerned, and inbound requests to those 

AoRs need to be delivered to the SIP-PBX for onward delivery to registered
UAs. 

Where the SIP-PBX uses a SIP Service Provider (SSP) for external traffic,
the 

SIP-PBX needs to be reachable from the SSP for inbound requests to the AoRs 

concerned.

 

Since a single SSP may support multiple thousands of such SMB 

SIP-PBX's, it is impractical and cost-prohibitive to manually provision
their 

IP addresses in every SIP node along paths to those SIP-PBXs.  Furthermore, 

IP addresses can be dynamically assigned and therefore can potentially
change 

relatively frequently.

 

Today dynamic reachability mechanisms are commonly used, 

based on the SIP REGISTER method.  Effectively a single REGISTER request 

registers the AoR of the SIP-PBX, so that any out-of-dialog request targeted


at a SIP URI for which the SIP-PBX is authoritative can be delivered from
the 

SSP to the SIP-PBX. However, implementations of this mechanism vary in
details, 

leading to interoperability issues between SIP-PBXs and SSPs, and the need
for 

equipment to support different variants.

 

The task of this working group is to to standardize a multiple AoR
registration 

mechanism for SIP that can be widely deployed by service providers at large

scale. The solution will support AoRs with E.164 addresses at 

a minimum, although support for a broader class of AoRs may be included.

 

The solution will utilize existing SIP mechanisms to the extent possible, 

although it is anticipated that small protocol extensions are likely to be

required, and hence a standards track (rather than BCP) deliverable is

expected. The solution will accommodate existing SIP extensions relating to

registration (e.g., Path, Service-Route [RFC 3608] and SIP-outbound) by
ensuring

that they are not precluded from use in the context of multiple AoR

registrations. The solution will incorporate a compatibility mechanism to 

ensure a deterministic outcome when interworking with entities that 

do not support multiple AoR registration.

 

The working group will work coordinate with SIP Forum and other industry
groups 

on requirements and other IETF working groups including DRINKS and SIPCORE.

 

Goals and Milestones

Dec 2009     Draft solicitation

Jan 2010     Interim meeting

Feb 2010     First Working Group Last Call

Mar 2010     Second Working Group Last Call

Apr 2010     Multiple AoR Registration specification to IESG (PS)


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<div class=3DSection1>

<p class=3DMsoNormal>Here is an updated version of the MARTINI charter =
with
proposed corrections incorporated:<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p =
class=3DMsoNormal>=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=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<o:p></o:p></p>

<p class=3DMsoNormal>Multiple AoR reachabiliTy INformatIon =
(MARTINI)<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Proposed Chair(s):<o:p></o:p></p>

<p class=3DMsoNormal>Spencer Dawkins &lt;spencer at =
wonderhamster.org&gt;<o:p></o:p></p>

<p class=3DMsoNormal>Bernard Aboba &lt;bernard_aboba at =
hotmail.com&gt;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Real-time Applications and Infrastructure Area =
Director(s):<o:p></o:p></p>

<p class=3DMsoNormal>Robert Sparks &lt;rjsparks at =
nostrum.com&gt;<o:p></o:p></p>

<p class=3DMsoNormal>Cullen Jennings &lt;fluffy at =
cisco.com&gt;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Real-time Applications and Infrastructure Area =
Advisor:<o:p></o:p></p>

<p class=3DMsoNormal>Cullen Jennings &lt;fluffy at =
cisco.com&gt;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Technical Advisor(s):<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Mailing Lists:<o:p></o:p></p>

<p class=3DMsoNormal>General Discussion: martini at =
ietf.org<o:p></o:p></p>

<p class=3DMsoNormal>To Subscribe: martini-request at =
ietf.org<o:p></o:p></p>

<p class=3DMsoNormal>In Body: subscribe<o:p></o:p></p>

<p class=3DMsoNormal>Archive:
http://www.ietf.org/mail-archive/web/martini/index.html<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Description of Working Group<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The MARTINI working group is chartered to specify a =
means by
<o:p></o:p></p>

<p class=3DMsoNormal>which an entity authoritative for SIP URIs can =
dynamically
register <o:p></o:p></p>

<p class=3DMsoNormal>reachability information for multiple AoRs with a =
service
provider.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The SIP protocol [RFC 3261 and its extensions] =
supports <o:p></o:p></p>

<p class=3DMsoNormal>multiple means of obtaining the connection =
information
necessary <o:p></o:p></p>

<p class=3DMsoNormal>to deliver out-of-dialog SIP requests to their =
intended
targets.&nbsp; <o:p></o:p></p>

<p class=3DMsoNormal>When a SIP Proxy needs&nbsp; to send a request to a =
target
address-of-record <o:p></o:p></p>

<p class=3DMsoNormal>(AoR) within its domain, it can use a location =
service to
obtain the <o:p></o:p></p>

<p class=3DMsoNormal>registered contact URI, together with any =
associated path<o:p></o:p></p>

<p class=3DMsoNormal>information [RFC 3327], and build a route set to =
reach the <o:p></o:p></p>

<p class=3DMsoNormal>target UA(s).&nbsp; The SIP REGISTER method can be =
used to
register contact <o:p></o:p></p>

<p class=3DMsoNormal>URIs and path information. SIP-outbound [RFC 5626] =
enhances
this mechanism <o:p></o:p></p>

<p class=3DMsoNormal>to cater for UAs behind NATs and firewalls. When a =
SIP UA or
Proxy needs to <o:p></o:p></p>

<p class=3DMsoNormal>send a request to a target for which it is not
authoritative, the UA/Proxy <o:p></o:p></p>

<p class=3DMsoNormal>can use RFC 3263 procedures for using DNS to =
resolve the
next-hop connection <o:p></o:p></p>

<p class=3DMsoNormal>information.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Practice has shown that existing mechanisms are not =
always sufficient
for <o:p></o:p></p>

<p class=3DMsoNormal>supporting SIP-PBXs for small/medium businesses =
(SMB). A
SIP-PBX might serve <o:p></o:p></p>

<p class=3DMsoNormal>tens or hundreds of AoRs by acting as the =
registrar/proxy.
UAs register with <o:p></o:p></p>

<p class=3DMsoNormal>the SIP-PBX on behalf of the AoRs concerned, and =
inbound
requests to those <o:p></o:p></p>

<p class=3DMsoNormal>AoRs need to be delivered to the SIP-PBX for onward =
delivery
to registered UAs. <o:p></o:p></p>

<p class=3DMsoNormal>Where the SIP-PBX uses a SIP Service Provider (SSP) =
for
external traffic, the <o:p></o:p></p>

<p class=3DMsoNormal>SIP-PBX needs to be reachable from the SSP for =
inbound
requests to the AoRs <o:p></o:p></p>

<p class=3DMsoNormal>concerned.<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>Since a single SSP may support multiple thousands =
of such
SMB <o:p></o:p></p>

<p class=3DMsoNormal>SIP-PBX's, it is impractical and cost-prohibitive =
to
manually provision their <o:p></o:p></p>

<p class=3DMsoNormal>IP addresses in every SIP node along paths to those
SIP-PBXs.&nbsp; Furthermore, <o:p></o:p></p>

<p class=3DMsoNormal>IP addresses can be dynamically assigned and =
therefore can
potentially change <o:p></o:p></p>

<p class=3DMsoNormal>relatively frequently.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Today dynamic reachability mechanisms are commonly =
used, <o:p></o:p></p>

<p class=3DMsoNormal>based on the SIP REGISTER method.&nbsp; Effectively =
a single
REGISTER request <o:p></o:p></p>

<p class=3DMsoNormal>registers the AoR of the SIP-PBX, so that any =
out-of-dialog
request targeted <o:p></o:p></p>

<p class=3DMsoNormal>at a SIP URI for which the SIP-PBX is authoritative =
can be
delivered from the <o:p></o:p></p>

<p class=3DMsoNormal>SSP to the SIP-PBX. However, implementations of =
this
mechanism vary in details, <o:p></o:p></p>

<p class=3DMsoNormal>leading to interoperability issues between SIP-PBXs =
and
SSPs, and the need for <o:p></o:p></p>

<p class=3DMsoNormal>equipment to support different =
variants.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The task of this working group is to to standardize =
a
multiple AoR registration <o:p></o:p></p>

<p class=3DMsoNormal>mechanism for SIP that can be widely deployed by =
service
providers at large<o:p></o:p></p>

<p class=3DMsoNormal>scale. The solution will support AoRs with E.164 =
addresses
at <o:p></o:p></p>

<p class=3DMsoNormal>a minimum, although support for a broader class of =
AoRs may
be included.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The solution will utilize existing SIP mechanisms =
to the
extent possible, <o:p></o:p></p>

<p class=3DMsoNormal>although it is anticipated that small protocol =
extensions
are likely to be<o:p></o:p></p>

<p class=3DMsoNormal>required, and hence a standards track (rather than =
BCP)
deliverable is<o:p></o:p></p>

<p class=3DMsoNormal>expected. The solution will accommodate existing =
SIP
extensions relating to<o:p></o:p></p>

<p class=3DMsoNormal>registration (e.g., Path, Service-Route [RFC 3608] =
and
SIP-outbound) by ensuring<o:p></o:p></p>

<p class=3DMsoNormal>that they are not precluded from use in the context =
of
multiple AoR<o:p></o:p></p>

<p class=3DMsoNormal>registrations. The solution will incorporate a =
compatibility
mechanism to <o:p></o:p></p>

<p class=3DMsoNormal>ensure a deterministic outcome when interworking =
with
entities that <o:p></o:p></p>

<p class=3DMsoNormal>do not support multiple AoR =
registration.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The working group will work coordinate with SIP =
Forum and
other industry groups <o:p></o:p></p>

<p class=3DMsoNormal>on requirements and other IETF working groups =
including
DRINKS and SIPCORE.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Goals and Milestones<o:p></o:p></p>

<p class=3DMsoNormal>Dec 2009&nbsp;&nbsp;&nbsp;&nbsp; Draft =
solicitation<o:p></o:p></p>

<p class=3DMsoNormal>Jan 2010&nbsp;&nbsp;&nbsp;&nbsp; Interim =
meeting<o:p></o:p></p>

<p class=3DMsoNormal>Feb 2010&nbsp;&nbsp;&nbsp;&nbsp; First Working =
Group Last Call<o:p></o:p></p>

<p class=3DMsoNormal>Mar 2010&nbsp;&nbsp;&nbsp;&nbsp; Second Working =
Group Last Call<o:p></o:p></p>

<p class=3DMsoNormal>Apr 2010&nbsp;&nbsp;&nbsp;&nbsp; Multiple AoR =
Registration specification to IESG
(PS)<o:p></o:p></p>

</div>

</body>

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From gonzalo.camarillo@ericsson.com  Thu Nov 26 09:35:44 2009
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Subject: [dispatch] Draft minutes of the main DISPATCH session
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Folks,

here you have the draft minutes of the main DISPATCH session.

http://www.ietf.org/proceedings/09nov/minutes/dispatch.txt

Comments are welcome.

We are working with the cochairs of the three ad-hoc meetings we had in 
Japan to prepare the minutes of those meetings as well (the minutes will 
be based on the raw notes that have already been distributed). As soon 
as those are ready, we will append them to the main DISPATCH minutes 
above so that everything is archived at the IETF.

Cheers,

Gonzalo
DISPATCH co-chair
