
From wwwrun@core3.amsl.com  Mon Jul  6 09:33:19 2009
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Subject: [Sip] Protocol Action: 'Message Body Handling in the Session Initiation Protocol (SIP)' to Proposed Standard
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The IESG has approved the following document:

- 'Message Body Handling in the Session Initiation Protocol (SIP) '
   <draft-ietf-sip-body-handling-06.txt> as a Proposed Standard

This document is the product of the Session Initiation Protocol Working 
Group. 

The IESG contact persons are Robert Sparks and Cullen Jennings.

A URL of this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-body-handling-06.txt

Technical Summary

  This document specifies how message bodies should be handled in SIP.
  Additionally, this document specifies SIP user agent support for MIME
  (Multipurpose Internet Mail Extensions) in message bodies.

Working Group Summary

  There is consensus in the working group to publish this document, and 
  is targeted for Standards Track.  Work on this document began in
  May 2007, and was adopted as a working group item in August 2007. 
  WGLC was issued in June 2008.

Document Quality

  This document specifically addresses an area of SIP that has been an
  interoperability problem in the past.  The SIPit interoperability 
  events have seen many problems in the area of interoperability of 
  MIME  handling.

  This document has been reviews by many participants over the lifetime 
  of the document, by the following members of the WG:

  - Paul Kyzivat
  - John Elwell
  - Francois Audet
  - Dan Wing
  - Eric Burger
  - Dale Worley
  - Jonathan Rosenberg
  - Cullen Jennings
  - Adam Roach

  Additionally, an extensive APPS area review of the document was been 
  performed by Dave Crocker in an early version of the this document.


Personnel

  Theo Zourzouvillys is the Document Shepherd.
  Robert Sparks is the Responsible Area Director.

RFC Editor Note

  (applies to version 06 of this document)

  The XML in figure 2 is invalid due to a missing '>'.  
  The Content-Length in the same message is also invalid.

  Please replace line 230, which is currently:
      Content-Length: 617
  with:
      Content-Length: 619


  Please replace line 250, which is currently:
      <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"
  with:
      <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists">


From adam@nostrum.com  Tue Jul  7 12:21:52 2009
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Subject: Re: [Sip] New HTTP Event Package Draft
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Theo Zourzouvillys wrote:
> 2.2, para 3:
>
>    [...] If a server
>    wishes to reject such requests, the "403 Forbidden" response code is
>    appropriate.
>
> It would be good to provide a way of returning to the client which
> ones it was unhappy with in the response, to save it having to go off
> and perform a HEAD on each original resource to find out which one has
> changed.
>   

Good point. I've modified it to read:

   [...] If a server
   wishes to reject such requests, the "403" (Forbidden) response code
   is appropriate.  Any "403" responses generated for this reason SHOULD
   contain a body of type "application/resource-lists+xml"; this body
   lists the offending URI or URIs.  See RFC 4826 [6] for the definition
   of the "application/resource-lists+xml" MIME type.


> 2.2, para 5:
>
>   Finally, for each such group, it initiates a subscription to the
> group's monitor-list URI;
>
> do you mean "monitor-group" instead of "monitor-list"?
>   

Yep. Good catch.

/a

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Subject: [Sip] I-D Action:draft-ietf-sip-record-route-fix-07.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.


	Title           : Addressing Record-Route issues in the Session Initiation Protocol (SIP)
	Author(s)       : T. Froment, et al.
	Filename        : draft-ietf-sip-record-route-fix-07.txt
	Pages           : 26
	Date            : 2009-07-08

A typical function of a Session Initiation Protocol (SIP) Proxy is to
insert a Record-Route header into initial, dialog creating requests
in order to make subsequent, in-dialog requests pass through it.
This header contains a SIP Uniform Resource Identifier (URI)
indicating where and how the subsequent requests should be sent to
reach the proxy.  Like any SIP URI, it can contain SIP or SIPS
schemes, IPv4 or IPv6 addresses, and URI parameters that could
influence the routing such as the transport parameter (for example
transport=tcp), or a compression indication like "comp=sigcomp".
When a proxy has to change some of those parameters between its
incoming and outgoing interfaces (multi-homed proxies, transport
protocol switching or IPv4 to IPv6 scenarios...), the question arises
on what should be put in Record-Route header(s).  It is just not
possible to make one header having the characteristics of both sides
at the same time.  This document aims to clarify these scenarios and
fix bugs already identified on this topic; it formally recommends the
use of the double Record-Route technique as an alternative to the
current RFC3261 text, which describes only a Record-Route rewriting
solution.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-record-route-fix-07.txt

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.


	Title           : An Extensible Markup Language (XML) Configuration Access Protocol (XCAP) Diff Event Package
	Author(s)       : J. Urpalainen, D. Willis
	Filename        : draft-ietf-sip-xcapevent-08.txt
	Pages           : 29
	Date            : 2009-07-09

This document describes an "xcap-diff" SIP (Session Initiation
Protocol) event package for the SIP Event Notification Framework,
which clients can use to receive notifications of changes to
Extensible Markup Language (XML) Configuration Access Protocol (XCAP)
resources.  The initial synchronization information exchange and
document updates are based on the XCAP Diff format.

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.


	Title           : Certificate Management Service for The Session Initiation Protocol (SIP)
	Author(s)       : C. Jennings, J. Fischl
	Filename        : draft-ietf-sip-certs-08.txt
	Pages           : 28
	Date            : 2009-07-13

This draft defines a Credential Service that allows Session
Initiation Protocol (SIP) User Agents (UAs) to use a SIP event
package to discover the certificates of other users.  This mechanism
allows user agents that want to contact a given Address-of-Record
(AOR) to retrieve that AOR's certificate by subscribing to the
Credential Service, which returns an authenticated response
containing that certificate.  The Credential Service also allows
users to store and retrieve their own certificates and private keys.

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From Karunesh.Sharma@globallogic.com  Wed Jul 29 20:58:26 2009
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List:

=20

I am having a question about SIP dialog validation using src/dst ip. If
there is established sip dialog between 'A' & 'B' then can 'C' who is
not party to existing dialog and who somehow knows all sip dialog
identifiers (Call-ID, To/From Tags) of the dialog between 'A' & 'B',
sends target refresh request with its own IP in contact. This is a
typical call hijacking case. If this is valid how to avoid that? Is
Authentication is only way out or do we have other alternate as well?

=20

I didn't find anything on this in 3261 or other related RFCs.

=20

Thanks

=20

K$

_______________________

=20


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xmlns:b=3D"urn:schemas-microsoft-com:office:publisher" =
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xmlns:html=3D"http://www.w3.org/TR/REC-html40" =
xmlns:q=3D"http://schemas.xmlsoap.org/soap/envelope/" =
xmlns:rtc=3D"http://microsoft.com/officenet/conferencing" =
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xmlns:mt=3D"http://schemas.microsoft.com/sharepoint/soap/meetings/" =
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<p class=3DMsoNormal><span =
style=3D'font-family:Consolas;color:black'>List:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:Consolas;color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:Consolas;color:black'>I =
am having a
question about SIP dialog validation using src/dst ip. If there is =
established
sip dialog between &#8216;A&#8217; &amp; &#8216;B&#8217; then can =
&#8216;C&#8217; who is not party to existing
dialog and who somehow knows all sip dialog identifiers (Call-ID, =
To/From Tags)
of the dialog between &#8216;A&#8217; &amp; &#8216;B&#8217;, sends =
target refresh request with its own
IP in contact. This is a typical call hijacking case. If this is valid =
how to
avoid that? Is Authentication is only way out or do we have other =
alternate as
well?<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:Consolas;color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:Consolas;color:black'>I =
didn&#8217;t find
anything on this in 3261 or other related RFCs.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:Consolas;color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:Consolas;color:black'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:Consolas;color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:Consolas;
color:black'>K$<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#FF8000'>_______________________</span><span =
style=3D'color:black'><o:p></o:p></span></p>

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

</div>

</body>

</html>

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From rjsparks@nostrum.com  Fri Jul 31 02:56:57 2009
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Subject: [Sip] Record-Route-fix
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In case you haven't been following the progress of Record-Route-fix in  
the tracker, it's been through IESG review and the editors are  
addressing a few comments.
One point the IESG made is that this document should move forward as  
Proposed Standard rather than BCP and the next revision of the draft  
will reflect that.

RjS

