
From drage@alcatel-lucent.com  Tue Jun  2 07:24:22 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: "sip@ietf.org" <sip@ietf.org>
Date: Tue, 2 Jun 2009 16:23:30 +0200
Thread-Topic: [sip] I-D Action:draft-ietf-sip-outbound-19.txt
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Subject: [Sip] [sip] I-D Action:draft-ietf-sip-outbound-19.txt
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Continuing my attempt to keep these going to the correct list, which is mea=
nt to be STILL sip.

And a reminder that it is the author's responsibility to update the list wi=
th the nature of the changes when a new version is issues.

regards

Keith =20

> -----Original Message-----
> From: dispatch-bounces@ietf.org
> [mailto:dispatch-bounces@ietf.org] On Behalf Of=20
> Internet-Drafts@ietf.org
> Sent: Monday, June 01, 2009 7:00 PM
> To: i-d-announce@ietf.org
> Cc: dispatch@ietf.org
> Subject: [dispatch] I-D Action:draft-ietf-sip-outbound-19.txt
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts=20
> directories.
> This draft is a work item of the Dispatch Working Group of the IETF.
>=20
>=20
> 	Title           : Managing Client Initiated Connections=20
> in the Session Initiation Protocol (SIP)
> 	Author(s)       : C. Jennings
> 	Filename        : draft-ietf-sip-outbound-19.txt
> 	Pages           : 49
> 	Date            : 2009-06-01
>=20
> The Session Initiation Protocol (SIP) allows proxy servers to initiate=20
> TCP connections or to send asynchronous UDP datagrams to User Agents=20
> in order to deliver requests.
> However, in a large number of real deployments, many practical=20
> considerations, such as the existence of firewalls and Network Address=20
> Translators (NATs) or the use of TLS with server-provided=20
> certificates, prevent servers from connecting to User Agents in this=20
> way.  This specification defines behaviors for User Agents, registrars=20
> and proxy servers that allow requests to be delivered on existing=20
> connections established by the User Agent.  It also defines keep alive=20
> behaviors needed to keep NAT bindings open and specifies the usage of=20
> multiple connections from the User Agent to its Registrar.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-19.txt
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> Below is the data which will enable a MIME compliant mail reader=20
> implementation to automatically retrieve the ASCII version of the=20
> Internet-Draft.
> =

From mary.barnes@nortel.com  Tue Jun  2 07:27:34 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>
Subject: Re: [Sip] [sip] I-D Action:draft-ietf-sip-outbound-19.txt
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There's one single change per the diff:
http://tools.ietf.org/wg/dispatch/draft-ietf-sip-outbound/draft-ietf-sip
-outbound-19-from-18.diff.html

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
DRAGE, Keith (Keith)
Sent: Tuesday, June 02, 2009 9:24 AM
To: sip@ietf.org
Subject: [Sip] [sip] I-D Action:draft-ietf-sip-outbound-19.txt

Continuing my attempt to keep these going to the correct list, which is
meant to be STILL sip.

And a reminder that it is the author's responsibility to update the list
with the nature of the changes when a new version is issues.

regards

Keith =20

> -----Original Message-----
> From: dispatch-bounces@ietf.org
> [mailto:dispatch-bounces@ietf.org] On Behalf Of=20
> Internet-Drafts@ietf.org
> Sent: Monday, June 01, 2009 7:00 PM
> To: i-d-announce@ietf.org
> Cc: dispatch@ietf.org
> Subject: [dispatch] I-D Action:draft-ietf-sip-outbound-19.txt
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts=20
> directories.
> This draft is a work item of the Dispatch Working Group of the IETF.
>=20
>=20
> 	Title           : Managing Client Initiated Connections=20
> in the Session Initiation Protocol (SIP)
> 	Author(s)       : C. Jennings
> 	Filename        : draft-ietf-sip-outbound-19.txt
> 	Pages           : 49
> 	Date            : 2009-06-01
>=20
> The Session Initiation Protocol (SIP) allows proxy servers to initiate

> TCP connections or to send asynchronous UDP datagrams to User Agents=20
> in order to deliver requests.
> However, in a large number of real deployments, many practical=20
> considerations, such as the existence of firewalls and Network Address

> Translators (NATs) or the use of TLS with server-provided=20
> certificates, prevent servers from connecting to User Agents in this=20
> way.  This specification defines behaviors for User Agents, registrars

> and proxy servers that allow requests to be delivered on existing=20
> connections established by the User Agent.  It also defines keep alive

> behaviors needed to keep NAT bindings open and specifies the usage of=20
> multiple connections from the User Agent to its Registrar.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-19.txt
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> Below is the data which will enable a MIME compliant mail reader=20
> implementation to automatically retrieve the ASCII version of the=20
> Internet-Draft.
>=20
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use
sip-implementors@cs.columbia.edu for questions on current sip Use
sipping@ietf.org for new developments on the application of sip

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From Glenn.Cahall@Level3.com  Thu Jun  4 09:45:39 2009
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From: "Cahall, Glenn" <Glenn.Cahall@Level3.com>
To: "chogg@nortelnetworks.com" <chogg@nortelnetworks.com>, "jdrosen@dynamicsoft.com" <jdrosen@dynamicsoft.com>, "zmolek@avaya.com" <zmolek@avaya.com>, "Marie-Noelle.Mathieu@compaq.com" <Marie-Noelle.Mathieu@compaq.com>, "sip@ietf.org" <sip@ietf.org>
Date: Thu, 4 Jun 2009 10:45:33 -0600
Thread-Topic: Reuse of Sip Call-Id question...
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Subject: [Sip] Reuse of Sip Call-Id question...
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Folks,

I work for a VoIP company, Level3 Communications, and have been doing some =
research on the internet about
a specific condition that has arisen here dealing with the Sip protocol.  Y=
our names came up during my research.
Was hoping I could get some feedback about the reuse of Sip Call-Id.

We have customer UAs that are reusing the same Sip Call-Id for multiple cal=
l attempts into our network.

After reviewing the Sip RFC 3261, both myself and my team concluded that th=
is was a clear violation of the spec.  based
on the following verbiage...

8.1.1.4 Call-ID

   The Call-ID header field acts as a unique identifier to group
   together a series of messages.  It MUST be the same for all requests
   and responses sent by either UA in a dialog.  It SHOULD be the same
   in each registration from a UA.

   In a new request created by a UAC outside of any dialog, the Call-ID
   header field MUST be selected by the UAC as a globally unique
   identifier over space and time unless overridden by method-specific
   behavior.  All SIP UAs must have a means to guarantee that the Call-
   ID header fields they produce will not be inadvertently generated by
   any other UA.  Note that when requests are retried after certain



However, some voice engineers are stating that the definition of "globally =
unique over time and space"  means only during the
context of a given call.  Therefore, once this context is over....Sip Call-=
Id can in fact be reused in the very next context.


What is your take on this?


Thanks,
Glenn E. Cahall
Sr. Architect
Level3 Communications







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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Folks,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>I work for a VoIP company, Level3 Communications, and ha=
ve
been doing some research on the internet about <o:p></o:p></span></font></p=
>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>a specific condition that has arisen here dealing with t=
he
Sip protocol.&nbsp; Your names came up during my research.<o:p></o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Was hoping I could get some feedback about the reuse of =
Sip
Call-Id.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>We have customer UAs that are reusing the same Sip Call-=
Id
for multiple call attempts into our network.&nbsp; <o:p></o:p></span></font=
></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>After reviewing the Sip RFC 3261, both myself and my tea=
m
concluded that this was a clear violation of the spec.&nbsp; based <o:p></o=
:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>on the following verbiage...<o:p></o:p></span></font></p=
>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>8.1.1.4 Call-ID<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>&nbsp;&nbsp; The Call-ID header field acts as a unique i=
dentifier to
group<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>&nbsp;&nbsp; together a series of messages.&nbsp; It MUS=
T be the same for
all requests<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>&nbsp;&nbsp; and responses sent by either UA in a dialog=
.&nbsp; It SHOULD
be the same<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>&nbsp;&nbsp; in each registration from a UA.<o:p></o:p><=
/span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>&nbsp;&nbsp; In a new request created by a UAC outside o=
f any dialog,
the Call-ID<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>&nbsp;&nbsp; header field MUST be selected by the UAC as=
 a globally
unique<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>&nbsp;&nbsp; identifier over space and time unless overr=
idden by
method-specific<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>&nbsp;&nbsp; behavior.&nbsp; All SIP UAs must have a mea=
ns to guarantee
that the Call-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>&nbsp;&nbsp; ID header fields they produce will not be i=
nadvertently
generated by<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>&nbsp;&nbsp; any other UA.&nbsp; Note that when requests=
 are retried after
certain<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>However, some voice engineers are stating that the
definition of &#8220;globally unique over time and space&#8221;&nbsp; means=
 only
during the <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>context of a given call.&nbsp; Therefore, once this cont=
ext is
over....Sip Call-Id can in fact be reused in the very next context.<o:p></o=
:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>What is your take on this?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Thanks,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Glenn E. Cahall<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Sr. Architect<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Level3 Communications<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
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Thread-Topic: [Sip] Reuse of Sip Call-Id question...
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Subject: Re: [Sip] Reuse of Sip Call-Id question...
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you will get the same response here as on sip-core.
=20
SIP call-id should be unique for each call, it cannot be re-used
=20
Regards,

Attila
=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Cahall, Glenn
Sent: 04 June 2009 17:46
To: chogg@nortelnetworks.com; jdrosen@dynamicsoft.com; zmolek@avaya.com;
Marie-Noelle.Mathieu@compaq.com; sip@ietf.org
Subject: [Sip] Reuse of Sip Call-Id question...



Folks,

I work for a VoIP company, Level3 Communications, and have been doing
some research on the internet about=20

a specific condition that has arisen here dealing with the Sip protocol.
Your names came up during my research.

Was hoping I could get some feedback about the reuse of Sip Call-Id.

We have customer UAs that are reusing the same Sip Call-Id for multiple
call attempts into our network.=20=20

After reviewing the Sip RFC 3261, both myself and my team concluded that
this was a clear violation of the spec.  based=20

on the following verbiage...

8.1.1.4 Call-ID

   The Call-ID header field acts as a unique identifier to group

   together a series of messages.  It MUST be the same for all requests

   and responses sent by either UA in a dialog.  It SHOULD be the same

   in each registration from a UA.

   In a new request created by a UAC outside of any dialog, the Call-ID

   header field MUST be selected by the UAC as a globally unique

   identifier over space and time unless overridden by method-specific

   behavior.  All SIP UAs must have a means to guarantee that the Call-

   ID header fields they produce will not be inadvertently generated by

   any other UA.  Note that when requests are retried after certain

However, some voice engineers are stating that the definition of
"globally unique over time and space"  means only during the=20

context of a given call.  Therefore, once this context is over....Sip
Call-Id can in fact be reused in the very next context.

What is your take on this?

Thanks,

Glenn E. Cahall

Sr. Architect

Level3 Communications


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML defanghtml_xmlns=3D"http://www.w3.org/TR/REC-html40"=20
defanghtml_xmlns:w=3D"urn:schemas-microsoft-com:office:word"=20
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<BODY lang=3DEN-US vLink=3Dpurple link=3Dblue>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D822391315-05062009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>you will get the same response here as on=20
sip-core.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D822391315-05062009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D822391315-05062009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>SIP call-id should be unique for each call, it can=
not be=20
re-used</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D822391315-05062009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D822391315-05062009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D822391315-05062009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2><BR>Attila</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D822391315-05062009></SPAN><SPAN=
=20
class=3D822391315-05062009></SPAN><SPAN class=3D822391315-05062009></SPAN><=
SPAN=20
class=3D822391315-05062009><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Cahall, Glenn<BR><B>Sent:=
</B>=20
04 June 2009 17:46<BR><B>To:</B> chogg@nortelnetworks.com;=20
jdrosen@dynamicsoft.com; zmolek@avaya.com; Marie-Noelle.Mathieu@compaq.com;=
=20
sip@ietf.org<BR><B>Subject:</B> [Sip] Reuse of Sip Call-Id=20
question...<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Folks,<O:P></O:P></DEFANGHTML=
_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">I work for a VoIP company, Le=
vel3=20
Communications, and have been doing some research on the internet about=20
<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">a specific condition that has=
 arisen=20
here dealing with the Sip protocol.&nbsp; Your names came up during my=20
research.<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Was hoping I could get some f=
eedback=20
about the reuse of Sip Call-Id.<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">We have customer UAs that are=
=20
reusing the same Sip Call-Id for multiple call attempts into our network.&n=
bsp;=20
<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">After reviewing the Sip RFC 3=
261,=20
both myself and my team concluded that this was a clear violation of the=20
spec.&nbsp; based <O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">on the following=20
verbiage...<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">8.1.1.4=20
Call-ID<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbsp;&nbsp; The Call-ID head=
er=20
field acts as a unique identifier to=20
group<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbsp;&nbsp; together a serie=
s of=20
messages.&nbsp; It MUST be the same for all=20
requests<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbsp;&nbsp; and responses se=
nt by=20
either UA in a dialog.&nbsp; It SHOULD be the=20
same<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbsp;&nbsp; in each registra=
tion=20
from a UA.<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbsp;&nbsp; In a new request=
=20
created by a UAC outside of any dialog, the=20
Call-ID<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbsp;&nbsp; header field MUS=
T be=20
selected by the UAC as a globally unique<O:P></O:P></DEFANGHTML_SPAN></FONT=
></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbsp;&nbsp; identifier over =
space=20
and time unless overridden by=20
method-specific<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbsp;&nbsp; behavior.&nbsp; =
All SIP=20
UAs must have a means to guarantee that the=20
Call-<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbsp;&nbsp; ID header fields=
 they=20
produce will not be inadvertently generated=20
by<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbsp;&nbsp; any other UA.&nb=
sp;=20
Note that when requests are retried after=20
certain<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">However, some voice engineers=
 are=20
stating that the definition of &#8220;globally unique over time and space&#=
8221;&nbsp; means=20
only during the <O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">context of a given call.&nbsp=
;=20
Therefore, once this context is over....Sip Call-Id can in fact be reused i=
n the=20
very next context.<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">What is your take on=20
this?<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Thanks,<O:P></O:P></DEFANGHTM=
L_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Glenn E.=20
Cahall<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Sr.=20
Architect<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Level3=20
Communications<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><O:P></O:P></DEFANGHTML_SPAN>=
</FONT></P></DIV><br><br>
<P align=3Dleft><FONT style=3D"BACKGROUND-COLOR: #ffffff">.</FONT></P>
</body></HTML>

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Date: Fri, 5 Jun 2009 11:45:36 -0400
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From: Martin Larochelle <martin@macadamian.com>
To: Attila Sipos <attila.sipos@vegastream.com>
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Cc: sip@ietf.org, chogg@nortelnetworks.com, Marie-Noelle.Mathieu@compaq.com, jdrosen@dynamicsoft.com, zmolek@avaya.com
Subject: Re: [Sip] Reuse of Sip Call-Id question...
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--0022152d655998b4aa046b9bc8d6
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This is right, although it does not mean that each call attempt must use a
new call-id.You wrote that "We have customer UAs that are reusing the same
Sip Call-Id for multiple call attempts into our network.", if by attempt,
you mean trying to make a call that fails with a error that can be retried
(as 407, 422) then in that case, it is acceptable to use the same call-id (
with an incremented C-Seq and empty to-tag)

--=20
Martin Larochelle
Software Architect


On Fri, Jun 5, 2009 at 11:15 AM, Attila Sipos
<attila.sipos@vegastream.com>wrote:

>  you will get the same response here as on sip-core.
>
> SIP call-id should be unique for each call, it cannot be re-used
>
> Regards,
>
> Attila
>
>
>  ------------------------------
> *From:* sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] *On Behalf Of =
*Cahall,
> Glenn
> *Sent:* 04 June 2009 17:46
> *To:* chogg@nortelnetworks.com; jdrosen@dynamicsoft.com; zmolek@avaya.com=
;
> Marie-Noelle.Mathieu@compaq.com; sip@ietf.org
> *Subject:* [Sip] Reuse of Sip Call-Id question...
>
>  Folks,
>
> I work for a VoIP company, Level3 Communications, and have been doing som=
e
> research on the internet about
>
> a specific condition that has arisen here dealing with the Sip protocol.
> Your names came up during my research.
>
> Was hoping I could get some feedback about the reuse of Sip Call-Id.
>
> We have customer UAs that are reusing the same Sip Call-Id for multiple
> call attempts into our network.
>
> After reviewing the Sip RFC 3261, both myself and my team concluded that
> this was a clear violation of the spec.  based
>
> on the following verbiage...
>
> 8.1.1.4 Call-ID
>
>    The Call-ID header field acts as a unique identifier to group
>
>    together a series of messages.  It MUST be the same for all requests
>
>    and responses sent by either UA in a dialog.  It SHOULD be the same
>
>    in each registration from a UA.
>
>    In a new request created by a UAC outside of any dialog, the Call-ID
>
>    header field MUST be selected by the UAC as a globally unique
>
>    identifier over space and time unless overridden by method-specific
>
>    behavior.  All SIP UAs must have a means to guarantee that the Call-
>
>    ID header fields they produce will not be inadvertently generated by
>
>    any other UA.  Note that when requests are retried after certain
>
>  However, some voice engineers are stating that the definition of
> =93globally unique over time and space=94  means only during the
>
> context of a given call.  Therefore, once this context is over....Sip
> Call-Id can in fact be reused in the very next context.
>
>  What is your take on this?
>
>  Thanks,
>
> Glenn E. Cahall
>
> Sr. Architect
>
> Level3 Communications
>
>
>
> .
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

--0022152d655998b4aa046b9bc8d6
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable

This is right, although it does not mean that each call attempt must use a =
new call-id.<div>You wrote that &quot;<span class=3D"Apple-style-span" styl=
e=3D"border-collapse: collapse; font-family: Arial; ">We have customer UAs =
that are reusing the same Sip Call-Id for multiple call attempts into our n=
etwork.<span class=3D"Apple-style-span" style=3D"border-collapse: separate;=
 font-family: arial; ">&quot;, if by attempt, you mean trying to make a cal=
l that fails with a error that can be retried (as 407, 422) then in that ca=
se, it is acceptable to use the same call-id ( with an incremented C-Seq an=
d empty to-tag)</span></span></div>
<div><br></div><div>-- <br>Martin Larochelle<br>Software Architect<br>
<br><br><div class=3D"gmail_quote">On Fri, Jun 5, 2009 at 11:15 AM, Attila =
Sipos <span dir=3D"ltr">&lt;<a href=3D"mailto:attila.sipos@vegastream.com">=
attila.sipos@vegastream.com</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex;">






<div lang=3D"EN-US" vlink=3D"purple" link=3D"blue">
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2">you will get the same response here as on=20
sip-core.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2"></font></span>=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2">SIP call-id should be unique for each call, it cannot be=20
re-used</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2"></font></span>=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2">Regards,</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2"><br>Attila</font></span></div>
<div dir=3D"ltr" align=3D"left"><span></span><span></span><span></span><spa=
n><font face=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>=A0</div>=
<br>
<div lang=3D"en-us" dir=3D"ltr" align=3D"left">
<hr>
<font face=3D"Tahoma" size=3D"2"><b>From:</b> <a href=3D"mailto:sip-bounces=
@ietf.org" target=3D"_blank">sip-bounces@ietf.org</a>=20
[mailto:<a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bounc=
es@ietf.org</a>] <b>On Behalf Of </b>Cahall, Glenn<br><b>Sent:</b>=20
04 June 2009 17:46<br><b>To:</b> <a href=3D"mailto:chogg@nortelnetworks.com=
" target=3D"_blank">chogg@nortelnetworks.com</a>;=20
<a href=3D"mailto:jdrosen@dynamicsoft.com" target=3D"_blank">jdrosen@dynami=
csoft.com</a>; <a href=3D"mailto:zmolek@avaya.com" target=3D"_blank">zmolek=
@avaya.com</a>; <a href=3D"mailto:Marie-Noelle.Mathieu@compaq.com" target=
=3D"_blank">Marie-Noelle.Mathieu@compaq.com</a>;=20
<a href=3D"mailto:sip@ietf.org" target=3D"_blank">sip@ietf.org</a><br><b>Su=
bject:</b> [Sip] Reuse of Sip Call-Id=20
question...<br></font><br></div><div><div></div><div class=3D"h5">
<div></div>
<div>
<p><font face=3D"Arial" size=3D"2">Folks,</font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2">I work for a VoIP company, Level3=20
Communications, and have been doing some research on the internet about=20
</font></p>
<p><font face=3D"Arial" size=3D"2">a specific condition that has arisen=20
here dealing with the Sip protocol.=A0 Your names came up during my=20
research.</font></p>
<p><font face=3D"Arial" size=3D"2">Was hoping I could get some feedback=20
about the reuse of Sip Call-Id.</font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2">We have customer UAs that are=20
reusing the same Sip Call-Id for multiple call attempts into our network.=
=A0=20
</font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2">After reviewing the Sip RFC 3261,=20
both myself and my team concluded that this was a clear violation of the=20
spec.=A0 based </font></p>
<p><font face=3D"Arial" size=3D"2">on the following=20
verbiage...</font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2">8.1.1.4=20
Call-ID</font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2">=A0=A0 The Call-ID header=20
field acts as a unique identifier to=20
group</font></p>
<p><font face=3D"Arial" size=3D"2">=A0=A0 together a series of=20
messages.=A0 It MUST be the same for all=20
requests</font></p>
<p><font face=3D"Arial" size=3D"2">=A0=A0 and responses sent by=20
either UA in a dialog.=A0 It SHOULD be the=20
same</font></p>
<p><font face=3D"Arial" size=3D"2">=A0=A0 in each registration=20
from a UA.</font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2">=A0=A0 In a new request=20
created by a UAC outside of any dialog, the=20
Call-ID</font></p>
<p><font face=3D"Arial" size=3D"2">=A0=A0 header field MUST be=20
selected by the UAC as a globally unique</font></p>
<p><font face=3D"Arial" size=3D"2">=A0=A0 identifier over space=20
and time unless overridden by=20
method-specific</font></p>
<p><font face=3D"Arial" size=3D"2">=A0=A0 behavior.=A0 All SIP=20
UAs must have a means to guarantee that the=20
Call-</font></p>
<p><font face=3D"Arial" size=3D"2">=A0=A0 ID header fields they=20
produce will not be inadvertently generated=20
by</font></p>
<p><font face=3D"Arial" size=3D"2">=A0=A0 any other UA.=A0=20
Note that when requests are retried after=20
certain</font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2">However, some voice engineers are=20
stating that the definition of =93globally unique over time and space=94=A0=
 means=20
only during the </font></p>
<p><font face=3D"Arial" size=3D"2">context of a given call.=A0=20
Therefore, once this context is over....Sip Call-Id can in fact be reused i=
n the=20
very next context.</font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2">What is your take on=20
this?</font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2">Thanks,</font></p>
<p><font face=3D"Arial" size=3D"2">Glenn E.=20
Cahall</font></p>
<p><font face=3D"Arial" size=3D"2">Sr.=20
Architect</font></p>
<p><font face=3D"Arial" size=3D"2">Level3=20
Communications</font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2"></font></p>
<p><font face=3D"Arial" size=3D"2"></font></p></div><br><br>
</div></div><p align=3D"left"><font style=3D"background-color:#ffffff">.</f=
ont></p>
</div>
<br>_______________________________________________<br>
Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sip" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs=
.columbia.edu</a> for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new develo=
pments on the application of sip<br></blockquote></div><br></div>

--0022152d655998b4aa046b9bc8d6--

From wwwrun@core3.amsl.com  Mon Jun  8 11:14:07 2009
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Subject: [Sip] Document Action: 'UA-Driven Privacy Mechanism for SIP' to Informational RFC
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The IESG has approved the following document:

- 'UA-Driven Privacy Mechanism for SIP '
   <draft-ietf-sip-ua-privacy-08.txt> as an Informational RFC

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

The IESG contact persons are Cullen Jennings and Robert Sparks.

A URL of this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-ua-privacy-08.txt

Technical summary.

This document defines a guideline for a User Agent (UA) to generate an
anonymous Session Initiation Protocol (SIP) message by utilizing
mechanisms such as Globally Routable User Agent URIs (GRUU) and Traversal
Using Relays around NAT (TURN) without the need for a privacy service
defined in RFC 3323.

Working group summary.

There is consensus in the working group to publish this document. 

Document Quality

There has been no specific indication of implementation.

Personnel

The document shepherd for this document was Keith Drage. The responsible
Area Director was Cullen Jennings

RFC Editor Note

In Section 5.1.2 
OLD
     sip:anonymous@atlanta.com
NEW
    sip:anonymous@example.com


IRTF Note
  none

IESG Note
  none

IANA Note
  none


From rjs@nostrum.com  Tue Jun  9 14:12:52 2009
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Subject: Re: [Sip] Draft ietf-sip-xcapevent revised, many open questions
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I'm going through -07 in detail now to figure out the next step, and  
just went back
through the email threads to make sure nothing got dropped.

I think we still have a rough edge related to what's below (and I  
don't understand Jari's response
now that I've read the draft carefully).

My read is that the default was intended to be the simplest (and  
required to implement) mode of
"no-patching" and that we just had a single typo in the first  
paragraph of section 4.3. The last
sentence in that first paragraph should say "no-patching" rather than  
"xcap-patching" and
the conflict that Dean points to below is resolved.

RjS

On May 27, 2009, at 5:07 AM, Jari Urpalainen wrote:

>> 1)  What's the default notification mode supposed to be? "xml-
>> patching" is listed as the default and fail-safe in some text, but  
>> "no-
>> patching" is also described as the singular "must implement" mode. It
>> seems to me that if a mode is the default if no mode is specified,
>> then that mode had better be the default.
>> This question is a primary source of confusion in Section 4.3, and  
>> I'd
>> appreciate some feedback. This whole section got a lot of edits, you
>> may want to usea difftool.
>>
> for implementers this is no big deal, although you MUST know which is
> the default if the parameter is missing, anyways it can be anyone of
> those, so i'm fine with this technical change.


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From: Jari Urpalainen <jari.urpalainen@nokia.com>
To: ext Robert Sparks <rjs@nostrum.com>
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Cc: "sip@ietf.org List" <sip@ietf.org>, ext Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Draft ietf-sip-xcapevent revised, many open questions
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On Tue, 2009-06-09 at 23:12 +0200, ext Robert Sparks wrote:
> I'm going through -07 in detail now to figure out the next step, and  
> just went back
> through the email threads to make sure nothing got dropped.
> 
> I think we still have a rough edge related to what's below (and I  
> don't understand Jari's response
> now that I've read the draft carefully).
> 
Sorry, i did not read the text that carefully (rather only Dean's
question)

My previous comment was about the default mode if client won't give any.
In 04 it was "xcap-patching" but it can be any of those, and Dean
proposed "no-patching" which is fine.

> My read is that the default was intended to be the simplest (and  
> required to implement) mode of
> "no-patching" and that we just had a single typo in the first  
> paragraph of section 4.3. The last
> sentence in that first paragraph should say "no-patching" rather than  
> "xcap-patching" and
> the conflict that Dean points to below is resolved.
> 
> RjS
> 
Right, the last sentence in 4.3 is _wrong_. The subscribers MAY
implement any of the modes, but it doesn't need to support patching at
all. The same applies to the notifier with the exception that it MUST
implement xcap-component subscriptions. The last sentence of the first
chapter of 4.3 is also wrong since the client could only subscribe to
xcap component changes, so it'd better to remove the last part ", as all
subscribers are required to support that mode."

br, jari


From wwwrun@core3.amsl.com  Wed Jun 10 14:34:18 2009
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Subject: [Sip] CORRECTED Protocol Action: 'Managing Client Initiated Connections in the Session Initiation Protocol (SIP)' to Proposed Standard
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The IESG has approved the following document:

- 'Managing Client Initiated Connections in the Session Initiation
Protocol (SIP) '
<draft-ietf-sip-outbound-20.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-outbound-20.txt

Technical Summary

This document defines an extension to the Session Initiation Protocol
(SIP) that provides for persistent and reusable connections between
SIP User Agents and SIP Proxy Servers. In particular, this allows
proxy servers to initiate TCP connections or to send asynchronous UDP
datagrams to User Agents in order to deliver requests. However, in a
large number of real deployments, many practical considerations, such
as the existence of firewalls and Network Address Translators (NATs)
or the use of TLS with server-provided certificates, prevent servers
from connecting to User Agents in this way. This specification
defines behaviors for User Agents, registrars and proxy servers that
allow requests to be delivered on existing connections established by
the User Agent. It also defines keep alive behaviors needed to keep
NAT bindings open and specifies the usage of multiple connections from
the User Agent to its Registrar.

Working Group Summary

The working group process on this document was exceptionally long. The


first WG version of the draft appeared in the summer of 2005. Working
group last call initiated in the summer of 2006 and extended until the
summer of 2008, requiring several iterations of the draft and the
assignment of Francois Audet as a "process champion" for the draft
within the working group. Most delays seem to have been related to
slow cycle time on the part of the authors, but the process was also
delayed by a number of changes occurring during the review cycle.

Particular sticking points included the keepalive mechanism and a
requirement for binding to multiple outbound proxies if so
configured. Both were eventually resolved by a widely-accepted
compromises. Additional work is being pursued in SIPCORE to allow
the use of the keepalive mechanism independently of the outbound
mechanism.

Document Quality

There have been implementations of previous versions of this
draft testing at SIPit events since 2005.

Personnel

Dean Willis is the document shepherd with significant assistance
from Francois Audet. Robert Sparks is the responsible area director.

From dean.willis@softarmor.com  Wed Jun 10 20:59:02 2009
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Outbound has been approved by the IESG.

-------- Original Message --------
Subject: I-D Action:draft-ietf-sip-outbound-20.txt
Date: Wed, 10 Jun 2009 13:15:01 -0700 (PDT)
From: Internet-Drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org
CC: dispatch@ietf.org

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


	Title           : Managing Client Initiated Connections in the Session
Initiation Protocol (SIP)
	Author(s)       : C. Jennings
	Filename        : draft-ietf-sip-outbound-20.txt
	Pages           : 49
	Date            : 2009-06-10

The Session Initiation Protocol (SIP) allows proxy servers to
initiate TCP connections or to send asynchronous UDP datagrams to
User Agents in order to deliver requests.  However, in a large number
of real deployments, many practical considerations, such as the
existence of firewalls and Network Address Translators (NATs) or the
use of TLS with server-provided certificates, prevent servers from
connecting to User Agents in this way.  This specification defines
behaviors for User Agents, registrars and proxy servers that allow
requests to be delivered on existing connections established by the
User Agent.  It also defines keep alive behaviors needed to keep NAT
bindings open and specifies the usage of multiple connections from
the User Agent to its Registrar.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-20.txt

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

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From dean.willis@softarmor.com  Wed Jun 10 20:59:24 2009
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Subject: [Sip] [Fwd: [dispatch] Protocol Action: 'Managing Client Initiated Connections in the Session Initiation Protocol (SIP)' to Proposed Standard]
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-------- Original Message --------
Subject: [dispatch] Protocol Action: 'Managing Client Initiated
Connections in the Session Initiation Protocol (SIP)' to Proposed Standard
Date: Wed, 10 Jun 2009 14:13:37 -0700 (PDT)
From: The IESG <iesg-secretary@ietf.org>
To: IETF-Announce <ietf-announce@ietf.org>
CC: dispatch mailing list <dispatch@ietf.org>,        dispatch chair
<dispatch-chairs@tools.ietf.org>,        Internet Architecture Board
<iab@iab.org>,        RFC Editor <rfc-editor@rfc-editor.org>

The IESG has approved the following document:

- 'Managing Client Initiated Connections in the Session Initiation
   Protocol (SIP) '
   <draft-ietf-sip-outbound-20.txt> as a Proposed Standard

This document is the product of the Dispatch 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-outbound-20.txt

Technical Summary

   This document defines an extension to the Session Initiation Protocol
   (SIP) that provides for persistent and reusable connections between
   SIP User Agents and SIP Proxy Servers. In particular, this allows
   proxy servers to initiate TCP connections or to send asynchronous UDP
   datagrams to User Agents in order to deliver requests.  However, in a
   large number of real deployments, many practical considerations, such
   as the existence of firewalls and Network Address Translators (NATs)
   or the use of TLS with server-provided certificates, prevent servers
   from connecting to User Agents in this way.  This specification
   defines behaviors for User Agents, registrars and proxy servers that
   allow requests to be delivered on existing connections established by
   the User Agent.  It also defines keep alive behaviors needed to keep
   NAT bindings open and specifies the usage of multiple connections from
   the User Agent to its Registrar.

Working Group Summary

   The working group process on this document was exceptionally long. The


   first WG version of the draft appeared in the summer of 2005. Working
   group last call initiated in the summer of 2006 and extended until the
   summer of 2008, requiring several iterations of the draft and the
   assignment of Francois Audet as a "process champion" for the draft
   within the working group. Most delays seem to have been related to
   slow cycle time on the part of the authors, but the process was also
   delayed by a number of changes occurring during the review cycle.

   Particular sticking points included the keepalive mechanism and a
   requirement for binding to multiple outbound proxies if so
   configured. Both were eventually resolved by a widely-accepted
   compromises. Additional work is being pursued in SIPCORE to allow
   the use of the keepalive mechanism independently of the outbound
   mechanism.

Document Quality

  There have been implementations of previous versions of this
  draft testing at SIPit events since 2005.

Personnel

   Dean Willis is the document shepherd with significant assistance
   from Francois Audet. Robert Sparks is the responsible area director.

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


From wwwrun@core3.amsl.com  Wed Jun 17 09:06:42 2009
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Date: Wed, 17 Jun 2009 09:06:42 -0700 (PDT)
Cc: sip@ietf.org
Subject: [Sip] Last Call: draft-ietf-sip-body-handling (Message Body Handling in the Session Initiation Protocol (SIP)) to Proposed Standard
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The IESG has received a request from the Session Initiation Protocol WG 
(sip) to consider the following document:

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

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send substantive comments to the
ietf@ietf.org mailing lists by 2009-07-01. Exceptionally, 
comments may be sent to iesg@ietf.org instead. In either case, please 
retain the beginning of the Subject line to allow automated sorting.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-body-handling-06.txt


IESG discussion can be tracked via
https://datatracker.ietf.org/public/pidtracker.cgi?command=view_id&dTag=16410&rfc_flag=0


From rjsparks@nostrum.com  Tue Jun 23 08:59:56 2009
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This is ready to go and I've asked it to be IETF LCed.

One thing that we will need to fix: The IPv6 address bearing SIP URIs  
in the examples are not right - the literals need to be enclosed in  
'[' ']' symbols
(as in sip:bob@[2001:db8::9:1]).

There is an outstanding issue being worked on reconciling the SIP (and  
other scheme's) URI handling of IPv6 literals with the genericURI  
syntax.
That should not hold up this document (the impact is only in some  
examples after all).

Some other early last call comments:

The last paragraph of the introduction (about essential corrections)  
is a holdover from earlier versions, before we decided to move this  
forward as a BCP.
It should probably be removed.

There is a sentence at the end of the quote on page 7 that should be  
split off into its own paragraph (Starting "It is to notice that").

A reference into RFC3261 to explain spirals (at the third paragraph of  
section 5) would help readers who aren't already very familiar with  
SIP proxy behavior.

Consider reinforcing that it is a proxy in proxy 2's situation that  
you are talking about in the 2nd to last paragraph of section 6.2.


RjS

From wwwrun@core3.amsl.com  Tue Jun 23 09:00:52 2009
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Subject: [Sip] Last Call: draft-ietf-sip-record-route-fix (Addressing Record-Route issues in the Session Initiation Protocol (SIP)) to BCP
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The IESG has received a request from the Session Initiation Protocol WG 
(sip) to consider the following document:

- 'Addressing Record-Route issues in the Session Initiation Protocol 
   (SIP) '
   <draft-ietf-sip-record-route-fix-06.txt> as a BCP

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send substantive comments to the
ietf@ietf.org mailing lists by 2009-07-07. Exceptionally, 
comments may be sent to iesg@ietf.org instead. In either case, please 
retain the beginning of the Subject line to allow automated sorting.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-record-route-fix-06.txt


IESG discussion can be tracked via
https://datatracker.ietf.org/public/pidtracker.cgi?command=view_id&dTag=16198&rfc_flag=0

