
From abhatnagar192006@gmail.com  Sat Aug  1 06:47:52 2009
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From: aayush bhatnagar <abhatnagar192006@gmail.com>
Date: Sat, 1 Aug 2009 19:17:31 +0530
Message-ID: <abcc59880908010647h1a4ae3f6m3f5feb4cfb71e1ac@mail.gmail.com>
To: Karunesh Sharma <Karunesh.Sharma@globallogic.com>
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Cc: sip@ietf.org
Subject: Re: [Sip] SIP Dialog Match
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--0016e645a3d86da50b047014c8c1
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It is possible..and is standardized. 'C' can 'logically' Join an existing
dialog between 'A' (UAC) and 'B' (UAS), by including a Join header in the
INVITE request.
It may be a security concern to allow C to Join the existing dialog, but it
can be a way to implement a call screening server (call tapping) for lawful
interception services.

For more details look here: http://www.rfc-editor.org/rfc/rfc3911.txt
<http://www.rfc-editor.org/rfc/rfc3911.txt>

On Thu, Jul 30, 2009 at 09:28, Karunesh Sharma <
Karunesh.Sharma@globallogic.com> wrote:

>  List:
>
>
>
> I am having a question about SIP dialog validation using src/dst ip. If
> there is established sip dialog between =91A=92 & =91B=92 then can =91C=
=92 who is not
> party to existing dialog and who somehow knows all sip dialog identifiers
> (Call-ID, To/From Tags) of the dialog between =91A=92 & =91B=92, sends ta=
rget
> refresh request with its own IP in contact. This is a typical call hijack=
ing
> case. If this is valid how to avoid that? Is Authentication is only way o=
ut
> or do we have other alternate as well?
>
>
>
> I didn=92t find anything on this in 3261 or other related RFCs.
>
>
>
> Thanks
>
>
>
> K$
>
> _______________________
>
>
>
> _______________________________________________
> 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
>



--=20
aayush
----------------------------------------------------
"A philosopher always knows what to do...until it happens to him"

--0016e645a3d86da50b047014c8c1
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It is possible..and is standardized. &#39;C&#39; can &#39;logically&#39; Jo=
in an existing dialog between &#39;A&#39; (UAC) and &#39;B&#39; (UAS), by i=
ncluding a Join header in the INVITE request.<div><br></div><div>It may be =
a security concern to allow C to Join the existing dialog, but it can be a =
way to implement a call screening server (call tapping) for lawful intercep=
tion services.<br>

<div><br></div><div>For more details look here:=A0<a href=3D"http://www.rfc=
-editor.org/rfc/rfc3911.txt">http://www.rfc-editor.org/rfc/rfc3911.txt</a><=
/div><div><a href=3D"http://www.rfc-editor.org/rfc/rfc3911.txt"></a><br><br=
>
<div class=3D"gmail_quote">
On Thu, Jul 30, 2009 at 09:28, Karunesh Sharma <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:Karunesh.Sharma@globallogic.com">Karunesh.Sharma@globallogic.co=
m</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">










<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">

<div>

<p><span style=3D"font-family:Consolas;color:black">List:</span></p>

<p><span style=3D"font-family:Consolas;color:black">=A0</span></p>

<p><span style=3D"font-family:Consolas;color:black">I am having a
question about SIP dialog validation using src/dst ip. If there is establis=
hed
sip dialog between =91A=92 &amp; =91B=92 then can =91C=92 who is not party =
to existing
dialog and who somehow knows all sip dialog identifiers (Call-ID, To/From T=
ags)
of the dialog between =91A=92 &amp; =91B=92, sends target refresh request w=
ith 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?</span></p>

<p><span style=3D"font-family:Consolas;color:black">=A0</span></p>

<p><span style=3D"font-family:Consolas;color:black">I didn=92t find
anything on this in 3261 or other related RFCs.</span></p>

<p><span style=3D"font-family:Consolas;color:black">=A0</span></p>

<p><span style=3D"font-family:Consolas;color:black">Thanks</span></p>

<p><span style=3D"font-family:Consolas;color:black">=A0</span></p>

<p><span style=3D"font-size:12.0pt;font-family:Consolas;color:black">K$</sp=
an></p>

<p><span style=3D"font-size:10.0pt;color:#FF8000">_______________________</=
span><span style=3D"color:black"></span></p>

<p>=A0</p>

</div>

</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><br clear=3D"all=
"><br>-- <br>aayush<br>----------------------------------------------------=
<br>

&quot;A philosopher always knows what to do...until it happens to him&quot;=
<br>
</div></div>

--0016e645a3d86da50b047014c8c1--

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Cc: sip@ietf.org, Karunesh Sharma <Karunesh.Sharma@globallogic.com>
Subject: Re: [Sip] SIP Dialog Match
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> On Thu, Jul 30, 2009 at 09:28, Karunesh Sharma
> <Karunesh.Sharma@globallogic.com> wrote:

>         I am having a question about SIP dialog validation using
>         src/dst ip. If there is established sip dialog between =E2=80=98A=
=E2=80=99 &
>         =E2=80=98B=E2=80=99 then can =E2=80=98C=E2=80=99 who is not party=
 to existing dialog and who
>         somehow knows all sip dialog identifiers (Call-ID, To/From
>         Tags) of the dialog between =E2=80=98A=E2=80=99 & =E2=80=98B=E2=
=80=99, 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?

For "basic" security, it is usually assumed that if an agent knows the
dialog identifiers, then it has the right to manipulate the dialog.
(See, e.g., the Replace and Join headers.)  For stricter security,
additional authentication mechanisms have to be deployed, but the IETF
has not standardized any such mechanism.

Dale




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To: Karunesh Sharma <Karunesh.Sharma@globallogic.com>
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On Mon, 2009-08-03 at 20:49 +0530, Karunesh Sharma wrote:
> What about sending INVITE to 'B' & getting all responses (which
> includes provisional response 1xx) from 'C'?

I'm not sure what you mean.

Dale



From Karunesh.Sharma@globallogic.com  Mon Aug  3 08:34:10 2009
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From: "Karunesh Sharma" <Karunesh.Sharma@globallogic.com>
To: "Dale Worley" <dworley@nortel.com>
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Dale:

Please see following trace

172.16.31.30 -> 10.0.4.6     SIP/SDP Request: INVITE
sip:23235556@10.0.4.6, with session description
192.168.4.67 -> 172.16.31.30 SIP Status: 100 Trying
192.168.4.67 -> 172.16.31.30 SIP/SDP Status: 183 Session Progress, with
session description
172.16.31.30 -> 192.168.4.67 SIP Request: PRACK
sip:23235556@192.168.4.67:5060
192.168.4.67 -> 172.16.31.30 SIP Status: 200 OK
192.168.4.67 -> 172.16.31.30 SIP/SDP Status: 200 OK, with session
description
172.16.31.30 -> 192.168.4.67 SIP Request: ACK
sip:23235556@192.168.4.67:5060
192.168.4.67 -> 172.16.31.30 SIP Request: BYE
sip:24098955@172.16.31.30:5060

172.16.31.30 (read as 'A') sent INVITE to 10.0.4.6 (read as 'B'). But
somehow 192.168.4.67 (read as 'C') responds. Now 'A' knows nothing about
'C', but still based on SIP dialog identifiers 'A' accepts that.

So is there any way to challenge or deny such rouge responses?

I was reading one draft
(http://ietfreport.isoc.org/idref/draft-cao-sip-response-identity/) but
that is expired.

K$
_______________________


-----Original Message-----
From: Dale Worley [mailto:dworley@nortel.com]=20
Sent: Monday, August 03, 2009 8:55 PM
To: Karunesh Sharma
Cc: aayush bhatnagar; sip@ietf.org
Subject: RE: [Sip] SIP Dialog Match

On Mon, 2009-08-03 at 20:49 +0530, Karunesh Sharma wrote:
> What about sending INVITE to 'B' & getting all responses (which
> includes provisional response 1xx) from 'C'?

I'm not sure what you mean.

Dale



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On Mon, 2009-08-03 at 21:03 +0530, Karunesh Sharma wrote:
> 172.16.31.30 (read as 'A') sent INVITE to 10.0.4.6 (read as 'B'). But
> somehow 192.168.4.67 (read as 'C') responds. Now 'A' knows nothing about
> 'C', but still based on SIP dialog identifiers 'A' accepts that.
> 
> So is there any way to challenge or deny such rouge responses?

Have you inquired as to how C knows of the dialog identifiers, and why
it is sending the responses?

Dale



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On Mon, 2009-08-03 at 21:21 +0530, Karunesh Sharma wrote:
> I am trying to get that info how 'C' has this dialog identifiers but I
> was wondering this as a major security issue for any SIP network.

If C can obtain the dialog identifiers but it is not supposed to be
participating in the dialog, then that is a major security issue.

Dale



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On Mon, 2009-08-03 at 21:28 +0530, Karunesh Sharma wrote:
> Is this a valid behavior for scenario like 'Call-Forwarding'?

In principle, the responses to a request should almost always come from
the address to which the request was sent.  But that is not usually
enforced, and there may be good reasons why it isn't happening in a
particular instance.

In regard to an observed instance of this behavior, you must first
determine the complete routing of the request in question and the reason
for each routing step.

Dale



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From: "Karunesh Sharma" <Karunesh.Sharma@globallogic.com>
To: "Dale Worley" <dworley@nortel.com>
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Subject: Re: [Sip] SIP Dialog Match
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Is this a valid behavior for scenario like 'Call-Forwarding'?

K$
_______________________


-----Original Message-----
From: Karunesh Sharma=20
Sent: Monday, August 03, 2009 9:22 PM
To: 'Dale Worley'
Cc: aayush bhatnagar; sip@ietf.org
Subject: RE: [Sip] SIP Dialog Match

I am trying to get that info how 'C' has this dialog identifiers but I
was wondering this as a major security issue for any SIP network.

K$
_______________________


-----Original Message-----
From: Dale Worley [mailto:dworley@nortel.com]=20
Sent: Monday, August 03, 2009 9:17 PM
To: Karunesh Sharma
Cc: aayush bhatnagar; sip@ietf.org
Subject: RE: [Sip] SIP Dialog Match

On Mon, 2009-08-03 at 21:03 +0530, Karunesh Sharma wrote:
> 172.16.31.30 (read as 'A') sent INVITE to 10.0.4.6 (read as 'B'). But
> somehow 192.168.4.67 (read as 'C') responds. Now 'A' knows nothing
about
> 'C', but still based on SIP dialog identifiers 'A' accepts that.
>=20
> So is there any way to challenge or deny such rouge responses?

Have you inquired as to how C knows of the dialog identifiers, and why
it is sending the responses?

Dale



From Karunesh.Sharma@globallogic.com  Mon Aug  3 09:22:32 2009
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I am trying to get that info how 'C' has this dialog identifiers but I
was wondering this as a major security issue for any SIP network.

K$
_______________________


-----Original Message-----
From: Dale Worley [mailto:dworley@nortel.com]=20
Sent: Monday, August 03, 2009 9:17 PM
To: Karunesh Sharma
Cc: aayush bhatnagar; sip@ietf.org
Subject: RE: [Sip] SIP Dialog Match

On Mon, 2009-08-03 at 21:03 +0530, Karunesh Sharma wrote:
> 172.16.31.30 (read as 'A') sent INVITE to 10.0.4.6 (read as 'B'). But
> somehow 192.168.4.67 (read as 'C') responds. Now 'A' knows nothing
about
> 'C', but still based on SIP dialog identifiers 'A' accepts that.
>=20
> So is there any way to challenge or deny such rouge responses?

Have you inquired as to how C knows of the dialog identifiers, and why
it is sending the responses?

Dale



From satyanarayana.t@motorola.com  Mon Aug  3 09:35:00 2009
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References: <5F063FC3F0C8F14E945465A68E63C43E9451DB@ex3-del1.synapse.com><abcc59880908010647h1a4ae3f6m3f5feb4cfb71e1ac@mail.gmail.com><1249249277.4106.1.camel@victoria-pingtel-com.us.nortel.com><5F063FC3F0C8F14E945465A68E63C43E945346@ex3-del1.synapse.com><1249313088.3791.0.camel@victoria-pingtel-com.us.nortel.com><5F063FC3F0C8F14E945465A68E63C43E945347@ex3-del1.synapse.com><1249314398.3791.5.camel@victoria-pingtel-com.us.nortel.com><5F063FC3F0C8F14E945465A68E63C43E94534A@ex3-del1.synapse.com> <1249315489.3791.10.camel@victoria-pingtel-com.us.nortel.com>
From: "T Satyanarayana-A12694" <satyanarayana.t@motorola.com>
To: "Dale Worley" <dworley@nortel.com>, "Karunesh Sharma" <Karunesh.Sharma@globallogic.com>
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In short, if some one wants to fake another client, it is possible since
the communication is in clear. If there are security concerns, then
security mechanisms (privacy etc) should be used.

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Dale Worley
Sent: Monday, August 03, 2009 9:35 PM
To: Karunesh Sharma
Cc: sip@ietf.org
Subject: Re: [Sip] SIP Dialog Match

On Mon, 2009-08-03 at 21:28 +0530, Karunesh Sharma wrote:
> Is this a valid behavior for scenario like 'Call-Forwarding'?

In principle, the responses to a request should almost always come from
the address to which the request was sent.  But that is not usually
enforced, and there may be good reasons why it isn't happening in a
particular instance.

In regard to an observed instance of this behavior, you must first
determine the complete routing of the request in question and the reason
for each routing step.

Dale


_______________________________________________
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

From Karunesh.Sharma@globallogic.com  Mon Aug  3 09:37:42 2009
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From: "Karunesh Sharma" <Karunesh.Sharma@globallogic.com>
To: "T Satyanarayana-A12694" <satyanarayana.t@motorola.com>, "Dale Worley" <dworley@nortel.com>
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Are you talking about *privacy* headers? I fail to understand here how
*privacy* header will help in such cases.

K$
_______________________


-----Original Message-----
From: T Satyanarayana-A12694 [mailto:satyanarayana.t@motorola.com]=20
Sent: Monday, August 03, 2009 10:02 PM
To: Dale Worley; Karunesh Sharma
Cc: sip@ietf.org
Subject: RE: [Sip] SIP Dialog Match


In short, if some one wants to fake another client, it is possible since
the communication is in clear. If there are security concerns, then
security mechanisms (privacy etc) should be used.

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Dale Worley
Sent: Monday, August 03, 2009 9:35 PM
To: Karunesh Sharma
Cc: sip@ietf.org
Subject: Re: [Sip] SIP Dialog Match

On Mon, 2009-08-03 at 21:28 +0530, Karunesh Sharma wrote:
> Is this a valid behavior for scenario like 'Call-Forwarding'?

In principle, the responses to a request should almost always come from
the address to which the request was sent.  But that is not usually
enforced, and there may be good reasons why it isn't happening in a
particular instance.

In regard to an observed instance of this behavior, you must first
determine the complete routing of the request in question and the reason
for each routing step.

Dale


_______________________________________________
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: "T Satyanarayana-A12694" <satyanarayana.t@motorola.com>
To: "Karunesh Sharma" <Karunesh.Sharma@globallogic.com>, "Dale Worley" <dworley@nortel.com>
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Subject: Re: [Sip] SIP Dialog Match
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No, privacy using encryption.

-----Original Message-----
From: Karunesh Sharma [mailto:Karunesh.Sharma@globallogic.com]=20
Sent: Monday, August 03, 2009 10:05 PM
To: T Satyanarayana-A12694; Dale Worley
Cc: sip@ietf.org
Subject: RE: [Sip] SIP Dialog Match

Are you talking about *privacy* headers? I fail to understand here how
*privacy* header will help in such cases.

K$
_______________________


-----Original Message-----
From: T Satyanarayana-A12694 [mailto:satyanarayana.t@motorola.com]
Sent: Monday, August 03, 2009 10:02 PM
To: Dale Worley; Karunesh Sharma
Cc: sip@ietf.org
Subject: RE: [Sip] SIP Dialog Match


In short, if some one wants to fake another client, it is possible since
the communication is in clear. If there are security concerns, then
security mechanisms (privacy etc) should be used.

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Dale Worley
Sent: Monday, August 03, 2009 9:35 PM
To: Karunesh Sharma
Cc: sip@ietf.org
Subject: Re: [Sip] SIP Dialog Match

On Mon, 2009-08-03 at 21:28 +0530, Karunesh Sharma wrote:
> Is this a valid behavior for scenario like 'Call-Forwarding'?

In principle, the responses to a request should almost always come from
the address to which the request was sent.  But that is not usually
enforced, and there may be good reasons why it isn't happening in a
particular instance.

In regard to an observed instance of this behavior, you must first
determine the complete routing of the request in question and the reason
for each routing step.

Dale


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References: <5F063FC3F0C8F14E945465A68E63C43E9451DB@ex3-del1.synapse.com><abcc59880908010647h1a4ae3f6m3f5feb4cfb71e1ac@mail.gmail.com><1249249277.4106.1.camel@victoria-pingtel-com.us.nortel.com><5F063FC3F0C8F14E945465A68E63C43E945346@ex3-del1.synapse.com><1249313088.3791.0.camel@victoria-pingtel-com.us.nortel.com><5F063FC3F0C8F14E945465A68E63C43E945347@ex3-del1.synapse.com><1249314398.3791.5.camel@victoria-pingtel-com.us.nortel.com> <5F063FC3F0C8F14E945465A68E63C43E94534A@ex3-del1.synapse.com>
From: "T Satyanarayana-A12694" <satyanarayana.t@motorola.com>
To: "Karunesh Sharma" <Karunesh.Sharma@globallogic.com>, "Dale Worley" <dworley@nortel.com>
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No, this isn't how it usually works. For call forwarding, there is
usually a proxy (or a softswitch) which does it on behalf of B.

However, if B chooses to do it himself: C (forwardee) gets it like a
regular call from B (forwarding agent) and C doesn't reply to A - it
replies to B. B then can shuffle the calls so that the call is directly
between A and C (through a set of target refreshes). However, this is
just hypothetical and not usually done in reality.

Regards
Satya T

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Karunesh Sharma
Sent: Monday, August 03, 2009 9:29 PM
To: Dale Worley
Cc: sip@ietf.org
Subject: Re: [Sip] SIP Dialog Match

Is this a valid behavior for scenario like 'Call-Forwarding'?

K$
_______________________


-----Original Message-----
From: Karunesh Sharma
Sent: Monday, August 03, 2009 9:22 PM
To: 'Dale Worley'
Cc: aayush bhatnagar; sip@ietf.org
Subject: RE: [Sip] SIP Dialog Match

I am trying to get that info how 'C' has this dialog identifiers but I
was wondering this as a major security issue for any SIP network.

K$
_______________________


-----Original Message-----
From: Dale Worley [mailto:dworley@nortel.com]
Sent: Monday, August 03, 2009 9:17 PM
To: Karunesh Sharma
Cc: aayush bhatnagar; sip@ietf.org
Subject: RE: [Sip] SIP Dialog Match

On Mon, 2009-08-03 at 21:03 +0530, Karunesh Sharma wrote:
> 172.16.31.30 (read as 'A') sent INVITE to 10.0.4.6 (read as 'B'). But=20
> somehow 192.168.4.67 (read as 'C') responds. Now 'A' knows nothing
about
> 'C', but still based on SIP dialog identifiers 'A' accepts that.
>=20
> So is there any way to challenge or deny such rouge responses?

Have you inquired as to how C knows of the dialog identifiers, and why
it is sending the responses?

Dale


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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-08.txt
	Pages           : 26
	Date            : 2009-08-04

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 not possible
to make one header have the characteristics of both interfaces 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.

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From ben.bonnaerens@alcatel-lucent.be  Wed Aug  5 07:25:06 2009
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Hi,
version -08 of record-route fix has just been submitted.

Change log:
- Changed its status from BCP to PS,
- Take into account IESG reviews and subsequent comments on the mailing
lists,=20
they were summarized in Robert Sparks's document (rrf.txt),=20
so it has been integrated in version -08 with minor refactoring from the
authors (me and Ben).

The thoughest part was related to the SIP/SIPS question raised by Cullen
(http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-route-fix/c
omment/98624/),
so here is the text that was added in version -08 to answer this
question:
"Thus, if the transport protocol changed between its
   incoming and outgoing sides, the proxy SHOULD use the double
   Record-Route technique and SHOULD add a transport parameter to each
   of the Record-Route URIs it inserts. With the exception that if TLS
is
   used as the transport protocol on either side of the proxy, the URI
   chosen to place in the Record-Route header field value reflecting the
   interface using TLS will need to leverage the [RFC3263] mechanisms to
   indicate that TLS must be used rather attempting to use the
deprecated
   "transport=3Dtls" URI parameter. See [RFC3261] Section 26.2.2 and
   [I-D.ietf-sip-sips] Section 3.1.4 for more discussion."

Brds,
Thomas & Ben

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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           : Connection Reuse in the Session Initiation Protocol (SIP)
	Author(s)       : V. Gurbani, et al.
	Filename        : draft-ietf-sip-connect-reuse-14.txt
	Pages           : 18
	Date            : 2009-08-05

This document enables a pair of communicating proxies to reuse a
congestion-controlled connection between themselves for sending
requests in the forward and backwards direction.  Because the
connection is essentially aliased for requests going in the backwards
direction, reuse is predicated upon both the communicating endpoints
authenticating themselves using X.509 certificates through TLS.  For
this reason, we only consider connection reuse for TLS over TCP and
TLS over SCTP.  This document also provides guidelines on connection
reuse and virtual SIP servers and the interaction of connection reuse
and DNS SRV lookups in SIP.

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Date: Wed, 05 Aug 2009 14:46:04 -0500
From: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
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Subject: [Sip] New version of connect-reuse issued, post IESG review
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All: I have released a new version of connect-reuse that
addresses the comments from IESG review.

The new version (-14) can be accessed from:
http://www.ietf.org/id/draft-ietf-sip-connect-reuse-14.txt

Here is the list of changes resulting from the IESG review:

- Changed ephemeral port numbers to be between 49152 and 65535,
   as suggested by IANA (comment from Gen-ART review.)
- In S8.1 modified text to be more compatible with domain-certs.
- In S8.1, the normative behavior in the first and third paragraphs
   seem to use "SHOULD" and "MUST" to describe the same behavior.
   Changed the MUST in the third paragraph to a SHOULD without
   any semantic loss.
- In S8.2, changed some text that makes it mandatory to follow
   the authentication steps in S9.2.  This change does not alter
   anything semantically.
- In S9.1 and 9.2, updated section numbers pointing to RFC YYYY.
- In S9.2, added 2 paragraphs on the impact of virtual servers
   and connection reuse on P1.

Thank you,

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

From john.elwell@siemens-enterprise.com  Thu Aug  6 00:36:55 2009
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Date: Thu, 06 Aug 2009 08:36:54 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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Well, I know what this means, but will your average reader understand
it? I think we are trying to say something along the lines that if the
transport protocol changes you SHOULD use the double Record-Route
technique, where the differences in transport protocol are reflected in
different values of the transport parameter (udp/tcp/sctp) and/or
different values of the URI scheme (sip/sips). Correct?

John

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of BONNAERENS Ben
> Sent: 05 August 2009 15:23
> To: sip@ietf.org
> Subject: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>=20
> Hi,
> version -08 of record-route fix has just been submitted.
>=20
> Change log:
> - Changed its status from BCP to PS,
> - Take into account IESG reviews and subsequent comments on=20
> the mailing
> lists,=20
> they were summarized in Robert Sparks's document (rrf.txt),=20
> so it has been integrated in version -08 with minor=20
> refactoring from the
> authors (me and Ben).
>=20
> The thoughest part was related to the SIP/SIPS question=20
> raised by Cullen
> (http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-r
> oute-fix/c
> omment/98624/),
> so here is the text that was added in version -08 to answer this
> question:
> "Thus, if the transport protocol changed between its
>    incoming and outgoing sides, the proxy SHOULD use the double
>    Record-Route technique and SHOULD add a transport parameter to each
>    of the Record-Route URIs it inserts. With the exception that if TLS
> is
>    used as the transport protocol on either side of the proxy, the URI
>    chosen to place in the Record-Route header field value=20
> reflecting the
>    interface using TLS will need to leverage the [RFC3263]=20
> mechanisms to
>    indicate that TLS must be used rather attempting to use the
> deprecated
>    "transport=3Dtls" URI parameter. See [RFC3261] Section 26.2.2 and
>    [I-D.ietf-sip-sips] Section 3.1.4 for more discussion."
>=20
> Brds,
> Thomas & Ben
> _______________________________________________
> 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
>=20

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From: "Karunesh Sharma" <Karunesh.Sharma@globallogic.com>
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Dale:

It's simple scenario where 'A' sent INVITE to 'B'. 'C' is man in middle
captured the INVITE and sent the response back to 'A'. I wonder why no
one ever had such case ever and why there is no established guidelines.

K$
_______________________

-----Original Message-----
From: Dale Worley [mailto:dworley@nortel.com]=20
Sent: Monday, August 03, 2009 9:35 PM
To: Karunesh Sharma
Cc: aayush bhatnagar; sip@ietf.org
Subject: RE: [Sip] SIP Dialog Match

On Mon, 2009-08-03 at 21:28 +0530, Karunesh Sharma wrote:
> Is this a valid behavior for scenario like 'Call-Forwarding'?

In principle, the responses to a request should almost always come from
the address to which the request was sent.  But that is not usually
enforced, and there may be good reasons why it isn't happening in a
particular instance.

In regard to an observed instance of this behavior, you must first
determine the complete routing of the request in question and the reason
for each routing step.

Dale



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From: "T Satyanarayana-A12694" <satyanarayana.t@motorola.com>
To: "Karunesh Sharma" <Karunesh.Sharma@globallogic.com>, "Dale Worley" <dworley@nortel.com>
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Well, strictly speaking, this is possible and most SIP networks are
vulnerable to this. This is no different from HTTP or SMTP where this is
possible.

The guidelines (there could be others as well) to overcome this are:

1. Implement a known-good-source kind of logic where you accept
responses coming only from a well known proxy. Ofcourse, this works only
when there is a proxy and when the source IP address of the response
itself is not modified.

2. Implement encryption (e.g. TLS) for signaling

Regards
Satya T

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Karunesh Sharma
Sent: Friday, August 07, 2009 6:37 PM
To: Dale Worley
Cc: sip@ietf.org
Subject: Re: [Sip] SIP Dialog Match

Dale:

It's simple scenario where 'A' sent INVITE to 'B'. 'C' is man in middle
captured the INVITE and sent the response back to 'A'. I wonder why no
one ever had such case ever and why there is no established guidelines.

K$
_______________________

-----Original Message-----
From: Dale Worley [mailto:dworley@nortel.com]
Sent: Monday, August 03, 2009 9:35 PM
To: Karunesh Sharma
Cc: aayush bhatnagar; sip@ietf.org
Subject: RE: [Sip] SIP Dialog Match

On Mon, 2009-08-03 at 21:28 +0530, Karunesh Sharma wrote:
> Is this a valid behavior for scenario like 'Call-Forwarding'?

In principle, the responses to a request should almost always come from
the address to which the request was sent.  But that is not usually
enforced, and there may be good reasons why it isn't happening in a
particular instance.

In regard to an observed instance of this behavior, you must first
determine the complete routing of the request in question and the reason
for each routing step.

Dale


_______________________________________________
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: sandip gangakhedkar <sandipfloyd@gmail.com>
Date: Thu, 6 Aug 2009 20:15:51 +0530
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Subject: [Sip] Contact header in 200 OK
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Hi all,

I'd like to know if the Contact header is mandatory for a 200 OK response to
an INVITE request.

The RFC doesnt explicitly mention anything in this regard although a 200 OK
is mandatory for a REGISTER request.

If the originating endpoint receives the Contact header in the 180 Ringing
response (which I assume is mandatory), then it knows the location of the
remote endpoint, in which case the Contact header would be redundant in the
200 OK response.

Waiting for some expert intervention.. :)

Regards,
-- 
Sandip Gangakhedkar
Mobile:+91-97396-99916
Office: +91-80-4363-3899
Profile: http://www.linkedin.com/in/sandipg

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Hi all,<br><br>I&#39;d like to know if the Contact header is mandatory for a 200 OK response to an INVITE request. <br><br>The RFC doesnt explicitly mention anything in this regard although a 200 OK is mandatory for a REGISTER request.<br>

<br>If the originating endpoint receives the Contact header in the 180 Ringing response (which I assume is mandatory), then it knows the location of the remote endpoint, in which case the Contact header would be redundant in the 200 OK response.<br>

<br>Waiting for some expert intervention.. :) <br><br>Regards,<br>-- <br>Sandip Gangakhedkar<br>Mobile:+91-97396-99916<br>Office: +91-80-4363-3899<br>Profile: <a href="http://www.linkedin.com/in/sandipg">http://www.linkedin.com/in/sandipg</a><br>



--0015174bdc4e40b8e504707a2e92--

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The RFC does mention this. In table 2, on page 162.
=20
      Header field          where   proxy ACK BYE CAN INV OPT REG
      ___________________________________________________________
      Contact                2xx           -   -   -   m   o   o

regards
Satya T

=20
________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
sandip gangakhedkar
Sent: Thursday, August 06, 2009 8:16 PM
To: sip@ietf.org
Subject: [Sip] Contact header in 200 OK


Hi all,

I'd like to know if the Contact header is mandatory for a 200 OK
response to an INVITE request.=20

The RFC doesnt explicitly mention anything in this regard although a 200
OK is mandatory for a REGISTER request.

If the originating endpoint receives the Contact header in the 180
Ringing response (which I assume is mandatory), then it knows the
location of the remote endpoint, in which case the Contact header would
be redundant in the 200 OK response.

Waiting for some expert intervention.. :)=20

Regards,
--=20
Sandip Gangakhedkar
Mobile:+91-97396-99916
Office: +91-80-4363-3899
Profile: http://www.linkedin.com/in/sandipg


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<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3527" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D388204613-07082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>The RFC does mention this. In table 2, on page=20
162.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D388204613-07082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D388204613-07082009><FONT=20
face=3D"Courier New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Header=20
field&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
where&nbsp;&nbsp;=20
proxy ACK BYE CAN INV OPT REG<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
___________________________________________________________<BR>&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;=20
Contact&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;=20
2xx&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-&nbsp;&nbsp;=20
-&nbsp;&nbsp; -&nbsp;&nbsp; m&nbsp;&nbsp; o&nbsp;&nbsp;=20
o<BR></FONT></SPAN></DIV>
<DIV><SPAN class=3D388204613-07082009></SPAN><FONT face=3DArial><FONT=20
color=3D#0000ff><FONT size=3D2>regards<BR><SPAN=20
class=3D388204613-07082009></SPAN>Satya&nbsp;T</FONT></FONT></FONT></DIV>=

<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D388204613-07082009></SPAN></FONT></FONT></FONT><BR>&nbsp;</DIV>
<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>sandip=20
gangakhedkar<BR><B>Sent:</B> Thursday, August 06, 2009 8:16 =
PM<BR><B>To:</B>=20
sip@ietf.org<BR><B>Subject:</B> [Sip] Contact header in 200=20
OK<BR></FONT><BR></DIV>
<DIV></DIV>Hi all,<BR><BR>I'd like to know if the Contact header is =
mandatory=20
for a 200 OK response to an INVITE request. <BR><BR>The RFC doesnt =
explicitly=20
mention anything in this regard although a 200 OK is mandatory for a =
REGISTER=20
request.<BR><BR>If the originating endpoint receives the Contact header =
in the=20
180 Ringing response (which I assume is mandatory), then it knows the =
location=20
of the remote endpoint, in which case the Contact header would be =
redundant in=20
the 200 OK response.<BR><BR>Waiting for some expert intervention.. :)=20
<BR><BR>Regards,<BR>-- <BR>Sandip=20
Gangakhedkar<BR>Mobile:+91-97396-99916<BR>Office: =
+91-80-4363-3899<BR>Profile:=20
<A=20
href=3D"http://www.linkedin.com/in/sandipg">http://www.linkedin.com/in/sa=
ndipg</A><BR></BODY></HTML>

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From ben.bonnaerens@alcatel-lucent.be  Fri Aug  7 06:57:45 2009
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From: "BONNAERENS Ben" <ben.bonnaerens@alcatel-lucent.be>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, <sip@ietf.org>
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Hello John,

> Well, I know what this means, but will your average reader=20
> understand it? I think we are trying to say something along=20
> the lines that if the transport protocol changes you SHOULD=20
> use the double Record-Route technique, where the differences=20
> in transport protocol are reflected in different values of=20
> the transport parameter (udp/tcp/sctp) and/or different=20
> values of the URI scheme (sip/sips). Correct?

Your phrasing could lead to the understanding that we suggest that=20
only a sips URI can be used to identify the TLS interface which is not
correct.
As the current text suggests, can the TLS interface be identified by any
URI
(sip or sips ; excluding the deprecated transport=3Dtls parameter) that=20
resolves via RFC3263 procedures to the TLS transport.

IMHO leaves your suggested phrasing more room for ambiguity
compared to the current text.

Best regards & thanks,

Ben. =20

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
> Sent: donderdag 6 augustus 2009 9:37
> To: BONNAERENS Ben; sip@ietf.org
> Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>=20
> Well, I know what this means, but will your average reader=20
> understand it? I think we are trying to say something along=20
> the lines that if the transport protocol changes you SHOULD=20
> use the double Record-Route technique, where the differences=20
> in transport protocol are reflected in different values of=20
> the transport parameter (udp/tcp/sctp) and/or different=20
> values of the URI scheme (sip/sips). Correct?
>=20
> John
>=20
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of=20
> > BONNAERENS Ben
> > Sent: 05 August 2009 15:23
> > To: sip@ietf.org
> > Subject: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> >=20
> > Hi,
> > version -08 of record-route fix has just been submitted.
> >=20
> > Change log:
> > - Changed its status from BCP to PS,
> > - Take into account IESG reviews and subsequent comments on the=20
> > mailing lists, they were summarized in Robert Sparks's document=20
> > (rrf.txt), so it has been integrated in version -08 with minor=20
> > refactoring from the authors (me and Ben).
> >=20
> > The thoughest part was related to the SIP/SIPS question raised by=20
> > Cullen=20
> (http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-r
> > oute-fix/c
> > omment/98624/),
> > so here is the text that was added in version -08 to answer this
> > question:
> > "Thus, if the transport protocol changed between its
> >    incoming and outgoing sides, the proxy SHOULD use the double
> >    Record-Route technique and SHOULD add a transport=20
> parameter to each
> >    of the Record-Route URIs it inserts. With the exception=20
> that if TLS=20
> > is
> >    used as the transport protocol on either side of the=20
> proxy, the URI
> >    chosen to place in the Record-Route header field value=20
> reflecting=20
> > the
> >    interface using TLS will need to leverage the [RFC3263]=20
> mechanisms=20
> > to
> >    indicate that TLS must be used rather attempting to use the=20
> > deprecated
> >    "transport=3Dtls" URI parameter. See [RFC3261] Section 26.2.2 and
> >    [I-D.ietf-sip-sips] Section 3.1.4 for more discussion."
> >=20
> > Brds,
> > Thomas & Ben
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20

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From: Yoav Gaziel <ygaziel@gmail.com>
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Subject: [Sip] Usage of the 'Route' header.
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Hi,

Regarding the usage of the Route header. Must this header also appear in the
RESPONSE to a request in which this header appeared, or we can rely on the
'Via' header for routing the response back to the sender?

I've read the relevant RFC's and couldn't find an explicit reference for
this matter. If someone can point me to the exact clause and RFC I'll be
grateful. Also, an explicit authoritative answer would be much appreciated.

(Note that i'm talking about the 'Route' field, and not the 'Record-Route'
field for which there are alot of references in the RFCs)

Thanks,
Yoav.

--0015174bedcecd94860470afaf70
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<div dir=3D"ltr"><div>Hi,</div>
<div>=A0</div>
<div>Regarding the usage of the Route header.=A0Must this header also appea=
r in the RESPONSE to a request in which this header appeared, or we can rel=
y on the &#39;Via&#39; header for routing the response back to the sender?<=
/div>

<div>=A0</div>
<div>I&#39;ve read the relevant RFC&#39;s and couldn&#39;t find an=A0explic=
it=A0reference for this matter. If someone can point me to the exact clause=
 and RFC I&#39;ll be grateful. Also, an explicit authoritative answer would=
 be much=A0appreciated.</div>

<div>=A0</div>
<div>(Note that i&#39;m talking about the &#39;Route&#39; field, and not th=
e &#39;Record-Route&#39; field for which there are alot of references in th=
e RFCs)</div>
<div>=A0</div>
<div>Thanks,</div>
<div>Yoav.</div></div>

--0015174bedcecd94860470afaf70--

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Hi,
=20
You always rely on the Via for routing responses back to the sender. You
never rely on the Route header for routing responses.
=20
Regards,
=20
Christer

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Yoav Gaziel
Sent: Sunday, August 09, 2009 9:36 AM
To: sip@ietf.org
Subject: [Sip] Usage of the 'Route' header.


Hi,
=20
Regarding the usage of the Route header. Must this header also appear in
the RESPONSE to a request in which this header appeared, or we can rely
on the 'Via' header for routing the response back to the sender?
=20
I've read the relevant RFC's and couldn't find an explicit reference for
this matter. If someone can point me to the exact clause and RFC I'll be
grateful. Also, an explicit authoritative answer would be much
appreciated.
=20
(Note that i'm talking about the 'Route' field, and not the
'Record-Route' field for which there are alot of references in the RFCs)
=20
Thanks,
Yoav.

------_=_NextPart_001_01CA18E1.A5BAEB2E
Content-Type: text/html;
	charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.3790.4555" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D832060711-09082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Hi,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D832060711-09082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D832060711-09082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>You always rely on the Via for routing =
responses back to=20
the sender. You never rely on the Route header for routing=20
responses.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D832060711-09082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D832060711-09082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D832060711-09082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D832060711-09082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></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>Yoav =
Gaziel<BR><B>Sent:</B>=20
Sunday, August 09, 2009 9:36 AM<BR><B>To:</B> =
sip@ietf.org<BR><B>Subject:</B>=20
[Sip] Usage of the 'Route' header.<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=3Dltr>
<DIV>Hi,</DIV>
<DIV>&nbsp;</DIV>
<DIV>Regarding the usage of the Route header.&nbsp;Must this header also =
appear=20
in the RESPONSE to a request in which this header appeared, or we can =
rely on=20
the 'Via' header for routing the response back to the sender?</DIV>
<DIV>&nbsp;</DIV>
<DIV>I've read the relevant RFC's and couldn't find=20
an&nbsp;explicit&nbsp;reference for this matter. If someone can point me =
to the=20
exact clause and RFC I'll be grateful. Also, an explicit authoritative =
answer=20
would be much&nbsp;appreciated.</DIV>
<DIV>&nbsp;</DIV>
<DIV>(Note that i'm talking about the 'Route' field, and not the =
'Record-Route'=20
field for which there are alot of references in the RFCs)</DIV>
<DIV>&nbsp;</DIV>
<DIV>Thanks,</DIV>
<DIV>Yoav.</DIV></DIV></BODY></HTML>

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Date: Sun, 9 Aug 2009 22:12:56 +0200
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From: Tomasz Zieleniewski <tzieleniewski@gmail.com>
To: Yoav Gaziel <ygaziel@gmail.com>
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Subject: Re: [Sip] Usage of the 'Route' header.
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Hi Yoav,

Route header is used only for request routuing,
responses have different procedures which utilize the Via header values.
BTW some time ago I wrote a short post about Route header, might give u some
more info.
http://ictbackyard.com/?p=10

Kind regards,
- Tomasz Zieleniewski

2009/8/9 Yoav Gaziel <ygaziel@gmail.com>

> Hi,
>
> Regarding the usage of the Route header. Must this header also appear in
> the RESPONSE to a request in which this header appeared, or we can rely on
> the 'Via' header for routing the response back to the sender?
>
> I've read the relevant RFC's and couldn't find an explicit reference for
> this matter. If someone can point me to the exact clause and RFC I'll be
> grateful. Also, an explicit authoritative answer would be much appreciated.
>
> (Note that i'm talking about the 'Route' field, and not the 'Record-Route'
> field for which there are alot of references in the RFCs)
>
> Thanks,
> Yoav.
>
> _______________________________________________
> 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
>

--001636c5a9bf587afb0470bb1855
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Hi Yoav,<br><br>Route header is used only for request routuing, <br>respons=
es have different procedures which utilize the Via header values.<br>BTW so=
me time ago I wrote a short post about Route header, might give u some more=
 info.<br>
<a href=3D"http://ictbackyard.com/?p=3D10">http://ictbackyard.com/?p=3D10</=
a><br><br>Kind regards,<br>- Tomasz Zieleniewski<br><br><div class=3D"gmail=
_quote">2009/8/9 Yoav Gaziel <span dir=3D"ltr">&lt;<a href=3D"mailto:ygazie=
l@gmail.com">ygaziel@gmail.com</a>&gt;</span><br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div dir=3D"ltr">=
<div>Hi,</div>
<div>=A0</div>
<div>Regarding the usage of the Route header.=A0Must this header also appea=
r in the RESPONSE to a request in which this header appeared, or we can rel=
y on the &#39;Via&#39; header for routing the response back to the sender?<=
/div>


<div>=A0</div>
<div>I&#39;ve read the relevant RFC&#39;s and couldn&#39;t find an=A0explic=
it=A0reference for this matter. If someone can point me to the exact clause=
 and RFC I&#39;ll be grateful. Also, an explicit authoritative answer would=
 be much=A0appreciated.</div>


<div>=A0</div>
<div>(Note that i&#39;m talking about the &#39;Route&#39; field, and not th=
e &#39;Record-Route&#39; field for which there are alot of references in th=
e RFCs)</div>
<div>=A0</div>
<div>Thanks,</div>
<div>Yoav.</div></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>

--001636c5a9bf587afb0470bb1855--

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From: "Goyal, Devki" <dgoyal@tnsi.com>
To: sandip gangakhedkar <sandipfloyd@gmail.com>, "sip@ietf.org" <sip@ietf.org>
Date: Fri, 7 Aug 2009 10:36:04 -0400
Thread-Topic: [Sip] Contact header in 200 OK
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Hello Sandip
Why do you say that RFC doesn't talk about Contact for 200 OK response of I=
NVITE?
It says that Contact header is mandatory for INVITE response only.


Header field          where   proxy ACK BYE CAN INV OPT REG

Contact                2xx           -   -   -   m   o   o



For OPTIONS, REGISTER it is optional and for rest of the messages (BYE, CAN=
CEL ) it is forbidden.



Regards

Devki


________________________________
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of sandi=
p gangakhedkar
Sent: Thursday, August 06, 2009 10:46 AM
To: sip@ietf.org
Subject: [Sip] Contact header in 200 OK

Hi all,

I'd like to know if the Contact header is mandatory for a 200 OK response t=
o an INVITE request.

The RFC doesnt explicitly mention anything in this regard although a 200 OK=
 is mandatory for a REGISTER request.

If the originating endpoint receives the Contact header in the 180 Ringing =
response (which I assume is mandatory), then it knows the location of the r=
emote endpoint, in which case the Contact header would be redundant in the =
200 OK response.

Waiting for some expert intervention.. :)

Regards,
--
Sandip Gangakhedkar
Mobile:+91-97396-99916
Office: +91-80-4363-3899
Profile: http://www.linkedin.com/in/sandipg

________________________________
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<div class=3D"Section1">
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Hello Sandip<o:p></o:p></span></font><=
/p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Why do you say that RFC doesn&#8217;t =
talk about Contact for 200 OK response of INVITE?<o:p></o:p></span></font><=
/p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">It says that Contact header is mandato=
ry for INVITE response only.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<pre style=3D"line-height:19.8pt;background:white"><font size=3D"2" color=
=3D"black" face=3D"Courier New"><span style=3D"font-size:10.0pt">Header fie=
ld&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; where&nbsp;&nbsp; =
proxy ACK BYE CAN INV OPT REG<o:p></o:p></span></font></pre>
<pre style=3D"line-height:19.8pt;background:white"><font size=3D"2" color=
=3D"black" face=3D"Courier New"><span style=3D"font-size:10.0pt">Contact&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp; 2xx&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; -&nbsp;&nbsp; -&nbsp;&nbsp; -&nbsp;&nbsp; m&nbsp;&nbsp; o&nbsp;&nbsp; o<o=
:p></o:p></span></font></pre>
<pre style=3D"line-height:19.8pt;background:white"><font size=3D"2" color=
=3D"black" face=3D"Courier New"><span style=3D"font-size:10.0pt"><o:p>&nbsp=
;</o:p></span></font></pre>
<pre style=3D"line-height:19.8pt;background:white"><font size=3D"2" color=
=3D"black" face=3D"Courier New"><span style=3D"font-size:10.0pt">For OPTION=
S, REGISTER it is optional and for rest of the messages (BYE, CANCEL ) it i=
s forbidden.<o:p></o:p></span></font></pre>
<pre style=3D"line-height:19.8pt;background:white"><font size=3D"2" color=
=3D"black" face=3D"Courier New"><span style=3D"font-size:10.0pt"><o:p>&nbsp=
;</o:p></span></font></pre>
<pre style=3D"line-height:19.8pt;background:white"><font size=3D"2" color=
=3D"black" face=3D"Courier New"><span style=3D"font-size:10.0pt">Regards<o:=
p></o:p></span></font></pre>
<pre style=3D"line-height:19.8pt;background:white"><font size=3D"2" color=
=3D"black" face=3D"Courier New"><span style=3D"font-size:10.0pt">Devki <o:p=
></o:p></span></font></pre>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
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<div>
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<hr size=3D"2" width=3D"100%" align=3D"center" tabindex=3D"-1">
</span></font></div>
<p class=3D"MsoNormal"><b><font size=3D"2" face=3D"Tahoma"><span style=3D"f=
ont-size:10.0pt;
font-family:Tahoma;font-weight:bold">From:</span></font></b><font size=3D"2=
" face=3D"Tahoma"><span style=3D"font-size:10.0pt;font-family:Tahoma"> sip-=
bounces@ietf.org [mailto:sip-bounces@ietf.org]
<b><span style=3D"font-weight:bold">On Behalf Of </span></b>sandip gangakhe=
dkar<br>
<b><span style=3D"font-weight:bold">Sent:</span></b> Thursday, August 06, 2=
009 10:46 AM<br>
<b><span style=3D"font-weight:bold">To:</span></b> sip@ietf.org<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> [Sip] Contact heade=
r in 200 OK</span></font><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">Hi all,<br>
<br>
I'd like to know if the Contact header is mandatory for a 200 OK response t=
o an INVITE request.
<br>
<br>
The RFC doesnt explicitly mention anything in this regard although a 200 OK=
 is mandatory for a REGISTER request.<br>
<br>
If the originating endpoint receives the Contact header in the 180 Ringing =
response (which I assume is mandatory), then it knows the location of the r=
emote endpoint, in which case the Contact header would be redundant in the =
200 OK response.<br>
<br>
Waiting for some expert intervention.. :) <br>
<br>
Regards,<br>
-- <br>
Sandip Gangakhedkar<br>
<st1:City w:st=3D"on"><st1:place w:st=3D"on">Mobile</st1:place></st1:City>:=
&#43;91-97396-99916<br>
Office: &#43;91-80-4363-3899<br>
Profile: <a href=3D"http://www.linkedin.com/in/sandipg">http://www.linkedin=
.com/in/sandipg</a><o:p></o:p></span></font></p>
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From sandipfloyd@gmail.com  Fri Aug  7 06:51:05 2009
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From: sandip gangakhedkar <sandipfloyd@gmail.com>
Date: Fri, 7 Aug 2009 19:20:46 +0530
Message-ID: <6208da1a0908070650y28ce2296u88dba1d101c59ef7@mail.gmail.com>
To: T Satyanarayana-A12694 <satyanarayana.t@motorola.com>, sip@ietf.org
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X-Mailman-Approved-At: Mon, 10 Aug 2009 04:54:17 -0700
Subject: Re: [Sip] Contact header in 200 OK
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--000e0cd4d51e1acad204708d870e
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Hi Satya,

Thanks for the reply..didn't locate that table at first :)

Regards,

Sandip

On Fri, Aug 7, 2009 at 7:17 PM, T Satyanarayana-A12694 <
satyanarayana.t@motorola.com> wrote:

>  The RFC does mention this. In table 2, on page 162.
>
>       Header field          where   proxy ACK BYE CAN INV OPT REG
>       ___________________________________________________________
>       Contact                2xx           -   -   -   m   o   o
> regards
> Satya T
>
>
>  ------------------------------
> *From:* sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] *On Behalf Of *sandip
> gangakhedkar
> *Sent:* Thursday, August 06, 2009 8:16 PM
> *To:* sip@ietf.org
> *Subject:* [Sip] Contact header in 200 OK
>
> Hi all,
>
> I'd like to know if the Contact header is mandatory for a 200 OK response
> to an INVITE request.
>
> The RFC doesnt explicitly mention anything in this regard although a 200 OK
> is mandatory for a REGISTER request.
>
> If the originating endpoint receives the Contact header in the 180 Ringing
> response (which I assume is mandatory), then it knows the location of the
> remote endpoint, in which case the Contact header would be redundant in the
> 200 OK response.
>
> Waiting for some expert intervention.. :)
>
> Regards,
> --
> Sandip Gangakhedkar
> Mobile:+91-97396-99916
> Office: +91-80-4363-3899
> Profile: http://www.linkedin.com/in/sandipg
>

--000e0cd4d51e1acad204708d870e
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Satya,<br><br>Thanks for the reply..didn&#39;t locate that table at firs=
t :)<br><br>Regards,<br><br>Sandip<br><br><div class=3D"gmail_quote">On Fri=
, Aug 7, 2009 at 7:17 PM, T Satyanarayana-A12694 <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:satyanarayana.t@motorola.com">satyanarayana.t@motorola.com</a=
>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">



<div>
<div dir=3D"ltr" align=3D"left"><span><font color=3D"#0000ff" size=3D"2" fa=
ce=3D"Arial">The RFC does mention this. In table 2, on page=20
162.</font></span></div>
<div dir=3D"ltr" align=3D"left"><span><font color=3D"#0000ff" size=3D"2" fa=
ce=3D"Arial"></font></span>=A0</div>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Courier New">=A0=A0=A0=
=A0=A0 Header=20
field=A0=A0=A0=A0=A0=A0=A0=A0=A0 where=A0=A0=20
proxy ACK BYE CAN INV OPT REG<br>=A0=A0=A0=A0=A0=20
___________________________________________________________<br>=A0=A0=A0=A0=
=A0=20
Contact=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=20
2xx=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 -=A0=A0=20
-=A0=A0 -=A0=A0 m=A0=A0 o=A0=A0=20
o<br></font></span></div>
<div><span></span><font face=3D"Arial"><font color=3D"#0000ff"><font size=
=3D"2">regards<br><span></span>Satya=A0T</font></font></font></div>
<div><font face=3D"Arial"><font color=3D"#0000ff"><font size=3D"2"><span></=
span></font></font></font><br>=A0</div>
<div dir=3D"ltr" align=3D"left" lang=3D"en-us">
<hr>
<font size=3D"2" face=3D"Tahoma"><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>sandip=20
gangakhedkar<br><b>Sent:</b> Thursday, August 06, 2009 8:16 PM<br><b>To:</b=
>=20
<a href=3D"mailto:sip@ietf.org" target=3D"_blank">sip@ietf.org</a><br><b>Su=
bject:</b> [Sip] Contact header in 200=20
OK<br></font><br></div><div><div></div><div class=3D"h5">
<div></div>Hi all,<br><br>I&#39;d like to know if the Contact header is man=
datory=20
for a 200 OK response to an INVITE request. <br><br>The RFC doesnt explicit=
ly=20
mention anything in this regard although a 200 OK is mandatory for a REGIST=
ER=20
request.<br><br>If the originating endpoint receives the Contact header in =
the=20
180 Ringing response (which I assume is mandatory), then it knows the locat=
ion=20
of the remote endpoint, in which case the Contact header would be redundant=
 in=20
the 200 OK response.<br><br>Waiting for some expert intervention.. :)=20
<br><br>Regards,<br>-- <br>Sandip=20
Gangakhedkar<br>Mobile:+91-97396-99916<br>Office: +91-80-4363-3899<br>Profi=
le:=20
<a href=3D"http://www.linkedin.com/in/sandipg" target=3D"_blank">http://www=
.linkedin.com/in/sandipg</a><br></div></div></div>
</blockquote></div><br><br>

--000e0cd4d51e1acad204708d870e--

From dean.willis@softarmor.com  Mon Aug 10 22:27:09 2009
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Please be advised of changes made as a result of IESG review:

-------- Original Message --------
Subject: Re: DISCUSS: draft-ietf-sip-record-route-fix
Date: Tue, 4 Aug 2009 23:04:35 +0200
From: thomas froment <tfroment@gmail.com>
To: iesg@ietf.org, sip-chairs@tools.ietf.org, 
draft-ietf-sip-record-route-fix@tools.ietf.org
CC: nils@ohlmeier.com
References: <985094.76974.qm@web56801.mail.re3.yahoo.com>	 
<AB4AE383BB184B41B4868050AEEBF50B@your029b8cecfe>	 
<4A5F59C7.5080500@softarmor.com>

Hi,
version -08 of record-route fix has just been submitted
(should be athttp://tools.ietf.org/html/draft-ietf-sip-record-route-fix 
soon)

Change log:
- Changed its status from BCP to PS,
- Take into account IESG reviews and subsequent comments on the
mailing lists, they were summarized in Robert Sparks's document
(rrf.txt), so it has been integrated in version -08 with minor
refactoring from the authors (me and Ben).

The thoughest part was related to the SIP/SIPS question raised by
Cullen 
(http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-route-fix/comment/98624/),
so here is the text that was added in version -08 to answer this
question:
"Thus, if the transport protocol changed between its
    incoming and outgoing sides, the proxy SHOULD use the double
    Record-Route technique and SHOULD add a transport parameter to each
    of the Record-Route URIs it inserts. With the exception that if TLS is
    used as the transport protocol on either side of the proxy, the URI
    chosen to place in the Record-Route header field value reflecting the
    interface using TLS will need to leverage the [RFC3263] mechanisms to
    indicate that TLS must be used rather attempting to use the deprecated
    "transport=tls" URI parameter. See [RFC3261] Section 26.2.2 and
    [I-D.ietf-sip-sips] Section 3.1.4 for more discussion."

Brds,
Thomas


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--------------020508030100090204010705--

From john.elwell@siemens-enterprise.com  Tue Aug 11 01:44:40 2009
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Date: Tue, 11 Aug 2009 09:43:57 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Thread-Topic: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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In that case, I obviously misunderstood the text in the current draft.
What exactly is the meaning of "will need to leverage the [RFC3263]
mechanisms"?

John

> -----Original Message-----
> From: BONNAERENS Ben [mailto:ben.bonnaerens@alcatel-lucent.be]=20
> Sent: 07 August 2009 14:57
> To: Elwell, John; sip@ietf.org
> Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>=20
> Hello John,
>=20
> > Well, I know what this means, but will your average reader=20
> > understand it? I think we are trying to say something along=20
> > the lines that if the transport protocol changes you SHOULD=20
> > use the double Record-Route technique, where the differences=20
> > in transport protocol are reflected in different values of=20
> > the transport parameter (udp/tcp/sctp) and/or different=20
> > values of the URI scheme (sip/sips). Correct?
>=20
> Your phrasing could lead to the understanding that we suggest that=20
> only a sips URI can be used to identify the TLS interface which is not
> correct.
> As the current text suggests, can the TLS interface be=20
> identified by any
> URI
> (sip or sips ; excluding the deprecated transport=3Dtls parameter) =
that=20
> resolves via RFC3263 procedures to the TLS transport.
>=20
> IMHO leaves your suggested phrasing more room for ambiguity
> compared to the current text.
>=20
> Best regards & thanks,
>=20
> Ben. =20
>=20
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
> > Sent: donderdag 6 augustus 2009 9:37
> > To: BONNAERENS Ben; sip@ietf.org
> > Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> >=20
> > Well, I know what this means, but will your average reader=20
> > understand it? I think we are trying to say something along=20
> > the lines that if the transport protocol changes you SHOULD=20
> > use the double Record-Route technique, where the differences=20
> > in transport protocol are reflected in different values of=20
> > the transport parameter (udp/tcp/sctp) and/or different=20
> > values of the URI scheme (sip/sips). Correct?
> >=20
> > John
> >=20
> > > -----Original Message-----
> > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> > Behalf Of=20
> > > BONNAERENS Ben
> > > Sent: 05 August 2009 15:23
> > > To: sip@ietf.org
> > > Subject: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> > >=20
> > > Hi,
> > > version -08 of record-route fix has just been submitted.
> > >=20
> > > Change log:
> > > - Changed its status from BCP to PS,
> > > - Take into account IESG reviews and subsequent comments on the=20
> > > mailing lists, they were summarized in Robert Sparks's document=20
> > > (rrf.txt), so it has been integrated in version -08 with minor=20
> > > refactoring from the authors (me and Ben).
> > >=20
> > > The thoughest part was related to the SIP/SIPS question raised by=20
> > > Cullen=20
> > (http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-r
> > > oute-fix/c
> > > omment/98624/),
> > > so here is the text that was added in version -08 to answer this
> > > question:
> > > "Thus, if the transport protocol changed between its
> > >    incoming and outgoing sides, the proxy SHOULD use the double
> > >    Record-Route technique and SHOULD add a transport=20
> > parameter to each
> > >    of the Record-Route URIs it inserts. With the exception=20
> > that if TLS=20
> > > is
> > >    used as the transport protocol on either side of the=20
> > proxy, the URI
> > >    chosen to place in the Record-Route header field value=20
> > reflecting=20
> > > the
> > >    interface using TLS will need to leverage the [RFC3263]=20
> > mechanisms=20
> > > to
> > >    indicate that TLS must be used rather attempting to use the=20
> > > deprecated
> > >    "transport=3Dtls" URI parameter. See [RFC3261] Section 26.2.2 =
and
> > >    [I-D.ietf-sip-sips] Section 3.1.4 for more discussion."
> > >=20
> > > Brds,
> > > Thomas & Ben
> > > _______________________________________________
> > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >=20
> >=20
>=20

From tzieleniewski@gmail.com  Tue Aug 11 04:01:38 2009
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Date: Tue, 11 Aug 2009 13:01:36 +0200
Message-ID: <5fd52d7a0908110401lc151128v891ec1289927a05d@mail.gmail.com>
From: Tomasz Zieleniewski <tzieleniewski@gmail.com>
To: BONNAERENS Ben <ben.bonnaerens@alcatel-lucent.be>
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Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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--001636c5bb7157420d0470dba0e6
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Hello Ben,

Can we say in other words that proxy in such case should
indicate used TLS transport by inserting the sips URI in the
Record-Route header instead of using transport=tls parameter.

Kind regards,
- Tomasz Zieleniewski

2009/8/7 BONNAERENS Ben <ben.bonnaerens@alcatel-lucent.be>

> Hello John,
>
> > Well, I know what this means, but will your average reader
> > understand it? I think we are trying to say something along
> > the lines that if the transport protocol changes you SHOULD
> > use the double Record-Route technique, where the differences
> > in transport protocol are reflected in different values of
> > the transport parameter (udp/tcp/sctp) and/or different
> > values of the URI scheme (sip/sips). Correct?
>
> Your phrasing could lead to the understanding that we suggest that
> only a sips URI can be used to identify the TLS interface which is not
> correct.
> As the current text suggests, can the TLS interface be identified by any
> URI
> (sip or sips ; excluding the deprecated transport=tls parameter) that
> resolves via RFC3263 procedures to the TLS transport.
>
> IMHO leaves your suggested phrasing more room for ambiguity
> compared to the current text.
>
> Best regards & thanks,
>
> Ben.
>
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> > Sent: donderdag 6 augustus 2009 9:37
> > To: BONNAERENS Ben; sip@ietf.org
> > Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> >
> > Well, I know what this means, but will your average reader
> > understand it? I think we are trying to say something along
> > the lines that if the transport protocol changes you SHOULD
> > use the double Record-Route technique, where the differences
> > in transport protocol are reflected in different values of
> > the transport parameter (udp/tcp/sctp) and/or different
> > values of the URI scheme (sip/sips). Correct?
> >
> > John
> >
> > > -----Original Message-----
> > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > Behalf Of
> > > BONNAERENS Ben
> > > Sent: 05 August 2009 15:23
> > > To: sip@ietf.org
> > > Subject: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> > >
> > > Hi,
> > > version -08 of record-route fix has just been submitted.
> > >
> > > Change log:
> > > - Changed its status from BCP to PS,
> > > - Take into account IESG reviews and subsequent comments on the
> > > mailing lists, they were summarized in Robert Sparks's document
> > > (rrf.txt), so it has been integrated in version -08 with minor
> > > refactoring from the authors (me and Ben).
> > >
> > > The thoughest part was related to the SIP/SIPS question raised by
> > > Cullen
> > (http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-r
> > > oute-fix/c
> > > omment/98624/),
> > > so here is the text that was added in version -08 to answer this
> > > question:
> > > "Thus, if the transport protocol changed between its
> > >    incoming and outgoing sides, the proxy SHOULD use the double
> > >    Record-Route technique and SHOULD add a transport
> > parameter to each
> > >    of the Record-Route URIs it inserts. With the exception
> > that if TLS
> > > is
> > >    used as the transport protocol on either side of the
> > proxy, the URI
> > >    chosen to place in the Record-Route header field value
> > reflecting
> > > the
> > >    interface using TLS will need to leverage the [RFC3263]
> > mechanisms
> > > to
> > >    indicate that TLS must be used rather attempting to use the
> > > deprecated
> > >    "transport=tls" URI parameter. See [RFC3261] Section 26.2.2 and
> > >    [I-D.ietf-sip-sips] Section 3.1.4 for more discussion."
> > >
> > > Brds,
> > > Thomas & Ben
> > > _______________________________________________
> > > 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
> > >
> >
> _______________________________________________
> 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
>

--001636c5bb7157420d0470dba0e6
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hello Ben,<br><br>Can we say in other words that proxy in such case should =
<br>indicate used TLS transport by inserting the sips URI in the <br>Record=
-Route header instead of using transport=3Dtls parameter.<br><br>Kind regar=
ds,<br>
- Tomasz Zieleniewski<br><br><div class=3D"gmail_quote">2009/8/7 BONNAERENS=
 Ben <span dir=3D"ltr">&lt;<a href=3D"mailto:ben.bonnaerens@alcatel-lucent.=
be">ben.bonnaerens@alcatel-lucent.be</a>&gt;</span><br><blockquote class=3D=
"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0=
pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello John,<br>
<div class=3D"im"><br>
&gt; Well, I know what this means, but will your average reader<br>
&gt; understand it? I think we are trying to say something along<br>
&gt; the lines that if the transport protocol changes you SHOULD<br>
&gt; use the double Record-Route technique, where the differences<br>
&gt; in transport protocol are reflected in different values of<br>
&gt; the transport parameter (udp/tcp/sctp) and/or different<br>
&gt; values of the URI scheme (sip/sips). Correct?<br>
<br>
</div>Your phrasing could lead to the understanding that we suggest that<br=
>
only a sips URI can be used to identify the TLS interface which is not<br>
correct.<br>
As the current text suggests, can the TLS interface be identified by any<br=
>
URI<br>
(sip or sips ; excluding the deprecated transport=3Dtls parameter) that<br>
resolves via RFC3263 procedures to the TLS transport.<br>
<br>
IMHO leaves your suggested phrasing more room for ambiguity<br>
compared to the current text.<br>
<br>
Best regards &amp; thanks,<br>
<font color=3D"#888888"><br>
Ben.<br>
</font><div><div></div><div class=3D"h5"><br>
&gt; -----Original Message-----<br>
&gt; From: Elwell, John [mailto:<a href=3D"mailto:john.elwell@siemens-enter=
prise.com">john.elwell@siemens-enterprise.com</a>]<br>
&gt; Sent: donderdag 6 augustus 2009 9:37<br>
&gt; To: BONNAERENS Ben; <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><b=
r>
&gt; Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes<br>
&gt;<br>
&gt; Well, I know what this means, but will your average reader<br>
&gt; understand it? I think we are trying to say something along<br>
&gt; the lines that if the transport protocol changes you SHOULD<br>
&gt; use the double Record-Route technique, where the differences<br>
&gt; in transport protocol are reflected in different values of<br>
&gt; the transport parameter (udp/tcp/sctp) and/or different<br>
&gt; values of the URI scheme (sip/sips). Correct?<br>
&gt;<br>
&gt; John<br>
&gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: <a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.or=
g</a> [mailto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org<=
/a>] On<br>
&gt; Behalf Of<br>
&gt; &gt; BONNAERENS Ben<br>
&gt; &gt; Sent: 05 August 2009 15:23<br>
&gt; &gt; To: <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
&gt; &gt; Subject: [Sip] draft-ietf-sip-record-route-fix-08 Changes<br>
&gt; &gt;<br>
&gt; &gt; Hi,<br>
&gt; &gt; version -08 of record-route fix has just been submitted.<br>
&gt; &gt;<br>
&gt; &gt; Change log:<br>
&gt; &gt; - Changed its status from BCP to PS,<br>
&gt; &gt; - Take into account IESG reviews and subsequent comments on the<b=
r>
&gt; &gt; mailing lists, they were summarized in Robert Sparks&#39;s docume=
nt<br>
&gt; &gt; (rrf.txt), so it has been integrated in version -08 with minor<br=
>
&gt; &gt; refactoring from the authors (me and Ben).<br>
&gt; &gt;<br>
&gt; &gt; The thoughest part was related to the SIP/SIPS question raised by=
<br>
&gt; &gt; Cullen<br>
&gt; (<a href=3D"http://datatracker.ietf.org/idtracker/draft-ietf-sip-recor=
d-r" target=3D"_blank">http://datatracker.ietf.org/idtracker/draft-ietf-sip=
-record-r</a><br>
&gt; &gt; oute-fix/c<br>
&gt; &gt; omment/98624/),<br>
&gt; &gt; so here is the text that was added in version -08 to answer this<=
br>
&gt; &gt; question:<br>
&gt; &gt; &quot;Thus, if the transport protocol changed between its<br>
&gt; &gt; =A0 =A0incoming and outgoing sides, the proxy SHOULD use the doub=
le<br>
&gt; &gt; =A0 =A0Record-Route technique and SHOULD add a transport<br>
&gt; parameter to each<br>
&gt; &gt; =A0 =A0of the Record-Route URIs it inserts. With the exception<br=
>
&gt; that if TLS<br>
&gt; &gt; is<br>
&gt; &gt; =A0 =A0used as the transport protocol on either side of the<br>
&gt; proxy, the URI<br>
&gt; &gt; =A0 =A0chosen to place in the Record-Route header field value<br>
&gt; reflecting<br>
&gt; &gt; the<br>
&gt; &gt; =A0 =A0interface using TLS will need to leverage the [RFC3263]<br=
>
&gt; mechanisms<br>
&gt; &gt; to<br>
&gt; &gt; =A0 =A0indicate that TLS must be used rather attempting to use th=
e<br>
&gt; &gt; deprecated<br>
&gt; &gt; =A0 =A0&quot;transport=3Dtls&quot; URI parameter. See [RFC3261] S=
ection 26.2.2 and<br>
&gt; &gt; =A0 =A0[I-D.ietf-sip-sips] Section 3.1.4 for more discussion.&quo=
t;<br>
&gt; &gt;<br>
&gt; &gt; Brds,<br>
&gt; &gt; Thomas &amp; Ben<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listi=
nfo/sip" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br=
>
&gt; &gt; This list is for NEW development of the core SIP Protocol Use<br>
&gt; &gt; <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implement=
ors@cs.columbia.edu</a> for questions on current sip Use<br>
&gt; &gt; <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new =
developments on the application of sip<br>
&gt; &gt;<br>
&gt;<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>
</div></div></blockquote></div><br>

--001636c5bb7157420d0470dba0e6--

From ben.bonnaerens@alcatel-lucent.be  Tue Aug 11 04:44:43 2009
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Hello John,

What we are trying to say is that the URI, identifying the TLS
interface, placed=20
in the Record-Route header must, after executing the RFC3263 DNS
procedureson it,=20
result in the TLS transport being selected.

This can be a sips URI or a sip URI (so called "best effort TLS" in
[I-D.ietf-sip-sips] chapter 3.1.3)=20
but we suggest to not use the deprecated "transport=3Dtls" parameter.

Best regards &Thanks,

Ben=20

RFC3263 Chapter 4.1
"   First, a client resolving a SIPS URI MUST discard any services that
   do not contain "SIPS" as the protocol in the service field.  The
   converse is not true, however.  A client resolving a SIP URI SHOULD
   retain records with "SIPS" as the protocol, if the client supports
   TLS." =20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Elwell, John
> Sent: dinsdag 11 augustus 2009 10:44
> To: BONNAERENS Ben; sip@ietf.org
> Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>=20
> In that case, I obviously misunderstood the text in the current draft.
> What exactly is the meaning of "will need to leverage the=20
> [RFC3263] mechanisms"?
>=20
> John
>=20
> > -----Original Message-----
> > From: BONNAERENS Ben [mailto:ben.bonnaerens@alcatel-lucent.be]
> > Sent: 07 August 2009 14:57
> > To: Elwell, John; sip@ietf.org
> > Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> >=20
> > Hello John,
> >=20
> > > Well, I know what this means, but will your average reader=20
> > > understand it? I think we are trying to say something along the=20
> > > lines that if the transport protocol changes you SHOULD use the=20
> > > double Record-Route technique, where the differences in transport=20
> > > protocol are reflected in different values of the transport=20
> > > parameter (udp/tcp/sctp) and/or different values of the=20
> URI scheme=20
> > > (sip/sips). Correct?
> >=20
> > Your phrasing could lead to the understanding that we suggest that=20
> > only a sips URI can be used to identify the TLS interface=20
> which is not=20
> > correct.
> > As the current text suggests, can the TLS interface be=20
> identified by=20
> > any URI (sip or sips ; excluding the deprecated transport=3Dtls=20
> > parameter) that resolves via RFC3263 procedures to the TLS=20
> transport.
> >=20
> > IMHO leaves your suggested phrasing more room for ambiguity=20
> compared=20
> > to the current text.
> >=20
> > Best regards & thanks,
> >=20
> > Ben. =20
> >=20
> > > -----Original Message-----
> > > From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> > > Sent: donderdag 6 augustus 2009 9:37
> > > To: BONNAERENS Ben; sip@ietf.org
> > > Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> > >=20
> > > Well, I know what this means, but will your average reader=20
> > > understand it? I think we are trying to say something along the=20
> > > lines that if the transport protocol changes you SHOULD use the=20
> > > double Record-Route technique, where the differences in transport=20
> > > protocol are reflected in different values of the transport=20
> > > parameter (udp/tcp/sctp) and/or different values of the=20
> URI scheme=20
> > > (sip/sips). Correct?
> > >=20
> > > John
> > >=20
> > > > -----Original Message-----
> > > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > > Behalf Of
> > > > BONNAERENS Ben
> > > > Sent: 05 August 2009 15:23
> > > > To: sip@ietf.org
> > > > Subject: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> > > >=20
> > > > Hi,
> > > > version -08 of record-route fix has just been submitted.
> > > >=20
> > > > Change log:
> > > > - Changed its status from BCP to PS,
> > > > - Take into account IESG reviews and subsequent comments on the=20
> > > > mailing lists, they were summarized in Robert Sparks's document=20
> > > > (rrf.txt), so it has been integrated in version -08 with minor=20
> > > > refactoring from the authors (me and Ben).
> > > >=20
> > > > The thoughest part was related to the SIP/SIPS question=20
> raised by=20
> > > > Cullen
> > > (http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-r
> > > > oute-fix/c
> > > > omment/98624/),
> > > > so here is the text that was added in version -08 to answer this
> > > > question:
> > > > "Thus, if the transport protocol changed between its
> > > >    incoming and outgoing sides, the proxy SHOULD use the double
> > > >    Record-Route technique and SHOULD add a transport
> > > parameter to each
> > > >    of the Record-Route URIs it inserts. With the exception
> > > that if TLS
> > > > is
> > > >    used as the transport protocol on either side of the
> > > proxy, the URI
> > > >    chosen to place in the Record-Route header field value
> > > reflecting
> > > > the
> > > >    interface using TLS will need to leverage the [RFC3263]
> > > mechanisms
> > > > to
> > > >    indicate that TLS must be used rather attempting to use the=20
> > > > deprecated
> > > >    "transport=3Dtls" URI parameter. See [RFC3261] Section=20
> 26.2.2 and
> > > >    [I-D.ietf-sip-sips] Section 3.1.4 for more discussion."
> > > >=20
> > > > Brds,
> > > > Thomas & Ben
> > > > _______________________________________________
> > > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > > sipping@ietf.org for new developments on the application of sip
> > > >=20
> > >=20
> >=20
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

From ben.bonnaerens@alcatel-lucent.be  Tue Aug 11 05:11:36 2009
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From: "BONNAERENS Ben" <ben.bonnaerens@alcatel-lucent.be>
To: "Tomasz Zieleniewski" <tzieleniewski@gmail.com>
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Hello Tomasz,
=20
It can be a sips URI but it is not restricted to sips.
=20
For example a SIP URI resolving to the TLS transport after executing the
RFC3263 procedures on it is possible.
=20
Best regards & thanks,
Ben.
=20
RFC3263 Chapter 4.1
"   First, a client resolving a SIPS URI MUST discard any services that
   do not contain "SIPS" as the protocol in the service field.  The
   converse is not true, however.  A client resolving a SIP URI SHOULD
   retain records with "SIPS" as the protocol, if the client supports
   TLS." =20
=20
[I-D.ietf-sip-sips] Section 3.1.3 ~ "best effort TLS"
=20
=20


________________________________

	From: Tomasz Zieleniewski [mailto:tzieleniewski@gmail.com]=20
	Sent: dinsdag 11 augustus 2009 13:02
	To: BONNAERENS Ben
	Cc: Elwell, John; sip@ietf.org
	Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
=09
=09
	Hello Ben,
=09
	Can we say in other words that proxy in such case should=20
	indicate used TLS transport by inserting the sips URI in the=20
	Record-Route header instead of using transport=3Dtls parameter.
=09
	Kind regards,
	- Tomasz Zieleniewski
=09
=09
	2009/8/7 BONNAERENS Ben <ben.bonnaerens@alcatel-lucent.be>
=09

		Hello John,
	=09

		> Well, I know what this means, but will your average
reader
		> understand it? I think we are trying to say something
along
		> the lines that if the transport protocol changes you
SHOULD
		> use the double Record-Route technique, where the
differences
		> in transport protocol are reflected in different
values of
		> the transport parameter (udp/tcp/sctp) and/or
different
		> values of the URI scheme (sip/sips). Correct?
	=09
	=09
		Your phrasing could lead to the understanding that we
suggest that
		only a sips URI can be used to identify the TLS
interface which is not
		correct.
		As the current text suggests, can the TLS interface be
identified by any
		URI
		(sip or sips ; excluding the deprecated transport=3Dtls
parameter) that
		resolves via RFC3263 procedures to the TLS transport.
	=09
		IMHO leaves your suggested phrasing more room for
ambiguity
		compared to the current text.
	=09
		Best regards & thanks,
	=09
		Ben.
	=09

		> -----Original Message-----
		> From: Elwell, John
[mailto:john.elwell@siemens-enterprise.com]
		> Sent: donderdag 6 augustus 2009 9:37
		> To: BONNAERENS Ben; sip@ietf.org
		> Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08
Changes
		>
		> Well, I know what this means, but will your average
reader
		> understand it? I think we are trying to say something
along
		> the lines that if the transport protocol changes you
SHOULD
		> use the double Record-Route technique, where the
differences
		> in transport protocol are reflected in different
values of
		> the transport parameter (udp/tcp/sctp) and/or
different
		> values of the URI scheme (sip/sips). Correct?
		>
		> John
		>
		> > -----Original Message-----
		> > From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org] On
		> Behalf Of
		> > BONNAERENS Ben
		> > Sent: 05 August 2009 15:23
		> > To: sip@ietf.org
		> > Subject: [Sip] draft-ietf-sip-record-route-fix-08
Changes
		> >
		> > Hi,
		> > version -08 of record-route fix has just been
submitted.
		> >
		> > Change log:
		> > - Changed its status from BCP to PS,
		> > - Take into account IESG reviews and subsequent
comments on the
		> > mailing lists, they were summarized in Robert
Sparks's document
		> > (rrf.txt), so it has been integrated in version -08
with minor
		> > refactoring from the authors (me and Ben).
		> >
		> > The thoughest part was related to the SIP/SIPS
question raised by
		> > Cullen
		>
(http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-r
		> > oute-fix/c
		> > omment/98624/),
		> > so here is the text that was added in version -08 to
answer this
		> > question:
		> > "Thus, if the transport protocol changed between its
		> >    incoming and outgoing sides, the proxy SHOULD use
the double
		> >    Record-Route technique and SHOULD add a transport
		> parameter to each
		> >    of the Record-Route URIs it inserts. With the
exception
		> that if TLS
		> > is
		> >    used as the transport protocol on either side of
the
		> proxy, the URI
		> >    chosen to place in the Record-Route header field
value
		> reflecting
		> > the
		> >    interface using TLS will need to leverage the
[RFC3263]
		> mechanisms
		> > to
		> >    indicate that TLS must be used rather attempting
to use the
		> > deprecated
		> >    "transport=3Dtls" URI parameter. See [RFC3261]
Section 26.2.2 and
		> >    [I-D.ietf-sip-sips] Section 3.1.4 for more
discussion."
		> >
		> > Brds,
		> > Thomas & Ben
		> > _______________________________________________
		> > 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
		> >
		>
		_______________________________________________
		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
	=09



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.5764" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D683474411-11082009>Hello Tomasz,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D683474411-11082009></SPAN></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =
size=3D2>It can=20
be a sips URI but it is not restricted to sips.</FONT></SPAN></DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =
size=3D2>For=20
example a SIP URI resolving to the TLS transport after executing=20
the</FONT></SPAN></DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =

size=3D2>RFC3263 procedures on it is possible.</FONT></SPAN></DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =
size=3D2>Best=20
regards &amp; thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =

size=3D2>Ben.</FONT></SPAN></DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =

size=3D2>RFC3263 Chapter 4.1<BR>"&nbsp;&nbsp; First, a client resolving =
a SIPS URI=20
MUST discard any services that<BR>&nbsp;&nbsp; do not contain "SIPS" as =
the=20
protocol in the service field.&nbsp; The<BR>&nbsp;&nbsp; converse is not =
true,=20
however.&nbsp; A client resolving a SIP URI SHOULD<BR>&nbsp;&nbsp; =
retain=20
records with "SIPS" as the protocol, if the client =
supports<BR>&nbsp;&nbsp;=20
TLS."&nbsp; </FONT></SPAN></DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =

size=3D2>[I-D.ietf-sip-sips] Section 3.1.3 ~ "best effort =
TLS"</FONT></SPAN></DIV>
<DIV><SPAN class=3D683474411-11082009><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Tomasz Zieleniewski=20
  [mailto:tzieleniewski@gmail.com] <BR><B>Sent:</B> dinsdag 11 augustus =
2009=20
  13:02<BR><B>To:</B> BONNAERENS Ben<BR><B>Cc:</B> Elwell, John;=20
  sip@ietf.org<BR><B>Subject:</B> Re: [Sip] =
draft-ietf-sip-record-route-fix-08=20
  Changes<BR></FONT><BR></DIV>
  <DIV></DIV>Hello Ben,<BR><BR>Can we say in other words that proxy in =
such case=20
  should <BR>indicate used TLS transport by inserting the sips URI in =
the=20
  <BR>Record-Route header instead of using transport=3Dtls =
parameter.<BR><BR>Kind=20
  regards,<BR>- Tomasz Zieleniewski<BR><BR>
  <DIV class=3Dgmail_quote>2009/8/7 BONNAERENS Ben <SPAN =
dir=3Dltr>&lt;<A=20
  =
href=3D"mailto:ben.bonnaerens@alcatel-lucent.be">ben.bonnaerens@alcatel-l=
ucent.be</A>&gt;</SPAN><BR>
  <BLOCKQUOTE class=3Dgmail_quote=20
  style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">Hello=20
    John,<BR>
    <DIV class=3Dim><BR>&gt; Well, I know what this means, but will your =
average=20
    reader<BR>&gt; understand it? I think we are trying to say something =

    along<BR>&gt; the lines that if the transport protocol changes you=20
    SHOULD<BR>&gt; use the double Record-Route technique, where the=20
    differences<BR>&gt; in transport protocol are reflected in different =
values=20
    of<BR>&gt; the transport parameter (udp/tcp/sctp) and/or =
different<BR>&gt;=20
    values of the URI scheme (sip/sips). Correct?<BR><BR></DIV>Your =
phrasing=20
    could lead to the understanding that we suggest that<BR>only a sips =
URI can=20
    be used to identify the TLS interface which is not<BR>correct.<BR>As =
the=20
    current text suggests, can the TLS interface be identified by=20
    any<BR>URI<BR>(sip or sips ; excluding the deprecated =
transport=3Dtls=20
    parameter) that<BR>resolves via RFC3263 procedures to the TLS=20
    transport.<BR><BR>IMHO leaves your suggested phrasing more room for=20
    ambiguity<BR>compared to the current text.<BR><BR>Best regards &amp; =

    thanks,<BR><FONT color=3D#888888><BR>Ben.<BR></FONT>
    <DIV>
    <DIV></DIV>
    <DIV class=3Dh5><BR>&gt; -----Original Message-----<BR>&gt; From: =
Elwell, John=20
    [mailto:<A=20
    =
href=3D"mailto:john.elwell@siemens-enterprise.com">john.elwell@siemens-en=
terprise.com</A>]<BR>&gt;=20
    Sent: donderdag 6 augustus 2009 9:37<BR>&gt; To: BONNAERENS Ben; <A=20
    href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>&gt; Subject: RE: =
[Sip]=20
    draft-ietf-sip-record-route-fix-08 Changes<BR>&gt;<BR>&gt; Well, I =
know what=20
    this means, but will your average reader<BR>&gt; understand it? I =
think we=20
    are trying to say something along<BR>&gt; the lines that if the =
transport=20
    protocol changes you SHOULD<BR>&gt; use the double Record-Route =
technique,=20
    where the differences<BR>&gt; in transport protocol are reflected in =

    different values of<BR>&gt; the transport parameter (udp/tcp/sctp) =
and/or=20
    different<BR>&gt; values of the URI scheme (sip/sips).=20
    Correct?<BR>&gt;<BR>&gt; John<BR>&gt;<BR>&gt; &gt; -----Original=20
    Message-----<BR>&gt; &gt; From: <A=20
    href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A> =
[mailto:<A=20
    href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A>] =
On<BR>&gt;=20
    Behalf Of<BR>&gt; &gt; BONNAERENS Ben<BR>&gt; &gt; Sent: 05 August =
2009=20
    15:23<BR>&gt; &gt; To: <A=20
    href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>&gt; &gt; Subject: =
[Sip]=20
    draft-ietf-sip-record-route-fix-08 Changes<BR>&gt; &gt;<BR>&gt; &gt; =

    Hi,<BR>&gt; &gt; version -08 of record-route fix has just been=20
    submitted.<BR>&gt; &gt;<BR>&gt; &gt; Change log:<BR>&gt; &gt; - =
Changed its=20
    status from BCP to PS,<BR>&gt; &gt; - Take into account IESG reviews =
and=20
    subsequent comments on the<BR>&gt; &gt; mailing lists, they were =
summarized=20
    in Robert Sparks's document<BR>&gt; &gt; (rrf.txt), so it has been=20
    integrated in version -08 with minor<BR>&gt; &gt; refactoring from =
the=20
    authors (me and Ben).<BR>&gt; &gt;<BR>&gt; &gt; The thoughest part =
was=20
    related to the SIP/SIPS question raised by<BR>&gt; &gt; =
Cullen<BR>&gt; (<A=20
    =
href=3D"http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-r"=20
    =
target=3D_blank>http://datatracker.ietf.org/idtracker/draft-ietf-sip-reco=
rd-r</A><BR>&gt;=20
    &gt; oute-fix/c<BR>&gt; &gt; omment/98624/),<BR>&gt; &gt; so here is =
the=20
    text that was added in version -08 to answer this<BR>&gt; &gt;=20
    question:<BR>&gt; &gt; "Thus, if the transport protocol changed =
between=20
    its<BR>&gt; &gt; &nbsp; &nbsp;incoming and outgoing sides, the proxy =
SHOULD=20
    use the double<BR>&gt; &gt; &nbsp; &nbsp;Record-Route technique and =
SHOULD=20
    add a transport<BR>&gt; parameter to each<BR>&gt; &gt; &nbsp; =
&nbsp;of the=20
    Record-Route URIs it inserts. With the exception<BR>&gt; that if =
TLS<BR>&gt;=20
    &gt; is<BR>&gt; &gt; &nbsp; &nbsp;used as the transport protocol on =
either=20
    side of the<BR>&gt; proxy, the URI<BR>&gt; &gt; &nbsp; &nbsp;chosen =
to place=20
    in the Record-Route header field value<BR>&gt; reflecting<BR>&gt; =
&gt;=20
    the<BR>&gt; &gt; &nbsp; &nbsp;interface using TLS will need to =
leverage the=20
    [RFC3263]<BR>&gt; mechanisms<BR>&gt; &gt; to<BR>&gt; &gt; &nbsp;=20
    &nbsp;indicate that TLS must be used rather attempting to use =
the<BR>&gt;=20
    &gt; deprecated<BR>&gt; &gt; &nbsp; &nbsp;"transport=3Dtls" URI =
parameter. See=20
    [RFC3261] Section 26.2.2 and<BR>&gt; &gt; &nbsp; =
&nbsp;[I-D.ietf-sip-sips]=20
    Section 3.1.4 for more discussion."<BR>&gt; &gt;<BR>&gt; &gt; =
Brds,<BR>&gt;=20
    &gt; Thomas &amp; Ben<BR>&gt; &gt;=20
    _______________________________________________<BR>&gt; &gt; Sip =
mailing=20
    list &nbsp;<A href=3D"https://www.ietf.org/mailman/listinfo/sip"=20
    =
target=3D_blank>https://www.ietf.org/mailman/listinfo/sip</A><BR>&gt; =
&gt;=20
    This list is for NEW development of the core SIP Protocol =
Use<BR>&gt; &gt;=20
    <A=20
    =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>=20
    for questions on current sip Use<BR>&gt; &gt; <A=20
    href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> for new =
developments on=20
    the application of sip<BR>&gt;=20
    =
&gt;<BR>&gt;<BR>_______________________________________________<BR>Sip=20
    mailing list &nbsp;<A =
href=3D"https://www.ietf.org/mailman/listinfo/sip"=20
    =
target=3D_blank>https://www.ietf.org/mailman/listinfo/sip</A><BR>This =
list is=20
    for NEW development of the core SIP Protocol<BR>Use <A=20
    =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>=20
    for questions on current sip<BR>Use <A=20
    href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> for new =
developments on=20
    the application of=20
sip<BR></DIV></DIV></BLOCKQUOTE></DIV><BR></BLOCKQUOTE></BODY></HTML>

------_=_NextPart_001_01CA1A7A.8BD8D78B--

From john.elwell@siemens-enterprise.com  Tue Aug 11 05:25:23 2009
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Date: Tue, 11 Aug 2009 13:21:10 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Cc: sip@ietf.org
Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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=20

> -----Original Message-----
> From: BONNAERENS Ben [mailto:ben.bonnaerens@alcatel-lucent.be]=20
> Sent: 11 August 2009 12:44
> To: Elwell, John
> Cc: sip@ietf.org; Thomas Froment
> Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>=20
> Hello John,
>=20
> What we are trying to say is that the URI, identifying the TLS
> interface, placed=20
> in the Record-Route header must, after executing the RFC3263 DNS
> procedureson it,=20
> result in the TLS transport being selected.
>=20
> This can be a sips URI or a sip URI (so called "best effort TLS" in
> [I-D.ietf-sip-sips] chapter 3.1.3)=20
> but we suggest to not use the deprecated "transport=3Dtls" parameter.
[JRE] So if you want to ensure TLS, either you use a sips URI or you
populate your DNS records such that a sip URI always resolves to TLS
transport - correct?
If so, I think the average reader would have a hard time deducing this
from the present text.

John

>=20
> Best regards &Thanks,
>=20
> Ben=20
>=20
> RFC3263 Chapter 4.1
> "   First, a client resolving a SIPS URI MUST discard any=20
> services that
>    do not contain "SIPS" as the protocol in the service field.  The
>    converse is not true, however.  A client resolving a SIP URI SHOULD
>    retain records with "SIPS" as the protocol, if the client supports
>    TLS." =20
>=20
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> > Behalf Of Elwell, John
> > Sent: dinsdag 11 augustus 2009 10:44
> > To: BONNAERENS Ben; sip@ietf.org
> > Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> >=20
> > In that case, I obviously misunderstood the text in the=20
> current draft.
> > What exactly is the meaning of "will need to leverage the=20
> > [RFC3263] mechanisms"?
> >=20
> > John
> >=20
> > > -----Original Message-----
> > > From: BONNAERENS Ben [mailto:ben.bonnaerens@alcatel-lucent.be]
> > > Sent: 07 August 2009 14:57
> > > To: Elwell, John; sip@ietf.org
> > > Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> > >=20
> > > Hello John,
> > >=20
> > > > Well, I know what this means, but will your average reader=20
> > > > understand it? I think we are trying to say something along the=20
> > > > lines that if the transport protocol changes you SHOULD use the=20
> > > > double Record-Route technique, where the differences in=20
> transport=20
> > > > protocol are reflected in different values of the transport=20
> > > > parameter (udp/tcp/sctp) and/or different values of the=20
> > URI scheme=20
> > > > (sip/sips). Correct?
> > >=20
> > > Your phrasing could lead to the understanding that we=20
> suggest that=20
> > > only a sips URI can be used to identify the TLS interface=20
> > which is not=20
> > > correct.
> > > As the current text suggests, can the TLS interface be=20
> > identified by=20
> > > any URI (sip or sips ; excluding the deprecated transport=3Dtls=20
> > > parameter) that resolves via RFC3263 procedures to the TLS=20
> > transport.
> > >=20
> > > IMHO leaves your suggested phrasing more room for ambiguity=20
> > compared=20
> > > to the current text.
> > >=20
> > > Best regards & thanks,
> > >=20
> > > Ben. =20
> > >=20
> > > > -----Original Message-----
> > > > From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> > > > Sent: donderdag 6 augustus 2009 9:37
> > > > To: BONNAERENS Ben; sip@ietf.org
> > > > Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> > > >=20
> > > > Well, I know what this means, but will your average reader=20
> > > > understand it? I think we are trying to say something along the=20
> > > > lines that if the transport protocol changes you SHOULD use the=20
> > > > double Record-Route technique, where the differences in=20
> transport=20
> > > > protocol are reflected in different values of the transport=20
> > > > parameter (udp/tcp/sctp) and/or different values of the=20
> > URI scheme=20
> > > > (sip/sips). Correct?
> > > >=20
> > > > John
> > > >=20
> > > > > -----Original Message-----
> > > > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > > > Behalf Of
> > > > > BONNAERENS Ben
> > > > > Sent: 05 August 2009 15:23
> > > > > To: sip@ietf.org
> > > > > Subject: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> > > > >=20
> > > > > Hi,
> > > > > version -08 of record-route fix has just been submitted.
> > > > >=20
> > > > > Change log:
> > > > > - Changed its status from BCP to PS,
> > > > > - Take into account IESG reviews and subsequent=20
> comments on the=20
> > > > > mailing lists, they were summarized in Robert=20
> Sparks's document=20
> > > > > (rrf.txt), so it has been integrated in version -08=20
> with minor=20
> > > > > refactoring from the authors (me and Ben).
> > > > >=20
> > > > > The thoughest part was related to the SIP/SIPS question=20
> > raised by=20
> > > > > Cullen
> > > > (http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-r
> > > > > oute-fix/c
> > > > > omment/98624/),
> > > > > so here is the text that was added in version -08 to=20
> answer this
> > > > > question:
> > > > > "Thus, if the transport protocol changed between its
> > > > >    incoming and outgoing sides, the proxy SHOULD use=20
> the double
> > > > >    Record-Route technique and SHOULD add a transport
> > > > parameter to each
> > > > >    of the Record-Route URIs it inserts. With the exception
> > > > that if TLS
> > > > > is
> > > > >    used as the transport protocol on either side of the
> > > > proxy, the URI
> > > > >    chosen to place in the Record-Route header field value
> > > > reflecting
> > > > > the
> > > > >    interface using TLS will need to leverage the [RFC3263]
> > > > mechanisms
> > > > > to
> > > > >    indicate that TLS must be used rather attempting=20
> to use the=20
> > > > > deprecated
> > > > >    "transport=3Dtls" URI parameter. See [RFC3261] Section=20
> > 26.2.2 and
> > > > >    [I-D.ietf-sip-sips] Section 3.1.4 for more discussion."
> > > > >=20
> > > > > Brds,
> > > > > Thomas & Ben
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > > sip-implementors@cs.columbia.edu for questions on=20
> > current sip Use=20
> > > > > sipping@ietf.org for new developments on the=20
> application of sip
> > > > >=20
> > > >=20
> > >=20
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip=20
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
>=20

From brett@broadsoft.com  Tue Aug 11 08:12:21 2009
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From: Brett Tate <brett@broadsoft.com>
To: "stlevy@cisco.com" <stlevy@cisco.com>, "sip@ietf.org" <sip@ietf.org>
Date: Tue, 11 Aug 2009 08:11:30 -0700
Thread-Topic: draft-levy-sip-diversion-10: examples containing uselessly quoted values
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Subject: [Sip] draft-levy-sip-diversion-10: examples containing uselessly quoted values
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Greetings,

Concerning draft-levy-sip-diversion-10, several examples still contain quot=
ed values which conflict with the need for quotes within the ABNF.  This is=
 leading to frequent interop issues and ambiguity.  Please remove the usele=
ss quotes from the examples (my preference) or clearly indicate that a usel=
essly quoted value is equivalent to the non quoted value.

For example... diversion-screen ABNF shows quotes are not needed for yes an=
d no.  However examples (such as 10.3.5) include quotes for yes.

Another example... diversion-privacy ABNF shows quotes are not needed for f=
ull and off.  However examples (such as 10.3.5) include quotes for full and=
 off.  The 10.3.6 example contains open quote off without the close quote.

Thanks,
Brett


From rjsparks@nostrum.com  Tue Aug 11 18:35:04 2009
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Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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John -

Would this substitution address your concern?

      When TLS is used on the transport on either side of the proxy  
the URI
      chosen to place in the Record-Route header field value  
reflecting the
      interface using TLS will need to use a sips: URI or ensure that  
a sip:
      URI resolves (per the procedures of [RFC3263]) to using TLS rather
      than attempting to use the deprecated  "transport=tls" URI  
parameter.
      See [RFC3261] Section 26.2.2 and [I-D.ietf-sip-sips] Section 3.1.4
      for more discussion

RjS

On Aug 11, 2009, at 7:21 AM, Elwell, John wrote:

>
>
>> -----Original Message-----
>> From: BONNAERENS Ben [mailto:ben.bonnaerens@alcatel-lucent.be]
>> Sent: 11 August 2009 12:44
>> To: Elwell, John
>> Cc: sip@ietf.org; Thomas Froment
>> Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>>
>> Hello John,
>>
>> What we are trying to say is that the URI, identifying the TLS
>> interface, placed
>> in the Record-Route header must, after executing the RFC3263 DNS
>> procedureson it,
>> result in the TLS transport being selected.
>>
>> This can be a sips URI or a sip URI (so called "best effort TLS" in
>> [I-D.ietf-sip-sips] chapter 3.1.3)
>> but we suggest to not use the deprecated "transport=tls" parameter.
> [JRE] So if you want to ensure TLS, either you use a sips URI or you
> populate your DNS records such that a sip URI always resolves to TLS
> transport - correct?
> If so, I think the average reader would have a hard time deducing this
> from the present text.
>
> John
>
>>
>> Best regards &Thanks,
>>
>> Ben
>>
>> RFC3263 Chapter 4.1
>> "   First, a client resolving a SIPS URI MUST discard any
>> services that
>>   do not contain "SIPS" as the protocol in the service field.  The
>>   converse is not true, however.  A client resolving a SIP URI SHOULD
>>   retain records with "SIPS" as the protocol, if the client supports
>>   TLS."
>>
>>> -----Original Message-----
>>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
>>> Behalf Of Elwell, John
>>> Sent: dinsdag 11 augustus 2009 10:44
>>> To: BONNAERENS Ben; sip@ietf.org
>>> Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>>>
>>> In that case, I obviously misunderstood the text in the
>> current draft.
>>> What exactly is the meaning of "will need to leverage the
>>> [RFC3263] mechanisms"?
>>>
>>> John
>>>
>>>> -----Original Message-----
>>>> From: BONNAERENS Ben [mailto:ben.bonnaerens@alcatel-lucent.be]
>>>> Sent: 07 August 2009 14:57
>>>> To: Elwell, John; sip@ietf.org
>>>> Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>>>>
>>>> Hello John,
>>>>
>>>>> Well, I know what this means, but will your average reader
>>>>> understand it? I think we are trying to say something along the
>>>>> lines that if the transport protocol changes you SHOULD use the
>>>>> double Record-Route technique, where the differences in
>> transport
>>>>> protocol are reflected in different values of the transport
>>>>> parameter (udp/tcp/sctp) and/or different values of the
>>> URI scheme
>>>>> (sip/sips). Correct?
>>>>
>>>> Your phrasing could lead to the understanding that we
>> suggest that
>>>> only a sips URI can be used to identify the TLS interface
>>> which is not
>>>> correct.
>>>> As the current text suggests, can the TLS interface be
>>> identified by
>>>> any URI (sip or sips ; excluding the deprecated transport=tls
>>>> parameter) that resolves via RFC3263 procedures to the TLS
>>> transport.
>>>>
>>>> IMHO leaves your suggested phrasing more room for ambiguity
>>> compared
>>>> to the current text.
>>>>
>>>> Best regards & thanks,
>>>>
>>>> Ben.
>>>>
>>>>> -----Original Message-----
>>>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>>>> Sent: donderdag 6 augustus 2009 9:37
>>>>> To: BONNAERENS Ben; sip@ietf.org
>>>>> Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>>>>>
>>>>> Well, I know what this means, but will your average reader
>>>>> understand it? I think we are trying to say something along the
>>>>> lines that if the transport protocol changes you SHOULD use the
>>>>> double Record-Route technique, where the differences in
>> transport
>>>>> protocol are reflected in different values of the transport
>>>>> parameter (udp/tcp/sctp) and/or different values of the
>>> URI scheme
>>>>> (sip/sips). Correct?
>>>>>
>>>>> John
>>>>>
>>>>>> -----Original Message-----
>>>>>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
>>>>> Behalf Of
>>>>>> BONNAERENS Ben
>>>>>> Sent: 05 August 2009 15:23
>>>>>> To: sip@ietf.org
>>>>>> Subject: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>>>>>>
>>>>>> Hi,
>>>>>> version -08 of record-route fix has just been submitted.
>>>>>>
>>>>>> Change log:
>>>>>> - Changed its status from BCP to PS,
>>>>>> - Take into account IESG reviews and subsequent
>> comments on the
>>>>>> mailing lists, they were summarized in Robert
>> Sparks's document
>>>>>> (rrf.txt), so it has been integrated in version -08
>> with minor
>>>>>> refactoring from the authors (me and Ben).
>>>>>>
>>>>>> The thoughest part was related to the SIP/SIPS question
>>> raised by
>>>>>> Cullen
>>>>> (http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-r
>>>>>> oute-fix/c
>>>>>> omment/98624/),
>>>>>> so here is the text that was added in version -08 to
>> answer this
>>>>>> question:
>>>>>> "Thus, if the transport protocol changed between its
>>>>>>   incoming and outgoing sides, the proxy SHOULD use
>> the double
>>>>>>   Record-Route technique and SHOULD add a transport
>>>>> parameter to each
>>>>>>   of the Record-Route URIs it inserts. With the exception
>>>>> that if TLS
>>>>>> is
>>>>>>   used as the transport protocol on either side of the
>>>>> proxy, the URI
>>>>>>   chosen to place in the Record-Route header field value
>>>>> reflecting
>>>>>> the
>>>>>>   interface using TLS will need to leverage the [RFC3263]
>>>>> mechanisms
>>>>>> to
>>>>>>   indicate that TLS must be used rather attempting
>> to use the
>>>>>> deprecated
>>>>>>   "transport=tls" URI parameter. See [RFC3261] Section
>>> 26.2.2 and
>>>>>>   [I-D.ietf-sip-sips] Section 3.1.4 for more discussion."
>>>>>>
>>>>>> Brds,
>>>>>> Thomas & Ben
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>
>>>>
>>> _______________________________________________
>>> 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
>>>
>>
> _______________________________________________
> 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


From john.elwell@siemens-enterprise.com  Wed Aug 12 00:01:24 2009
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Date: Wed, 12 Aug 2009 07:41:52 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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To: Robert Sparks <rjsparks@nostrum.com>
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Cc: sip@ietf.org, BONNAERENS Ben <ben.bonnaerens@alcatel-lucent.be>
Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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Robert,

Yes, that would be a lot better.

Thanks.

John
=20

> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]=20
> Sent: 12 August 2009 02:35
> To: Elwell, John
> Cc: BONNAERENS Ben; sip@ietf.org
> Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>=20
> John -
>=20
> Would this substitution address your concern?
>=20
>       When TLS is used on the transport on either side of the proxy =20
> the URI
>       chosen to place in the Record-Route header field value =20
> reflecting the
>       interface using TLS will need to use a sips: URI or=20
> ensure that =20
> a sip:
>       URI resolves (per the procedures of [RFC3263]) to using=20
> TLS rather
>       than attempting to use the deprecated  "transport=3Dtls" URI =20
> parameter.
>       See [RFC3261] Section 26.2.2 and [I-D.ietf-sip-sips]=20
> Section 3.1.4
>       for more discussion
>=20
> RjS
>=20
> On Aug 11, 2009, at 7:21 AM, Elwell, John wrote:
>=20
> >
> >
> >> -----Original Message-----
> >> From: BONNAERENS Ben [mailto:ben.bonnaerens@alcatel-lucent.be]
> >> Sent: 11 August 2009 12:44
> >> To: Elwell, John
> >> Cc: sip@ietf.org; Thomas Froment
> >> Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> >>
> >> Hello John,
> >>
> >> What we are trying to say is that the URI, identifying the TLS
> >> interface, placed
> >> in the Record-Route header must, after executing the RFC3263 DNS
> >> procedureson it,
> >> result in the TLS transport being selected.
> >>
> >> This can be a sips URI or a sip URI (so called "best effort TLS" in
> >> [I-D.ietf-sip-sips] chapter 3.1.3)
> >> but we suggest to not use the deprecated "transport=3Dtls" =
parameter.
> > [JRE] So if you want to ensure TLS, either you use a sips URI or you
> > populate your DNS records such that a sip URI always resolves to TLS
> > transport - correct?
> > If so, I think the average reader would have a hard time=20
> deducing this
> > from the present text.
> >
> > John
> >
> >>
> >> Best regards &Thanks,
> >>
> >> Ben
> >>
> >> RFC3263 Chapter 4.1
> >> "   First, a client resolving a SIPS URI MUST discard any
> >> services that
> >>   do not contain "SIPS" as the protocol in the service field.  The
> >>   converse is not true, however.  A client resolving a SIP=20
> URI SHOULD
> >>   retain records with "SIPS" as the protocol, if the=20
> client supports
> >>   TLS."
> >>
> >>> -----Original Message-----
> >>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> >>> Behalf Of Elwell, John
> >>> Sent: dinsdag 11 augustus 2009 10:44
> >>> To: BONNAERENS Ben; sip@ietf.org
> >>> Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> >>>
> >>> In that case, I obviously misunderstood the text in the
> >> current draft.
> >>> What exactly is the meaning of "will need to leverage the
> >>> [RFC3263] mechanisms"?
> >>>
> >>> John
> >>>
> >>>> -----Original Message-----
> >>>> From: BONNAERENS Ben [mailto:ben.bonnaerens@alcatel-lucent.be]
> >>>> Sent: 07 August 2009 14:57
> >>>> To: Elwell, John; sip@ietf.org
> >>>> Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> >>>>
> >>>> Hello John,
> >>>>
> >>>>> Well, I know what this means, but will your average reader
> >>>>> understand it? I think we are trying to say something along the
> >>>>> lines that if the transport protocol changes you SHOULD use the
> >>>>> double Record-Route technique, where the differences in
> >> transport
> >>>>> protocol are reflected in different values of the transport
> >>>>> parameter (udp/tcp/sctp) and/or different values of the
> >>> URI scheme
> >>>>> (sip/sips). Correct?
> >>>>
> >>>> Your phrasing could lead to the understanding that we
> >> suggest that
> >>>> only a sips URI can be used to identify the TLS interface
> >>> which is not
> >>>> correct.
> >>>> As the current text suggests, can the TLS interface be
> >>> identified by
> >>>> any URI (sip or sips ; excluding the deprecated transport=3Dtls
> >>>> parameter) that resolves via RFC3263 procedures to the TLS
> >>> transport.
> >>>>
> >>>> IMHO leaves your suggested phrasing more room for ambiguity
> >>> compared
> >>>> to the current text.
> >>>>
> >>>> Best regards & thanks,
> >>>>
> >>>> Ben.
> >>>>
> >>>>> -----Original Message-----
> >>>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> >>>>> Sent: donderdag 6 augustus 2009 9:37
> >>>>> To: BONNAERENS Ben; sip@ietf.org
> >>>>> Subject: RE: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> >>>>>
> >>>>> Well, I know what this means, but will your average reader
> >>>>> understand it? I think we are trying to say something along the
> >>>>> lines that if the transport protocol changes you SHOULD use the
> >>>>> double Record-Route technique, where the differences in
> >> transport
> >>>>> protocol are reflected in different values of the transport
> >>>>> parameter (udp/tcp/sctp) and/or different values of the
> >>> URI scheme
> >>>>> (sip/sips). Correct?
> >>>>>
> >>>>> John
> >>>>>
> >>>>>> -----Original Message-----
> >>>>>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> >>>>> Behalf Of
> >>>>>> BONNAERENS Ben
> >>>>>> Sent: 05 August 2009 15:23
> >>>>>> To: sip@ietf.org
> >>>>>> Subject: [Sip] draft-ietf-sip-record-route-fix-08 Changes
> >>>>>>
> >>>>>> Hi,
> >>>>>> version -08 of record-route fix has just been submitted.
> >>>>>>
> >>>>>> Change log:
> >>>>>> - Changed its status from BCP to PS,
> >>>>>> - Take into account IESG reviews and subsequent
> >> comments on the
> >>>>>> mailing lists, they were summarized in Robert
> >> Sparks's document
> >>>>>> (rrf.txt), so it has been integrated in version -08
> >> with minor
> >>>>>> refactoring from the authors (me and Ben).
> >>>>>>
> >>>>>> The thoughest part was related to the SIP/SIPS question
> >>> raised by
> >>>>>> Cullen
> >>>>> (http://datatracker.ietf.org/idtracker/draft-ietf-sip-record-r
> >>>>>> oute-fix/c
> >>>>>> omment/98624/),
> >>>>>> so here is the text that was added in version -08 to
> >> answer this
> >>>>>> question:
> >>>>>> "Thus, if the transport protocol changed between its
> >>>>>>   incoming and outgoing sides, the proxy SHOULD use
> >> the double
> >>>>>>   Record-Route technique and SHOULD add a transport
> >>>>> parameter to each
> >>>>>>   of the Record-Route URIs it inserts. With the exception
> >>>>> that if TLS
> >>>>>> is
> >>>>>>   used as the transport protocol on either side of the
> >>>>> proxy, the URI
> >>>>>>   chosen to place in the Record-Route header field value
> >>>>> reflecting
> >>>>>> the
> >>>>>>   interface using TLS will need to leverage the [RFC3263]
> >>>>> mechanisms
> >>>>>> to
> >>>>>>   indicate that TLS must be used rather attempting
> >> to use the
> >>>>>> deprecated
> >>>>>>   "transport=3Dtls" URI parameter. See [RFC3261] Section
> >>> 26.2.2 and
> >>>>>>   [I-D.ietf-sip-sips] Section 3.1.4 for more discussion."
> >>>>>>
> >>>>>> Brds,
> >>>>>> Thomas & Ben
> >>>>>> _______________________________________________
> >>>>>> 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
> >>>>>>
> >>>>>
> >>>>
> >>> _______________________________________________
> >>> 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=20
> application of sip
> >>>
> >>
> > _______________________________________________
> > 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
>=20
>=20

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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-09.txt
	Pages           : 26
	Date            : 2009-08-12

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 not possible
to make one header have the characteristics of both interfaces 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-09.txt

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On Aug 11, 2009, at 8:35 PM, Robert Sparks wrote:

> John -
>
> Would this substitution address your concern?
>
>     When TLS is used on the transport on either side of the proxy  
> the URI
>     chosen to place in the Record-Route header field value  
> reflecting the
>     interface using TLS will need to use a sips: URI or ensure that  
> a sip:
>     URI resolves (per the procedures of [RFC3263]) to using TLS rather
>     than attempting to use the deprecated  "transport=tls" URI  
> parameter.
>     See [RFC3261] Section 26.2.2 and [I-D.ietf-sip-sips] Section 3.1.4
>     for more discussion
>


That makes my head hurt!

Compound sentences should make sense even with clauses redacted.

Let's play some games with this mind twister:


First I'll try to scope the sub-clauses:

> (When TLS is used (on the transport (on either side of the proxy)))  
> the URI (chosen to place (in the Record-Route header field value  
> (reflecting the interface using TLS))) will need to use a sips: URI  
> or ensure that a sip: URI resolves (per the procedures of [RFC3263])  
> to using TLS (rather than attempting to use the deprecated   
> "transport=tls" URI parameter).
>

One reduction to see if it makes sense:

> When TLS is used the URI chosen to place in the Record-Route header  
> field value will need to use a sips: URI or ensure that a sip: URI  
> resolves (per the procedures of [RFC3263]) to using TLS rather than  
> attempting to use the deprecated  "transport=tls" URI parameter.


Let's pull some smaller pieces out of this:

> When TLS is used the URI chosen to place in the Record-Route header  
> field value will need to use a sips: URI.


"chosen to place" is clumsy. Let's substitute "placed", add a comma  
for the subordination,  and see if it gets better.

> When TLS is used, the URI placed in the Record-Route header field  
> value will need to use a sips: URI.

What does it mean for a URI to use a URI? I think perhaps it needs to  
BE a SIPS URI.

Rewritten:

> When TLS is used, the URI placed in the Record-Route header field  
> value will need to be a sips: URI.

Now lets look at the other half of that clause:

> When TLS is used, the URI placed in the Record-Route header field  
> value will need to ensure that a sip: URI resolves (per the  
> procedures of [RFC3263]) to using TLS rather than attempting to use  
> the deprecated  "transport=tls" URI parameter.


How does a URI ensure that a ensure something about a URI?

Maybe you mean:

> When TLS is used, the URI placed in the Record-Route header field  
> value MUST resolve (per the procedures of [RFC3263]) to using TLS  
> rather than attempting to use the deprecated  "transport=tls" URI  
> parameter.


I'm still stumbling over "resolve to using TLS rather than attempting  
to use the deprecated  "transport=tls" URI parameter." So let's just  
lop off the sentence from "rather than . . ." and move that out.

What's wrong with

> When TLS is used, the URI placed in the Record-Route header field  
> value MUST resolve (per the procedures of [RFC3263]) to using TLS .

Is "resolve to using TLS" the same thing as "resolve using TLS"? I  
think they're subtly different, but the complexity still scares me. I  
think we're trying to say that the application of RFC 3263 procedures  
to the URI must result in a next-hop that is reachable only by TLS.

So let's try a rewrite based on these assumptions. How about this:

> When TLS is used, the URI placed in the Record-Route header field  
> MUST encode a next-hop that will be reached using TLS. There are two  
> ways for this to work. The first way is for the URI placed in the  
> Record-Route to be a sips: URI. The second is for the URI placed in  
> the Record-Route to be constructed such that application of  RFC  
> 3263 resolution procedures
> to that URI produces a result reachable only using TLS. Proxies  
> compliant with this specification MUST NOT use a "transport=tls"  
> parameter on the URI placed in the Record-Route because the  
> "transport=tls" usage was deprecated by RFC 3261.

Does that mean the same thing to you as the original megasentence did?

--
Dean




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Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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I skipped straight to the end (will come back and read how you got  
there later).
But yes, I think the rewrite you have at the end says the the same  
thing.

RjS
On Aug 12, 2009, at 2:42 PM, Dean Willis wrote:

>
> On Aug 11, 2009, at 8:35 PM, Robert Sparks wrote:
>
>> John -
>>
>> Would this substitution address your concern?
>>
>>    When TLS is used on the transport on either side of the proxy  
>> the URI
>>    chosen to place in the Record-Route header field value  
>> reflecting the
>>    interface using TLS will need to use a sips: URI or ensure that  
>> a sip:
>>    URI resolves (per the procedures of [RFC3263]) to using TLS rather
>>    than attempting to use the deprecated  "transport=tls" URI  
>> parameter.
>>    See [RFC3261] Section 26.2.2 and [I-D.ietf-sip-sips] Section 3.1.4
>>    for more discussion
>>
>
>
> That makes my head hurt!
>
> Compound sentences should make sense even with clauses redacted.
>
> Let's play some games with this mind twister:
>
>
> First I'll try to scope the sub-clauses:
>
>> (When TLS is used (on the transport (on either side of the proxy)))  
>> the URI (chosen to place (in the Record-Route header field value  
>> (reflecting the interface using TLS))) will need to use a sips: URI  
>> or ensure that a sip: URI resolves (per the procedures of  
>> [RFC3263]) to using TLS (rather than attempting to use the  
>> deprecated  "transport=tls" URI parameter).
>>
>
> One reduction to see if it makes sense:
>
>> When TLS is used the URI chosen to place in the Record-Route header  
>> field value will need to use a sips: URI or ensure that a sip: URI  
>> resolves (per the procedures of [RFC3263]) to using TLS rather than  
>> attempting to use the deprecated  "transport=tls" URI parameter.
>
>
> Let's pull some smaller pieces out of this:
>
>> When TLS is used the URI chosen to place in the Record-Route header  
>> field value will need to use a sips: URI.
>
>
> "chosen to place" is clumsy. Let's substitute "placed", add a comma  
> for the subordination,  and see if it gets better.
>
>> When TLS is used, the URI placed in the Record-Route header field  
>> value will need to use a sips: URI.
>
> What does it mean for a URI to use a URI? I think perhaps it needs  
> to BE a SIPS URI.
>
> Rewritten:
>
>> When TLS is used, the URI placed in the Record-Route header field  
>> value will need to be a sips: URI.
>
> Now lets look at the other half of that clause:
>
>> When TLS is used, the URI placed in the Record-Route header field  
>> value will need to ensure that a sip: URI resolves (per the  
>> procedures of [RFC3263]) to using TLS rather than attempting to use  
>> the deprecated  "transport=tls" URI parameter.
>
>
> How does a URI ensure that a ensure something about a URI?
>
> Maybe you mean:
>
>> When TLS is used, the URI placed in the Record-Route header field  
>> value MUST resolve (per the procedures of [RFC3263]) to using TLS  
>> rather than attempting to use the deprecated  "transport=tls" URI  
>> parameter.
>
>
> I'm still stumbling over "resolve to using TLS rather than  
> attempting to use the deprecated  "transport=tls" URI parameter." So  
> let's just lop off the sentence from "rather than . . ." and move  
> that out.
>
> What's wrong with
>
>> When TLS is used, the URI placed in the Record-Route header field  
>> value MUST resolve (per the procedures of [RFC3263]) to using TLS .
>
> Is "resolve to using TLS" the same thing as "resolve using TLS"? I  
> think they're subtly different, but the complexity still scares me.  
> I think we're trying to say that the application of RFC 3263  
> procedures to the URI must result in a next-hop that is reachable  
> only by TLS.
>
> So let's try a rewrite based on these assumptions. How about this:
>
>> When TLS is used, the URI placed in the Record-Route header field  
>> MUST encode a next-hop that will be reached using TLS. There are  
>> two ways for this to work. The first way is for the URI placed in  
>> the Record-Route to be a sips: URI. The second is for the URI  
>> placed in the Record-Route to be constructed such that application  
>> of  RFC 3263 resolution procedures
>> to that URI produces a result reachable only using TLS. Proxies  
>> compliant with this specification MUST NOT use a "transport=tls"  
>> parameter on the URI placed in the Record-Route because the  
>> "transport=tls" usage was deprecated by RFC 3261.
>
> Does that mean the same thing to you as the original megasentence did?
>
> --
> Dean
>
>
>


From dean.willis@softarmor.com  Wed Aug 12 13:18:02 2009
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On Aug 11, 2009, at 8:35 PM, Robert Sparks wrote:

> John -
>
> Would this substitution address your concern?
>
>    When TLS is used on the transport on either side of the proxy the  
> URI
>    chosen to place in the Record-Route header field value reflecting  
> the
>    interface using TLS will need to use a sips: URI or ensure that a  
> sip:
>    URI resolves (per the procedures of [RFC3263]) to using TLS rather
>    than attempting to use the deprecated  "transport=tls" URI  
> parameter.
>    See [RFC3261] Section 26.2.2 and [I-D.ietf-sip-sips] Section 3.1.4
>    for more discussion
>


That makes my head hurt!

Compound sentences should make sense even with clauses redacted.

Let's play some games with this mind twister:


First I'll try to scope the sub-clauses:

> (When TLS is used (on the transport (on either side of the proxy)))  
> the URI (chosen to place (in the Record-Route header field value  
> (reflecting the interface using TLS))) will need to use a sips: URI  
> or ensure that a sip: URI resolves (per the procedures of [RFC3263])  
> to using TLS (rather than attempting to use the deprecated   
> "transport=tls" URI parameter).
>

One reduction to see if it makes sense:

> When TLS is used the URI chosen to place in the Record-Route header  
> field value will need to use a sips: URI or ensure that a sip: URI  
> resolves (per the procedures of [RFC3263]) to using TLS rather than  
> attempting to use the deprecated  "transport=tls" URI parameter.


Let's pull some smaller pieces out of this:

> When TLS is used the URI chosen to place in the Record-Route header  
> field value will need to use a sips: URI.


"chosen to place" is clumsy. Let's substitute "placed", add a comma  
for the subordination,  and see if it gets better.

> When TLS is used, the URI placed in the Record-Route header field  
> value will need to use a sips: URI.

What does it mean for a URI to use a URI? I think perhaps it needs to  
BE a SIPS URI.

Rewritten:

> When TLS is used, the URI placed in the Record-Route header field  
> value will need to be a sips: URI.

Now lets look at the other half of that clause:

> When TLS is used, the URI placed in the Record-Route header field  
> value will need to ensure that a sip: URI resolves (per the  
> procedures of [RFC3263]) to using TLS rather than attempting to use  
> the deprecated  "transport=tls" URI parameter.


How does a URI ensure that a ensure something about a URI?

Maybe you mean:

> When TLS is used, the URI placed in the Record-Route header field  
> value MUST resolve (per the procedures of [RFC3263]) to using TLS  
> rather than attempting to use the deprecated  "transport=tls" URI  
> parameter.


I'm still stumbling over "resolve to using TLS rather than attempting  
to use the deprecated  "transport=tls" URI parameter." So let's just  
lop off the sentence from "rather than . . ." and move that out.

What's wrong with

> When TLS is used, the URI placed in the Record-Route header field  
> value MUST resolve (per the procedures of [RFC3263]) to using TLS .

Is "resolve to using TLS" the same thing as "resolve using TLS"? I  
think they're subtly different, but the complexity still scares me. I  
think we're trying to say that the application of RFC 3263 procedures  
to the URI must result in a next-hop that is reachable only by TLS.

So let's try a rewrite based on these assumptions. How about this:

> When TLS is used, the URI placed in the Record-Route header field  
> MUST encode a next-hop that will be reached using TLS. There are two  
> ways for this to work. The first way is for the URI placed in the  
> Record-Route to be a sips: URI. The second is for the URI placed in  
> the Record-Route to be constructed such that application of  RFC  
> 3263 resolution procedures
> to that URI produces a result reachable only using TLS. Proxies  
> compliant with this specification MUST NOT use a "transport=tls"  
> parameter on the URI placed in the Record-Route because the  
> "transport=tls" usage was deprecated by RFC 3261.

Does that mean the same thing to you as the original megasentence did?

--
Dean




From john.elwell@siemens-enterprise.com  Thu Aug 13 03:47:29 2009
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Date: Thu, 13 Aug 2009 11:42:51 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Cc: sip@ietf.org, BONNAERENS Ben <ben.bonnaerens@alcatel-lucent.be>
Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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OK for me too.=20

John

> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]=20
> Sent: 12 August 2009 21:15
> To: Dean Willis
> Cc: Elwell, John; sip@ietf.org; BONNAERENS Ben
> Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>=20
> I skipped straight to the end (will come back and read how you got =20
> there later).
> But yes, I think the rewrite you have at the end says the the same =20
> thing.
>=20
> RjS
> On Aug 12, 2009, at 2:42 PM, Dean Willis wrote:
>=20
> >
> > On Aug 11, 2009, at 8:35 PM, Robert Sparks wrote:
> >
> >> John -
> >>
> >> Would this substitution address your concern?
> >>
> >>    When TLS is used on the transport on either side of the proxy =20
> >> the URI
> >>    chosen to place in the Record-Route header field value =20
> >> reflecting the
> >>    interface using TLS will need to use a sips: URI or=20
> ensure that =20
> >> a sip:
> >>    URI resolves (per the procedures of [RFC3263]) to using=20
> TLS rather
> >>    than attempting to use the deprecated  "transport=3Dtls" URI =20
> >> parameter.
> >>    See [RFC3261] Section 26.2.2 and [I-D.ietf-sip-sips]=20
> Section 3.1.4
> >>    for more discussion
> >>
> >
> >
> > That makes my head hurt!
> >
> > Compound sentences should make sense even with clauses redacted.
> >
> > Let's play some games with this mind twister:
> >
> >
> > First I'll try to scope the sub-clauses:
> >
> >> (When TLS is used (on the transport (on either side of the=20
> proxy))) =20
> >> the URI (chosen to place (in the Record-Route header field value =20
> >> (reflecting the interface using TLS))) will need to use a=20
> sips: URI =20
> >> or ensure that a sip: URI resolves (per the procedures of =20
> >> [RFC3263]) to using TLS (rather than attempting to use the =20
> >> deprecated  "transport=3Dtls" URI parameter).
> >>
> >
> > One reduction to see if it makes sense:
> >
> >> When TLS is used the URI chosen to place in the=20
> Record-Route header =20
> >> field value will need to use a sips: URI or ensure that a=20
> sip: URI =20
> >> resolves (per the procedures of [RFC3263]) to using TLS=20
> rather than =20
> >> attempting to use the deprecated  "transport=3Dtls" URI parameter.
> >
> >
> > Let's pull some smaller pieces out of this:
> >
> >> When TLS is used the URI chosen to place in the=20
> Record-Route header =20
> >> field value will need to use a sips: URI.
> >
> >
> > "chosen to place" is clumsy. Let's substitute "placed", add=20
> a comma =20
> > for the subordination,  and see if it gets better.
> >
> >> When TLS is used, the URI placed in the Record-Route header field =20
> >> value will need to use a sips: URI.
> >
> > What does it mean for a URI to use a URI? I think perhaps it needs =20
> > to BE a SIPS URI.
> >
> > Rewritten:
> >
> >> When TLS is used, the URI placed in the Record-Route header field =20
> >> value will need to be a sips: URI.
> >
> > Now lets look at the other half of that clause:
> >
> >> When TLS is used, the URI placed in the Record-Route header field =20
> >> value will need to ensure that a sip: URI resolves (per the =20
> >> procedures of [RFC3263]) to using TLS rather than=20
> attempting to use =20
> >> the deprecated  "transport=3Dtls" URI parameter.
> >
> >
> > How does a URI ensure that a ensure something about a URI?
> >
> > Maybe you mean:
> >
> >> When TLS is used, the URI placed in the Record-Route header field =20
> >> value MUST resolve (per the procedures of [RFC3263]) to using TLS =20
> >> rather than attempting to use the deprecated  "transport=3Dtls" URI =
=20
> >> parameter.
> >
> >
> > I'm still stumbling over "resolve to using TLS rather than =20
> > attempting to use the deprecated  "transport=3Dtls" URI=20
> parameter." So =20
> > let's just lop off the sentence from "rather than . . ." and move =20
> > that out.
> >
> > What's wrong with
> >
> >> When TLS is used, the URI placed in the Record-Route header field =20
> >> value MUST resolve (per the procedures of [RFC3263]) to using TLS .
> >
> > Is "resolve to using TLS" the same thing as "resolve using TLS"? I =20
> > think they're subtly different, but the complexity still=20
> scares me. =20
> > I think we're trying to say that the application of RFC 3263 =20
> > procedures to the URI must result in a next-hop that is reachable =20
> > only by TLS.
> >
> > So let's try a rewrite based on these assumptions. How about this:
> >
> >> When TLS is used, the URI placed in the Record-Route header field =20
> >> MUST encode a next-hop that will be reached using TLS. There are =20
> >> two ways for this to work. The first way is for the URI placed in =20
> >> the Record-Route to be a sips: URI. The second is for the URI =20
> >> placed in the Record-Route to be constructed such that=20
> application =20
> >> of  RFC 3263 resolution procedures
> >> to that URI produces a result reachable only using TLS. Proxies =20
> >> compliant with this specification MUST NOT use a "transport=3Dtls"  =

> >> parameter on the URI placed in the Record-Route because the =20
> >> "transport=3Dtls" usage was deprecated by RFC 3261.
> >
> > Does that mean the same thing to you as the original=20
> megasentence did?
> >
> > --
> > Dean
> >
> >
> >
>=20
>=20

From ben.bonnaerens@alcatel-lucent.be  Thu Aug 13 05:16:11 2009
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From: "BONNAERENS Ben" <ben.bonnaerens@alcatel-lucent.be>
To: "Dean Willis" <dean.willis@softarmor.com>, "Robert Sparks" <rjsparks@nostrum.com>
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Hello Dean,

Sorry to make your head hurt ;-)

To me the -09 sentence was clear but I do think you have a point on the
compound sentences.

Still I would like to comment on the "only" in following sentence of the
text you proposed:
"The second is for the URI placed in the Record-Route to be constructed
such that application of  RFC =20
3263 resolution procedures to that URI produces a result reachable only
using TLS."

Why should that URI _only_ be reachable via TLS?
The DNS for that URI can be populated with NAPTR records
for multiple transports with TLS as preferred I assume.

I would also replace "sips: URI" by "SIPS URI" to be consistent across
the document.

Resulting in following updated text:
"When TLS is used on the transport on either side of the proxy,
the URI placed in the Record-Route header field  MUST encode=20
a next-hop that will be reached using TLS. There are two =20
ways for this to work. The first way is for the URI placed in the =20
Record-Route to be a SIPS URI.The second is for the URI placed in =20
the Record-Route to be constructed such that application of =20
[RFC3263] resolution procedures to that URI results in TLS being
selected.=20
Proxies compliant with this specification MUST NOT use a =
"transport=3Dtls"

parameter on the URI placed in the Record-Route because the =20
"transport=3Dtls" usage was deprecated by RFC 3261."

Can we agree on this text?

Best regards & Thanks,

Ben.

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: woensdag 12 augustus 2009 22:17
> To: Robert Sparks
> Cc: Elwell, John; sip@ietf.org; BONNAERENS Ben
> Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>=20
>=20
> On Aug 11, 2009, at 8:35 PM, Robert Sparks wrote:
>=20
> > John -
> >
> > Would this substitution address your concern?
> >
> >    When TLS is used on the transport on either side of the=20
> proxy the=20
> > URI
> >    chosen to place in the Record-Route header field value=20
> reflecting=20
> > the
> >    interface using TLS will need to use a sips: URI or ensure that a
> > sip:
> >    URI resolves (per the procedures of [RFC3263]) to using=20
> TLS rather
> >    than attempting to use the deprecated  "transport=3Dtls" URI=20
> > parameter.
> >    See [RFC3261] Section 26.2.2 and [I-D.ietf-sip-sips]=20
> Section 3.1.4
> >    for more discussion
> >
>=20
>=20
> That makes my head hurt!
>=20
> Compound sentences should make sense even with clauses redacted.
>=20
> Let's play some games with this mind twister:
>=20
>=20
> First I'll try to scope the sub-clauses:
>=20
> > (When TLS is used (on the transport (on either side of the proxy)))=20
> > the URI (chosen to place (in the Record-Route header field value=20
> > (reflecting the interface using TLS))) will need to use a=20
> sips: URI or=20
> > ensure that a sip: URI resolves (per the procedures of [RFC3263])
> > to using TLS (rather than attempting to use the deprecated  =20
> > "transport=3Dtls" URI parameter).
> >
>=20
> One reduction to see if it makes sense:
>=20
> > When TLS is used the URI chosen to place in the Record-Route header=20
> > field value will need to use a sips: URI or ensure that a sip: URI=20
> > resolves (per the procedures of [RFC3263]) to using TLS rather than=20
> > attempting to use the deprecated  "transport=3Dtls" URI parameter.
>=20
>=20
> Let's pull some smaller pieces out of this:
>=20
> > When TLS is used the URI chosen to place in the Record-Route header=20
> > field value will need to use a sips: URI.
>=20
>=20
> "chosen to place" is clumsy. Let's substitute "placed", add a comma =20
> for the subordination,  and see if it gets better.
>=20
> > When TLS is used, the URI placed in the Record-Route header field =20
> > value will need to use a sips: URI.
>=20
> What does it mean for a URI to use a URI? I think perhaps it=20
> needs to =20
> BE a SIPS URI.
>=20
> Rewritten:
>=20
> > When TLS is used, the URI placed in the Record-Route header field =20
> > value will need to be a sips: URI.
>=20
> Now lets look at the other half of that clause:
>=20
> > When TLS is used, the URI placed in the Record-Route header field =20
> > value will need to ensure that a sip: URI resolves (per the =20
> > procedures of [RFC3263]) to using TLS rather than=20
> attempting to use =20
> > the deprecated  "transport=3Dtls" URI parameter.
>=20
>=20
> How does a URI ensure that a ensure something about a URI?
>=20
> Maybe you mean:
>=20
> > When TLS is used, the URI placed in the Record-Route header field =20
> > value MUST resolve (per the procedures of [RFC3263]) to using TLS =20
> > rather than attempting to use the deprecated  "transport=3Dtls" URI  =

> > parameter.
>=20
>=20
> I'm still stumbling over "resolve to using TLS rather than=20
> attempting =20
> to use the deprecated  "transport=3Dtls" URI parameter." So let's just =
=20
> lop off the sentence from "rather than . . ." and move that out.
>=20
> What's wrong with
>=20
> > When TLS is used, the URI placed in the Record-Route header field =20
> > value MUST resolve (per the procedures of [RFC3263]) to using TLS .
>=20
> Is "resolve to using TLS" the same thing as "resolve using TLS"? I =20
> think they're subtly different, but the complexity still=20
> scares me. I =20
> think we're trying to say that the application of RFC 3263=20
> procedures =20
> to the URI must result in a next-hop that is reachable only by TLS.
>=20
> So let's try a rewrite based on these assumptions. How about this:
>=20
> > When TLS is used, the URI placed in the Record-Route header field =20
> > MUST encode a next-hop that will be reached using TLS.=20
> There are two =20
> > ways for this to work. The first way is for the URI placed in the =20
> > Record-Route to be a sips: URI. The second is for the URI=20
> placed in =20
> > the Record-Route to be constructed such that application of  RFC =20
> > 3263 resolution procedures
> > to that URI produces a result reachable only using TLS. Proxies =20
> > compliant with this specification MUST NOT use a "transport=3Dtls" =20
> > parameter on the URI placed in the Record-Route because the =20
> > "transport=3Dtls" usage was deprecated by RFC 3261.
>=20
> Does that mean the same thing to you as the original megasentence did?
>=20
> --
> Dean
>=20
>=20
>=20
>=20

From dean.willis@softarmor.com  Thu Aug 13 14:15:15 2009
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On Aug 13, 2009, at 6:46 AM, BONNAERENS Ben wrote:

> Hello Dean,
>
> Sorry to make your head hurt ;-)
>
> To me the -09 sentence was clear but I do think you have a point on  
> the
> compound sentences.
>
> Still I would like to comment on the "only" in following sentence of  
> the
> text you proposed:
> "The second is for the URI placed in the Record-Route to be  
> constructed
> such that application of  RFC
> 3263 resolution procedures to that URI produces a result reachable  
> only
> using TLS."
>
> Why should that URI _only_ be reachable via TLS?
> The DNS for that URI can be populated with NAPTR records
> for multiple transports with TLS as preferred I assume.
>


Dunno WHY it should say that; that's what I thought I read in the  
compound sentence I was trying to straighten out.

Hmm. Could it be that since SIPS (draft-ietf-siip-sips) requires e2e  
TLS even more strongly than did 3261 for original sips URIs. So if a  
URI used in a record-route resolves both TLS and non-TLS, there's a  
possibility that the return stroke might not use TLS, thereby breaking  
the e2e TLS requirement. But your proposed wording change seems to  
have the same effect.


> I would also replace "sips: URI" by "SIPS URI" to be consistent across
> the document.
>

That makes sense; I  responded in the wording I thought Robert had  
used. But consistency with the document would be preferred.  Your  
usage is also consistent with draft-ietf-sip-sips.

> Resulting in following updated text:
> "When TLS is used on the transport on either side of the proxy,
> the URI placed in the Record-Route header field  MUST encode
> a next-hop that will be reached using TLS. There are two
> ways for this to work. The first way is for the URI placed in the
> Record-Route to be a SIPS URI.The second is for the URI placed in
> the Record-Route to be constructed such that application of
> [RFC3263] resolution procedures to that URI results in TLS being
> selected.
> Proxies compliant with this specification MUST NOT use a  
> "transport=tls"
>
> parameter on the URI placed in the Record-Route because the
> "transport=tls" usage was deprecated by RFC 3261."
>
> Can we agree on this text?


Looks OK to me.

--
Dean


From gshaham@juniper.net  Fri Aug 14 07:30:38 2009
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References: <8144761F31F48D43AD53D09F5350E38004C00CE5@FRVELSMBS22.ad2.ad.alcatel.com>A<0D5F89FAC29E2C41B98A6A762007F5D002393EA0@GBNTHT12009MSX.gb002.siemens.net><8144761F31F48D43AD53D09F5350E38004C01534@FRVELSMBS22.ad2.ad.alcatel.com><0D5F89FAC29E2C41B98A6A762007F5D0023D22FA@GBNTHT12009MSX.gb002.siemens.net><8144761F31F48D43AD53D09F5350E38004C6042C@FRVELSMBS22.ad2.ad.alcatel.com><0D5F89FAC29E2C41B98A6A762007F5D0023D2478@GBNTHT12009MSX.gb002.siemens.net><C7293435-6D29-46EC-82C6-27F86754B3FF@nostrum.com><00CBDB5D-5BD3-437E-9369-FDB2587FF039@softarmor.com><8144761F31F48D43AD53D09F5350E38004C60C04@FRVELSMBS22.ad2.ad.alcatel.com> <BF11270C-1BFD-401D-846D-4903FDCA3B17@softarmor.com>
From: Gilad Shaham <gshaham@juniper.net>
To: "Dean Willis" <dean.willis@softarmor.com>, "BONNAERENS Ben" <ben.bonnaerens@alcatel-lucent.be>
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Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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Hi Ben,

I have also read the draft and everything was clear to me and seemed
easy to read until I reached 6.1 and 6.2.

One suggestion I have to make the text more readable is to separate the
problem description from the solution. Meaning today it's:

1. Problem 1 & actions
2. Problem 2 & actions
3. Problem 3 & actions

I'm proposing to change it to
l. Problem 1
2. Problem 2
3. Problem 3

Describe each problem/scenario without describing the solution and give
examples if needed.

Following that provide all the recommended actions.

Implementers should have an easy way to see and understand all the
changes they require to conform to the RFC. For example, In RFC 4320
it's easy to follow through because it describes the changes very simply
(the problems are described in more detail in RFC 4321). I'm not
suggesting changing this draft into 2 RFCs, just asking to focus on the
problems and then focus on the actions.

In any case maybe there's a different solution for this and maybe it's
just me, but I fear that if the text would not be clear enough then many
vendors would have a hard time assessing the effort needed to implement
it and thus the adoption rate of this spec would be affected.

Final note - section 7 has a summary that may or may not provide all
necessary actions, but I simply had a lot to go through in section 6
that I almost never reached section 7.

Thanks,
Gilad

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Dean Willis
Sent: Friday, August 14, 2009 12:12 AM
To: BONNAERENS Ben
Cc: sip@ietf.org
Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes


On Aug 13, 2009, at 6:46 AM, BONNAERENS Ben wrote:

> Hello Dean,
>
> Sorry to make your head hurt ;-)
>
> To me the -09 sentence was clear but I do think you have a point on =20
> the
> compound sentences.
>
> Still I would like to comment on the "only" in following sentence of =20
> the
> text you proposed:
> "The second is for the URI placed in the Record-Route to be =20
> constructed
> such that application of  RFC
> 3263 resolution procedures to that URI produces a result reachable =20
> only
> using TLS."
>
> Why should that URI _only_ be reachable via TLS?
> The DNS for that URI can be populated with NAPTR records
> for multiple transports with TLS as preferred I assume.
>


Dunno WHY it should say that; that's what I thought I read in the =20
compound sentence I was trying to straighten out.

Hmm. Could it be that since SIPS (draft-ietf-siip-sips) requires e2e =20
TLS even more strongly than did 3261 for original sips URIs. So if a =20
URI used in a record-route resolves both TLS and non-TLS, there's a =20
possibility that the return stroke might not use TLS, thereby breaking =20
the e2e TLS requirement. But your proposed wording change seems to =20
have the same effect.


> I would also replace "sips: URI" by "SIPS URI" to be consistent across
> the document.
>

That makes sense; I  responded in the wording I thought Robert had =20
used. But consistency with the document would be preferred.  Your =20
usage is also consistent with draft-ietf-sip-sips.

> Resulting in following updated text:
> "When TLS is used on the transport on either side of the proxy,
> the URI placed in the Record-Route header field  MUST encode
> a next-hop that will be reached using TLS. There are two
> ways for this to work. The first way is for the URI placed in the
> Record-Route to be a SIPS URI.The second is for the URI placed in
> the Record-Route to be constructed such that application of
> [RFC3263] resolution procedures to that URI results in TLS being
> selected.
> Proxies compliant with this specification MUST NOT use a =20
> "transport=3Dtls"
>
> parameter on the URI placed in the Record-Route because the
> "transport=3Dtls" usage was deprecated by RFC 3261."
>
> Can we agree on this text?


Looks OK to me.

--
Dean

_______________________________________________
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: "Olle E. Johansson" <oej@edvina.net>
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Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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13 aug 2009 kl. 23.12 skrev Dean Willis:
>
>
> Hmm. Could it be that since SIPS (draft-ietf-siip-sips) requires e2e  
> TLS even more strongly than did 3261 for original sips URIs. So if a  
> URI used in a record-route resolves both TLS and non-TLS, there's a  
> possibility that the return stroke might not use TLS, thereby  
> breaking the e2e TLS requirement. But your proposed wording change  
> seems to have the same effect.
>


I did not read the draft as SIPS requiring e2e TLS. We still use  
proxys and have clear communication within the proxy. E2E to me is UAC  
-> UAS with encryption ALL the way and no middleman.

Did I misunderstand?

/O

From rjsparks@nostrum.com  Fri Aug 14 08:44:53 2009
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Cc: sip@ietf.org, BONNAERENS Ben <ben.bonnaerens@alcatel-lucent.be>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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Hi Gilad -

This document is already very far along the process, we are doing fine- 
tuning based on IESG review
and while I truly appreciate insight that will make a document more  
readable, it would not be appropriate
to restructure the text at this time.

I hope more reviewers of future documents show the same kind of  
concern for the implementer you
exhibit here.

RjS

On Aug 14, 2009, at 9:28 AM, Gilad Shaham wrote:

> Hi Ben,
>
> I have also read the draft and everything was clear to me and seemed
> easy to read until I reached 6.1 and 6.2.
>
> One suggestion I have to make the text more readable is to separate  
> the
> problem description from the solution. Meaning today it's:
>
> 1. Problem 1 & actions
> 2. Problem 2 & actions
> 3. Problem 3 & actions
>
> I'm proposing to change it to
> l. Problem 1
> 2. Problem 2
> 3. Problem 3
>
> Describe each problem/scenario without describing the solution and  
> give
> examples if needed.
>
> Following that provide all the recommended actions.
>
> Implementers should have an easy way to see and understand all the
> changes they require to conform to the RFC. For example, In RFC 4320
> it's easy to follow through because it describes the changes very  
> simply
> (the problems are described in more detail in RFC 4321). I'm not
> suggesting changing this draft into 2 RFCs, just asking to focus on  
> the
> problems and then focus on the actions.
>
> In any case maybe there's a different solution for this and maybe it's
> just me, but I fear that if the text would not be clear enough then  
> many
> vendors would have a hard time assessing the effort needed to  
> implement
> it and thus the adoption rate of this spec would be affected.
>
> Final note - section 7 has a summary that may or may not provide all
> necessary actions, but I simply had a lot to go through in section 6
> that I almost never reached section 7.
>
> Thanks,
> Gilad
>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Dean Willis
> Sent: Friday, August 14, 2009 12:12 AM
> To: BONNAERENS Ben
> Cc: sip@ietf.org
> Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
>
>
> On Aug 13, 2009, at 6:46 AM, BONNAERENS Ben wrote:
>
>> Hello Dean,
>>
>> Sorry to make your head hurt ;-)
>>
>> To me the -09 sentence was clear but I do think you have a point on
>> the
>> compound sentences.
>>
>> Still I would like to comment on the "only" in following sentence of
>> the
>> text you proposed:
>> "The second is for the URI placed in the Record-Route to be
>> constructed
>> such that application of  RFC
>> 3263 resolution procedures to that URI produces a result reachable
>> only
>> using TLS."
>>
>> Why should that URI _only_ be reachable via TLS?
>> The DNS for that URI can be populated with NAPTR records
>> for multiple transports with TLS as preferred I assume.
>>
>
>
> Dunno WHY it should say that; that's what I thought I read in the
> compound sentence I was trying to straighten out.
>
> Hmm. Could it be that since SIPS (draft-ietf-siip-sips) requires e2e
> TLS even more strongly than did 3261 for original sips URIs. So if a
> URI used in a record-route resolves both TLS and non-TLS, there's a
> possibility that the return stroke might not use TLS, thereby breaking
> the e2e TLS requirement. But your proposed wording change seems to
> have the same effect.
>
>
>> I would also replace "sips: URI" by "SIPS URI" to be consistent  
>> across
>> the document.
>>
>
> That makes sense; I  responded in the wording I thought Robert had
> used. But consistency with the document would be preferred.  Your
> usage is also consistent with draft-ietf-sip-sips.
>
>> Resulting in following updated text:
>> "When TLS is used on the transport on either side of the proxy,
>> the URI placed in the Record-Route header field  MUST encode
>> a next-hop that will be reached using TLS. There are two
>> ways for this to work. The first way is for the URI placed in the
>> Record-Route to be a SIPS URI.The second is for the URI placed in
>> the Record-Route to be constructed such that application of
>> [RFC3263] resolution procedures to that URI results in TLS being
>> selected.
>> Proxies compliant with this specification MUST NOT use a
>> "transport=tls"
>>
>> parameter on the URI placed in the Record-Route because the
>> "transport=tls" usage was deprecated by RFC 3261."
>>
>> Can we agree on this text?
>
>
> Looks OK to me.
>
> --
> Dean
>
> _______________________________________________
> 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
> _______________________________________________
> 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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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-10.txt
	Pages           : 26
	Date            : 2009-08-14

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 not possible
to make one header have the characteristics of both interfaces 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-10.txt

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From dean.willis@softarmor.com  Sun Aug 16 19:32:22 2009
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Cc: sip@ietf.org, BONNAERENS Ben <ben.bonnaerens@alcatel-lucent.be>
Subject: Re: [Sip] draft-ietf-sip-record-route-fix-08 Changes
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On Aug 14, 2009, at 9:31 AM, Olle E. Johansson wrote:

>
> 13 aug 2009 kl. 23.12 skrev Dean Willis:
>>
>>
>> Hmm. Could it be that since SIPS (draft-ietf-siip-sips) requires  
>> e2e TLS even more strongly than did 3261 for original sips URIs. So  
>> if a URI used in a record-route resolves both TLS and non-TLS,  
>> there's a possibility that the return stroke might not use TLS,  
>> thereby breaking the e2e TLS requirement. But your proposed wording  
>> change seems to have the same effect.
>>
>
>
> I did not read the draft as SIPS requiring e2e TLS. We still use  
> proxys and have clear communication within the proxy. E2E to me is  
> UAC -> UAS with encryption ALL the way and no middleman.
>
> Did I misunderstand?


It kind of  on what you mean by "end to end". Draft-ietf-sip-sips  
requires that TLS be used on every hop whenever a SIPS  URI is used,  
whereas RFC 3261 allowed an exemption for the "last hop". In the  
language of the discussion at the time, this was considered "end to  
end TLS" even though it wasn't "UA to UA TLS".

Under 3261, one could legitimately double record route at a proxy  
performing contact routing with a first route entry having a SIPS URI  
and a second having a SIP URI.

With draft-ietf-sip-sips, that behavior is forbidden if the result  
would be to fail to use TLS on the route-segment indicated with a SIPS  
URI.

The record-route-fix document is essentially promoting a very special  
exception by noting that it is possible to construct a SIP URI such  
that resolution of the SIP URI using the procedures of RFC 3263 would  
guarantee that TLS is used, even though the URI itself is SIP instead  
of SIPS.

Consequently, the working I suggested was intended to convey the idea  
that the "special SIP URI" MUST resolve only to TLS and MUST NOT  
resolve to a hop that would not use TLS, since the only case in which  
we RFC 3261 would have allowed us to not use TLS was deprecated by  
draft-ietf-sip-sips.

--
Dean




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Hi All=20

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

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

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

Please review and comment

Regards=20
Ranjit=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>New version of =
"draft-avasasarala-dispatch-comm-diversion-info" draft submitted</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3527" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2>Hi All</FONT>=20
</DIV>
<P><FONT face=3DArial color=3D#0000ff size=3D2>We have submitted an =
updated version of=20
draft-avasasarala-dispatch-comm-diversion-info </FONT></P>
<P><FONT face=3DArial size=3D2><FONT color=3D#0000ff>It can be accessed =
at: </FONT><A=20
href=3D"http://www.ietf.org/internet-drafts/draft-avasarala-dispatch-comm=
-div-notification-01.txt"=20
target=3D_top rel=3Dnofollow jQuery1250492662043=3D"3"><SPAN><FONT=20
face=3D"Times New Roman" color=3D#0000ff=20
size=3D3>http://www.ietf.org/internet-drafts/draft-avasarala-dispatch-com=
m-div-notification-01.txt</FONT></SPAN></A></FONT></P>
<P><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2>This draft =
proposes a SIP=20
Event package for Communication Diversions Notification Information and =
conforms=20
to procedures and schema described in 3GPP TS 24.604.<SPAN=20
class=3D061545906-17082009>&nbsp;</SPAN></FONT></FONT></FONT></P>
<P><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D061545906-17082009>Please review and=20
comment</SPAN></FONT></FONT></FONT></P>
<P><FONT face=3DArial color=3D#0000ff size=3D2>Regards</FONT> <BR><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Ranjit</FONT> </P></BODY></HTML>

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From wwwrun@core3.amsl.com  Mon Aug 17 11:43:12 2009
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From: The IESG <iesg-secretary@ietf.org>
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Cc: sip mailing list <sip@ietf.org>, sip chair <sip-chairs@tools.ietf.org>, Internet Architecture Board <iab@iab.org>, RFC Editor <rfc-editor@rfc-editor.org>
Subject: [Sip] Protocol Action: 'An Extensible Markup Language (XML) Configuration Access Protocol (XCAP) Diff Event Package' to Proposed Standard
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The IESG has approved the following document:

- 'An Extensible Markup Language (XML) Configuration Access Protocol 
   (XCAP) Diff Event Package '
   <draft-ietf-sip-xcapevent-08.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-xcapevent-08.txt

Technical Summary

The XCAP (XML Configuration Access Protocol, RFC 4825) protocol
provides for change control on structured data. This specification
provides a SIP Event Paackage (RFC 3325) that can be used to monitor
changes to an XCAP document. Change notifications may include
differential changes expressed using the XCAP-Diff format described in
draft-ietf-simple-xcap-diff. This specification also provides for
initial synchronization between a stored version of an XCAP document
and a remote copy being monitored using the protocol of thsi  
specification.


Working Group Summary

This specification developed out of the larger body of configuration
management work related to SIP (RFC 3261). In particular, it relates
to XCAP (RFC 4825) and the SIP Configuration Framework
(draft-ietf-sipping-config-framework). The content of this
specification was split off from the configuration framework document
about two years ago, and has progressed independently.

Document Quality

There are existing implementations of earlier versions of the
protocol, and a significant number of vendors have committed to
implementation of the specification in the context of the Open Mobile
Alliances's Push-to-Talk Over Cellularspecification. Adam Roach
provided specific guidance and review on the SIP Event Package aspects
of this specification.


Personnel

The Proto Shepherd is Dean Willis.


RFC Editor Note

(Relevant to version -08)

These nits have been identified during /post IESG review:

1) Section 4.4 first paragraph second sentence:
   OLD:
    Only a simple subset of that format is required, a flat list of XCAP
R-URIs.
                                                                        
^^^^^^
   NEW:
    Only a simple subset of that format is required, a flat list of XCAP
request URIs.

2) Section 4.4 last paragraph, last sentence:
   OLD: (<service> element)
   NEW: (where a <service> element is selected)

3) Section 4.7, 5th paragraph, last sentence:
   OLD: In a corner case,
   NEW: In a corner case (such as when the XML prolog changes),

4) Section 4.7, 12th paragraph (end of page 10), 3rd sentence:
   OLD: list of XCAP R-URIs
   NEW: list of XCAP request URIs

5) Section 4.7, last paragraph, last sentence: (double not)
   OLD: does not not have
                 ^^^
   NEW: does not have

6) Section 5 last paragraph: (Replace <services> with <service> twice)
  OLD:
   Note that the resource-list "index" document included only the new
   ETag value, as the document existed during the subscription time.  In
   the "pidf-manipulation" collection, there is only a single document
   for which the user has read privilege.  The <services> element exists
                                                       ^
   within the rls-services "index" document and its content is shown.
   Note also that the <services> element was located using the Default
                              ^
   Document Namespace (no prefix in XCAP Node Selector value) although
   it has an "s" prefix in the source document. 

  New:
   Note that the resource-list "index" document included only the new
   ETag value, as the document existed during the subscription time.  In
   the "pidf-manipulation" collection, there is only a single document
   for which the user has read privilege.  The <service> element exists
   within the rls-services "index" document and its content is shown.
   Note also that the <service> element was located using the Default
   Document Namespace (no prefix in XCAP Node Selector value) although
   it has an "s" prefix in the source document.

7) Appendix A - First paragraph: 
   OLD: the SIP R-URIs of 
   NEW: the SIP request URIs of


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Thread-Topic: [Sip] New version of"draft-avasasarala-dispatch-comm-diversion-info" draft submitted
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From: Gilad Shaham <gshaham@juniper.net>
To: "Avasarala Ranjit-A20990" <ranjit@motorola.com>, <dispatch@ietf.org>, <sip@ietf.org>
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Subject: Re: [Sip] New version of"draft-avasasarala-dispatch-comm-diversion-info" draft submitted
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Hi,

=20

See some comments

=20

Page 5:

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

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

=20

Page 12:

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

=20

Page 14:

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

=20

Page 16:

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

Should be

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

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

=20

Page 21:

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

=20

Thanks

Gilad

=20

________________________________

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

=20

Hi All=20

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

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

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

Please review and comment

Regards=20
Ranjit=20


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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

<div>

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

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

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

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

</div>

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

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

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

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

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

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

<p><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial;color:blue'>Regards</span></font> <br>
<font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial;color:blue'>Ranjit</span></font> <o:p></o:p></p>

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From wwwrun@core3.amsl.com  Tue Aug 18 10:47:53 2009
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Date: Tue, 18 Aug 2009 10:47:53 -0700 (PDT)
Cc: sip mailing list <sip@ietf.org>, sip chair <sip-chairs@tools.ietf.org>, Internet Architecture Board <iab@iab.org>, RFC Editor <rfc-editor@rfc-editor.org>
Subject: [Sip] Protocol Action: 'Addressing Record-Route issues in the Session Initiation Protocol (SIP)' to Proposed Standard
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The IESG has approved the following document:

- 'Addressing Record-Route issues in the Session Initiation Protocol 
   (SIP) '
   <draft-ietf-sip-record-route-fix-10.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-record-route-fix-10.txt

Technical Summary

  Multi-homed SIP proxies (having two or more interfaces)
  raise special requirements when handling SIP requests that
  have different ingress and egress interfaces.  One
  technique that has been widely deployed is to insert a
  single Record-Route header describing the egress interface.
  When another request in the  same dialog then traverses the
  proxy in a reverse direction, the proxy must edit the
  headers to reflect the new egress interface. This document
  formalizes another technique that has also been widely
  deployed and found to be simpler and more effective.  Using
  this technique, the proxy inserts two record-route headers,
  one for the ingress interface and one for the egress
  interface. This keeps the route symmetric, such that it can
  be traversed in the reverse direction without requiring
  examination and editing by the proxy, thereby reducing
  system complexity and processing load in the proxy.

Working Group Summary

  There is very strong consensus around the recommendations
  in this document. The working group spent significant time
  debating whether to produce these recommendations as a set
  of corrections to RFC3261 or as a BCP with the latter result.

Document Quality

   The practices documented here are significantly deployed
   and have been tested at many interoperability test events.   

Personnel

   Nils Ohlmeier is the Document Shepherd.
   Robert Sparks is the responsible Area Director.


From sunil.bhagat@wipro.com  Mon Aug 24 02:57:20 2009
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Hi All,

 

I have a small query related to HOLD request being sent out.

 

For a hold request, can gateway respond with different media
capabilities in 200 OK response?

 

Is the HOLD treated as any normal Re-INVITE?

 

Regards,

Sunil.

 


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10=
.0pt;
font-family:Arial'>Hi All,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10=
.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:10=
.0pt;
font-family:Arial'>I have a small query related to HOLD request being sent o=
ut.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10=
.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:10=
.0pt;
font-family:Arial'>For a hold request, can gateway respond with different me=
dia
capabilities in 200 OK response?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10=
.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:10=
.0pt;
font-family:Arial'>Is the HOLD treated as any normal Re-INVITE?<o:p></o:p></=
span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10=
.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:10=
.0pt;
font-family:Arial'>Regards,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:10=
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Hi All,

=20

I have a small query related to HOLD request being sent out.

=20

For a hold request, can gateway respond with different media
capabilities in 200 OK response?

=20

Is the HOLD treated as any normal Re-INVITE?

=20

Regards,

Sunil.

=20


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
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sent out.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
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style=3D'font-size:10.0pt;
font-family:Arial'>For a hold request, can gateway respond with =
different media
capabilities in 200 OK response?<o:p></o:p></span></font></p>

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

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style=3D'font-size:10.0pt;
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Re-INVITE?<o:p></o:p></span></font></p>

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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
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no it should not. For a Hold request, you do not send any media
capabilities. U only change the existing media description lines to
indicate HOLD.
=20

Regards=20
Ranjit=20

=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
sunil.bhagat@wipro.com
Sent: Monday, August 24, 2009 3:25 PM
To: sip@ietf.org; sip-implementors@lists.cs.columbia.edu;
sip@core3.amsl.com
Subject: [Sip] 200 OK response for hold with different media
capabilities



Hi All,

=20

I have a small query related to HOLD request being sent out.

=20

For a hold request, can gateway respond with different media
capabilities in 200 OK response?

=20

Is the HOLD treated as any normal Re-INVITE?

=20

Regards,

Sunil.

=20

Please do not print this email unless it is absolutely necessary.=20

The information contained in this electronic message and any attachments
to this message are intended for the exclusive use of the addressee(s)
and may contain proprietary, confidential or privileged information. If
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<DIV dir=3Dltr align=3Dleft><SPAN class=3D832170311-24082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>no it should not. For a Hold request, you do =
not send any=20
media capabilities. U only change the existing media description lines =
to=20
indicate HOLD.</FONT></SPAN></DIV>
<DIV>&nbsp;</DIV><!-- Converted from text/rtf format -->
<P><SPAN lang=3Den-us><FONT face=3DArial color=3D#0000ff =
size=3D2>Regards</FONT></SPAN>=20
<BR><SPAN lang=3Den-us><FONT face=3DArial color=3D#0000ff =
size=3D2>Ranjit</FONT></SPAN>=20
</P>
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<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=20
</B>sunil.bhagat@wipro.com<BR><B>Sent:</B> Monday, August 24, 2009 3:25=20
PM<BR><B>To:</B> sip@ietf.org; sip-implementors@lists.cs.columbia.edu;=20
sip@core3.amsl.com<BR><B>Subject:</B> [Sip] 200 OK response for hold =
with=20
different media capabilities<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Hi =
All,<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">I have a small query =
related to HOLD=20
request being sent out.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">For a hold request, can =
gateway=20
respond with different media capabilities in 200 OK=20
response?<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Is the HOLD treated as any =
normal=20
Re-INVITE?<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Regards,<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: <sunil.bhagat@wipro.com>, <sip@ietf.org>, <sip-implementors@lists.cs.columbia.edu>, <sip@core3.amsl.com>
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no it should not. For a Hold request, you do not send any media
capabilities. U only change the existing media description lines to
indicate HOLD.
=20

Regards=20
Ranjit=20

=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
sunil.bhagat@wipro.com
Sent: Monday, August 24, 2009 3:25 PM
To: sip@ietf.org; sip-implementors@lists.cs.columbia.edu;
sip@core3.amsl.com
Subject: [Sip] 200 OK response for hold with different media
capabilities



Hi All,

=20

I have a small query related to HOLD request being sent out.

=20

For a hold request, can gateway respond with different media
capabilities in 200 OK response?

=20

Is the HOLD treated as any normal Re-INVITE?

=20

Regards,

Sunil.

=20

Please do not print this email unless it is absolutely necessary.=20

The information contained in this electronic message and any attachments
to this message are intended for the exclusive use of the addressee(s)
and may contain proprietary, confidential or privileged information. If
you are not the intended recipient, you should not disseminate,
distribute or copy this e-mail. Please notify the sender immediately and
destroy all copies of this message and any attachments.=20

WARNING: Computer viruses can be transmitted via email. The recipient
should check this email and any attachments for the presence of viruses.
The company accepts no liability for any damage caused by any virus
transmitted by this email.=20

www.wipro.com=20


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D832170311-24082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>no it should not. For a Hold request, you do =
not send any=20
media capabilities. U only change the existing media description lines =
to=20
indicate HOLD.</FONT></SPAN></DIV>
<DIV>&nbsp;</DIV><!-- Converted from text/rtf format -->
<P><SPAN lang=3Den-us><FONT face=3DArial color=3D#0000ff =
size=3D2>Regards</FONT></SPAN>=20
<BR><SPAN lang=3Den-us><FONT face=3DArial color=3D#0000ff =
size=3D2>Ranjit</FONT></SPAN>=20
</P>
<DIV>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of=20
</B>sunil.bhagat@wipro.com<BR><B>Sent:</B> Monday, August 24, 2009 3:25=20
PM<BR><B>To:</B> sip@ietf.org; sip-implementors@lists.cs.columbia.edu;=20
sip@core3.amsl.com<BR><B>Subject:</B> [Sip] 200 OK response for hold =
with=20
different media capabilities<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Hi =
All,<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">I have a small query =
related to HOLD=20
request being sent out.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">For a hold request, can =
gateway=20
respond with different media capabilities in 200 OK=20
response?<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Is the HOLD treated as any =
normal=20
Re-INVITE?<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
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From vikram.chhibber@gmail.com  Mon Aug 24 11:04:57 2009
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Date: Mon, 24 Aug 2009 11:05:01 -0700
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From: Vikram Chhibber <vikram.chhibber@gmail.com>
To: sunil.bhagat@wipro.com
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Subject: Re: [Sip] 200 OK response for hold with different media capabilities
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The gateway can respond with different media capabilities in 200 OK
response for offer-less INVITE not for offer
that we view as for hold.
The gateway should create response as per rules defined in RFC 3264.
As far as SIP signalling is concerned, hold INVITE request is treated
as a regular re-INVITE request.

On Mon, Aug 24, 2009 at 2:55 AM, <sunil.bhagat@wipro.com> wrote:
> Hi All,
>
>
>
> I have a small query related to HOLD request being sent out.
>
>
>
> For a hold request, can gateway respond with different media capabilities=
 in
> 200 OK response?
>
>
>
> Is the HOLD treated as any normal Re-INVITE?
>
>
>
> Regards,
>
> Sunil.
>
>
>
> Please do not print this email unless it is absolutely necessary.
>
> The information contained in this electronic message and any attachments =
to
> this message are intended for the exclusive use of the addressee(s) and m=
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>
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>
> _______________________________________________
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From vikram.chhibber@gmail.com  Mon Aug 24 11:04:59 2009
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Subject: Re: [Sip] 200 OK response for hold with different media capabilities
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The gateway can respond with different media capabilities in 200 OK
response for offer-less INVITE not for offer
that we view as for hold.
The gateway should create response as per rules defined in RFC 3264.
As far as SIP signalling is concerned, hold INVITE request is treated
as a regular re-INVITE request.

On Mon, Aug 24, 2009 at 2:55 AM, <sunil.bhagat@wipro.com> wrote:
> Hi All,
>
>
>
> I have a small query related to HOLD request being sent out.
>
>
>
> For a hold request, can gateway respond with different media capabilities=
 in
> 200 OK response?
>
>
>
> Is the HOLD treated as any normal Re-INVITE?
>
>
>
> Regards,
>
> Sunil.
>
>
>
> Please do not print this email unless it is absolutely necessary.
>
> The information contained in this electronic message and any attachments =
to
> this message are intended for the exclusive use of the addressee(s) and m=
ay
> contain proprietary, confidential or privileged information. If you are n=
ot
> the intended recipient, you should not disseminate, distribute or copy th=
is
> e-mail. Please notify the sender immediately and destroy all copies of th=
is
> message and any attachments.
>
> WARNING: Computer viruses can be transmitted via email. The recipient sho=
uld
> check this email and any attachments for the presence of viruses. The
> company accepts no liability for any damage caused by any virus transmitt=
ed
> by this email.
>
> www.wipro.com
>
> _______________________________________________
> Sip mailing list =A0https://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
>

From pkyzivat@cisco.com  Mon Aug 24 11:45:31 2009
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Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold with	different media capabilities
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Avasarala Ranjit-A20990 wrote:
> no it should not. For a Hold request, you do not send any media
> capabilities. U only change the existing media description lines to
> indicate HOLD.

I beg to differ.

Technically there is no HOLD request.
All there is is an offer that offers to change a media attribute, from 
sendrecv to sendonly or inactive, or from sendonly or recvonly to inactive.

The response may change anything that is permitted to be changed in the 
answer, given the offer. So it may not add m-lines, but it can certainly 
change media addresses, and within limits may change media formats (codecs).

	Thanks,
	Paul

> Regards 
> Ranjit 
> 
>  
> 
> ________________________________
> 
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> sunil.bhagat@wipro.com
> Sent: Monday, August 24, 2009 3:25 PM
> To: sip@ietf.org; sip-implementors@lists.cs.columbia.edu;
> sip@core3.amsl.com
> Subject: [Sip] 200 OK response for hold with different media
> capabilities
> 
> 
> 
> Hi All,
> 
>  
> 
> I have a small query related to HOLD request being sent out.
> 
>  
> 
> For a hold request, can gateway respond with different media
> capabilities in 200 OK response?
> 
>  
> 
> Is the HOLD treated as any normal Re-INVITE?
> 
>  
> 
> Regards,
> 
> Sunil.
> 
>  
> 
> Please do not print this email unless it is absolutely necessary. 
> 
> The information contained in this electronic message and any attachments
> to this message are intended for the exclusive use of the addressee(s)
> and may contain proprietary, confidential or privileged information. If
> you are not the intended recipient, you should not disseminate,
> distribute or copy this e-mail. Please notify the sender immediately and
> destroy all copies of this message and any attachments. 
> 
> WARNING: Computer viruses can be transmitted via email. The recipient
> should check this email and any attachments for the presence of viruses.
> The company accepts no liability for any damage caused by any virus
> transmitted by this email. 
> 
> www.wipro.com 
> 
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@lists.cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
> 

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Date: Mon, 24 Aug 2009 14:36:37 -0400
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Cc: sip@ietf.org, sunil.bhagat@wipro.com, sip@core3.amsl.com, sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold with	different media capabilities
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Avasarala Ranjit-A20990 wrote:
> no it should not. For a Hold request, you do not send any media
> capabilities. U only change the existing media description lines to
> indicate HOLD.

I beg to differ.

Technically there is no HOLD request.
All there is is an offer that offers to change a media attribute, from 
sendrecv to sendonly or inactive, or from sendonly or recvonly to inactive.

The response may change anything that is permitted to be changed in the 
answer, given the offer. So it may not add m-lines, but it can certainly 
change media addresses, and within limits may change media formats (codecs).

	Thanks,
	Paul

> Regards 
> Ranjit 
> 
>  
> 
> ________________________________
> 
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> sunil.bhagat@wipro.com
> Sent: Monday, August 24, 2009 3:25 PM
> To: sip@ietf.org; sip-implementors@lists.cs.columbia.edu;
> sip@core3.amsl.com
> Subject: [Sip] 200 OK response for hold with different media
> capabilities
> 
> 
> 
> Hi All,
> 
>  
> 
> I have a small query related to HOLD request being sent out.
> 
>  
> 
> For a hold request, can gateway respond with different media
> capabilities in 200 OK response?
> 
>  
> 
> Is the HOLD treated as any normal Re-INVITE?
> 
>  
> 
> Regards,
> 
> Sunil.
> 
>  
> 
> Please do not print this email unless it is absolutely necessary. 
> 
> The information contained in this electronic message and any attachments
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> 
> www.wipro.com 
> 
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> 

From ranjit@motorola.com  Mon Aug 24 18:43:08 2009
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Thread-Topic: [Sip-implementors] [Sip] 200 OK response for hold with	different media capabilities
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References: <E45B59E8A17574419150DA25A71AED070699DDC1@BLR-EC-MBX02.wipro.com> <750BBC72E178114F9DC4872EBFF29A5B08E15FA3@ZMY16EXM66.ds.mot.com> <4A92DDB5.90400@cisco.com>
From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Cc: sip@ietf.org, sunil.bhagat@wipro.com, sip@core3.amsl.com, sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold with	different media capabilities
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Hi Paul

The HOLD request is nothing but a re INVITE request indicating HOLD.

Could you give a use case where such a thing may happen? I have not come
across any scenario where in the process of putting remote party on
HOLD, media parameters like IP/codec getting changed.  If this happens,
then it cannot be considered a HOLD request.

So how do u distinguish a HOLD request from a non HOLD one if in both
the cases the media parameters (other than a=3D.....) change?


Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Tuesday, August 25, 2009 12:07 AM
To: Avasarala Ranjit-A20990
Cc: sunil.bhagat@wipro.com; sip@ietf.org;
sip-implementors@lists.cs.columbia.edu; sip@core3.amsl.com
Subject: Re: [Sip-implementors] [Sip] 200 OK response for hold with
different media capabilities



Avasarala Ranjit-A20990 wrote:
> no it should not. For a Hold request, you do not send any media=20
> capabilities. U only change the existing media description lines to=20
> indicate HOLD.

I beg to differ.

Technically there is no HOLD request.
All there is is an offer that offers to change a media attribute, from
sendrecv to sendonly or inactive, or from sendonly or recvonly to
inactive.

The response may change anything that is permitted to be changed in the
answer, given the offer. So it may not add m-lines, but it can certainly
change media addresses, and within limits may change media formats
(codecs).

	Thanks,
	Paul

> Regards
> Ranjit
>=20
> =20
>=20
> ________________________________
>=20
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of=20
> sunil.bhagat@wipro.com
> Sent: Monday, August 24, 2009 3:25 PM
> To: sip@ietf.org; sip-implementors@lists.cs.columbia.edu;
> sip@core3.amsl.com
> Subject: [Sip] 200 OK response for hold with different media=20
> capabilities
>=20
>=20
>=20
> Hi All,
>=20
> =20
>=20
> I have a small query related to HOLD request being sent out.
>=20
> =20
>=20
> For a hold request, can gateway respond with different media=20
> capabilities in 200 OK response?
>=20
> =20
>=20
> Is the HOLD treated as any normal Re-INVITE?
>=20
> =20
>=20
> Regards,
>=20
> Sunil.
>=20
> =20
>=20
> Please do not print this email unless it is absolutely necessary.=20
>=20
> The information contained in this electronic message and any=20
> attachments to this message are intended for the exclusive use of the=20
> addressee(s) and may contain proprietary, confidential or privileged=20
> information. If you are not the intended recipient, you should not=20
> disseminate, distribute or copy this e-mail. Please notify the sender=20
> immediately and destroy all copies of this message and any
attachments.
>=20
> WARNING: Computer viruses can be transmitted via email. The recipient=20
> should check this email and any attachments for the presence of
viruses.
> The company accepts no liability for any damage caused by any virus=20
> transmitted by this email.
>=20
> www.wipro.com
>=20
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@lists.cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>=20

From ranjit@motorola.com  Mon Aug 24 18:43:15 2009
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Thread-Topic: [Sip-implementors] [Sip] 200 OK response for hold with	different media capabilities
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References: <E45B59E8A17574419150DA25A71AED070699DDC1@BLR-EC-MBX02.wipro.com> <750BBC72E178114F9DC4872EBFF29A5B08E15FA3@ZMY16EXM66.ds.mot.com> <4A92DDB5.90400@cisco.com>
From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold with	different media capabilities
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Hi Paul

The HOLD request is nothing but a re INVITE request indicating HOLD.

Could you give a use case where such a thing may happen? I have not come
across any scenario where in the process of putting remote party on
HOLD, media parameters like IP/codec getting changed.  If this happens,
then it cannot be considered a HOLD request.

So how do u distinguish a HOLD request from a non HOLD one if in both
the cases the media parameters (other than a=3D.....) change?


Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Tuesday, August 25, 2009 12:07 AM
To: Avasarala Ranjit-A20990
Cc: sunil.bhagat@wipro.com; sip@ietf.org;
sip-implementors@lists.cs.columbia.edu; sip@core3.amsl.com
Subject: Re: [Sip-implementors] [Sip] 200 OK response for hold with
different media capabilities



Avasarala Ranjit-A20990 wrote:
> no it should not. For a Hold request, you do not send any media=20
> capabilities. U only change the existing media description lines to=20
> indicate HOLD.

I beg to differ.

Technically there is no HOLD request.
All there is is an offer that offers to change a media attribute, from
sendrecv to sendonly or inactive, or from sendonly or recvonly to
inactive.

The response may change anything that is permitted to be changed in the
answer, given the offer. So it may not add m-lines, but it can certainly
change media addresses, and within limits may change media formats
(codecs).

	Thanks,
	Paul

> Regards
> Ranjit
>=20
> =20
>=20
> ________________________________
>=20
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of=20
> sunil.bhagat@wipro.com
> Sent: Monday, August 24, 2009 3:25 PM
> To: sip@ietf.org; sip-implementors@lists.cs.columbia.edu;
> sip@core3.amsl.com
> Subject: [Sip] 200 OK response for hold with different media=20
> capabilities
>=20
>=20
>=20
> Hi All,
>=20
> =20
>=20
> I have a small query related to HOLD request being sent out.
>=20
> =20
>=20
> For a hold request, can gateway respond with different media=20
> capabilities in 200 OK response?
>=20
> =20
>=20
> Is the HOLD treated as any normal Re-INVITE?
>=20
> =20
>=20
> Regards,
>=20
> Sunil.
>=20
> =20
>=20
> Please do not print this email unless it is absolutely necessary.=20
>=20
> The information contained in this electronic message and any=20
> attachments to this message are intended for the exclusive use of the=20
> addressee(s) and may contain proprietary, confidential or privileged=20
> information. If you are not the intended recipient, you should not=20
> disseminate, distribute or copy this e-mail. Please notify the sender=20
> immediately and destroy all copies of this message and any
attachments.
>=20
> WARNING: Computer viruses can be transmitted via email. The recipient=20
> should check this email and any attachments for the presence of
viruses.
> The company accepts no liability for any damage caused by any virus=20
> transmitted by this email.
>=20
> www.wipro.com
>=20
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@lists.cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>=20

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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Vikram Chhibber" <vikram.chhibber@gmail.com>, <sunil.bhagat@wipro.com>
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Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold withdifferent media capabilities
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Hi Vikram

Gateway can respond with media capabilities to a offer less INVITE if it =
is acting as a 3PCC, not otherwise. =20


Regards
Ranjit

-----Original Message-----
From: sip-implementors-bounces@lists.cs.columbia.edu =
[mailto:sip-implementors-bounces@lists.cs.columbia.edu] On Behalf Of =
Vikram Chhibber
Sent: Monday, August 24, 2009 11:35 PM
To: sunil.bhagat@wipro.com
Cc: sip@ietf.org; sip@core3.amsl.com; =
sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] [Sip] 200 OK response for hold =
withdifferent media capabilities

The gateway can respond with different media capabilities in 200 OK =
response for offer-less INVITE not for offer that we view as for hold.
The gateway should create response as per rules defined in RFC 3264.
As far as SIP signalling is concerned, hold INVITE request is treated as =
a regular re-INVITE request.

On Mon, Aug 24, 2009 at 2:55 AM, <sunil.bhagat@wipro.com> wrote:
> Hi All,
>
>
>
> I have a small query related to HOLD request being sent out.
>
>
>
> For a hold request, can gateway respond with different media=20
> capabilities in 200 OK response?
>
>
>
> Is the HOLD treated as any normal Re-INVITE?
>
>
>
> Regards,
>
> Sunil.
>
>
>
> Please do not print this email unless it is absolutely necessary.
>
> The information contained in this electronic message and any=20
> attachments to this message are intended for the exclusive use of the=20
> addressee(s) and may contain proprietary, confidential or privileged=20
> information. If you are not the intended recipient, you should not=20
> disseminate, distribute or copy this e-mail. Please notify the sender=20
> immediately and destroy all copies of this message and any =
attachments.
>
> WARNING: Computer viruses can be transmitted via email. The recipient=20
> should check this email and any attachments for the presence of=20
> viruses. The company accepts no liability for any damage caused by any =

> virus transmitted by this email.
>
> www.wipro.com
>
> _______________________________________________
> Sip mailing list =A0https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sipping@ietf.org for new developments on the application of sip
>

_______________________________________________
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Thread-Topic: [Sip-implementors] [Sip] 200 OK response for hold withdifferent media capabilities
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References: <E45B59E8A17574419150DA25A71AED070699DDC1@BLR-EC-MBX02.wipro.com> <ac8baa2c0908241105p4e2626f8vbbf37748a208c565@mail.gmail.com>
From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Vikram Chhibber" <vikram.chhibber@gmail.com>, <sunil.bhagat@wipro.com>
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Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold withdifferent media capabilities
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Hi Vikram

Gateway can respond with media capabilities to a offer less INVITE if it =
is acting as a 3PCC, not otherwise. =20


Regards
Ranjit

-----Original Message-----
From: sip-implementors-bounces@lists.cs.columbia.edu =
[mailto:sip-implementors-bounces@lists.cs.columbia.edu] On Behalf Of =
Vikram Chhibber
Sent: Monday, August 24, 2009 11:35 PM
To: sunil.bhagat@wipro.com
Cc: sip@ietf.org; sip@core3.amsl.com; =
sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] [Sip] 200 OK response for hold =
withdifferent media capabilities

The gateway can respond with different media capabilities in 200 OK =
response for offer-less INVITE not for offer that we view as for hold.
The gateway should create response as per rules defined in RFC 3264.
As far as SIP signalling is concerned, hold INVITE request is treated as =
a regular re-INVITE request.

On Mon, Aug 24, 2009 at 2:55 AM, <sunil.bhagat@wipro.com> wrote:
> Hi All,
>
>
>
> I have a small query related to HOLD request being sent out.
>
>
>
> For a hold request, can gateway respond with different media=20
> capabilities in 200 OK response?
>
>
>
> Is the HOLD treated as any normal Re-INVITE?
>
>
>
> Regards,
>
> Sunil.
>
>
>
> Please do not print this email unless it is absolutely necessary.
>
> The information contained in this electronic message and any=20
> attachments to this message are intended for the exclusive use of the=20
> addressee(s) and may contain proprietary, confidential or privileged=20
> information. If you are not the intended recipient, you should not=20
> disseminate, distribute or copy this e-mail. Please notify the sender=20
> immediately and destroy all copies of this message and any =
attachments.
>
> WARNING: Computer viruses can be transmitted via email. The recipient=20
> should check this email and any attachments for the presence of=20
> viruses. The company accepts no liability for any damage caused by any =

> virus transmitted by this email.
>
> www.wipro.com
>
> _______________________________________________
> Sip mailing list =A0https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sipping@ietf.org for new developments on the application of sip
>

_______________________________________________
Sip-implementors mailing list
Sip-implementors@lists.cs.columbia.edu
https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

From abhatnagar192006@gmail.com  Mon Aug 24 19:53:16 2009
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Date: Tue, 25 Aug 2009 02:53:21 +0000
Message-ID: <abcc59880908241953l64c834a0xf0b213ca92fdb22c@mail.gmail.com>
From: aayush bhatnagar <abhatnagar192006@gmail.com>
To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
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Cc: sip@ietf.org, sunil.bhagat@wipro.com, sip@core3.amsl.com, sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold withdifferent media capabilities
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Once the media streams(which are encoded in a set of codecs negotiated
earlier) have requested to be put on hold, then the answer should not
try to change any media attributes or codecs (that were negotiated
earlier).

By providing a different codec set in response to a HOLD request would
be equavalent to modifing the media characteristics of the current
session, not putting the media on hold.


On 25/08/2009, Avasarala Ranjit-A20990 <ranjit@motorola.com> wrote:
> Hi Vikram
>
> Gateway can respond with media capabilities to a offer less INVITE if it =
is
> acting as a 3PCC, not otherwise.
>
>
> Regards
> Ranjit
>
> -----Original Message-----
> From: sip-implementors-bounces@lists.cs.columbia.edu
> [mailto:sip-implementors-bounces@lists.cs.columbia.edu] On Behalf Of Vikr=
am
> Chhibber
> Sent: Monday, August 24, 2009 11:35 PM
> To: sunil.bhagat@wipro.com
> Cc: sip@ietf.org; sip@core3.amsl.com; sip-implementors@lists.cs.columbia.=
edu
> Subject: Re: [Sip-implementors] [Sip] 200 OK response for hold withdiffer=
ent
> media capabilities
>
> The gateway can respond with different media capabilities in 200 OK respo=
nse
> for offer-less INVITE not for offer that we view as for hold.
> The gateway should create response as per rules defined in RFC 3264.
> As far as SIP signalling is concerned, hold INVITE request is treated as =
a
> regular re-INVITE request.
>
> On Mon, Aug 24, 2009 at 2:55 AM, <sunil.bhagat@wipro.com> wrote:
>> Hi All,
>>
>>
>>
>> I have a small query related to HOLD request being sent out.
>>
>>
>>
>> For a hold request, can gateway respond with different media
>> capabilities in 200 OK response?
>>
>>
>>
>> Is the HOLD treated as any normal Re-INVITE?
>>
>>
>>
>> Regards,
>>
>> Sunil.
>>
>>
>>
>> Please do not print this email unless it is absolutely necessary.
>>
>> The information contained in this electronic message and any
>> attachments to this message are intended for the exclusive use of the
>> addressee(s) and may contain proprietary, confidential or privileged
>> information. If you are not the intended recipient, you should not
>> disseminate, distribute or copy this e-mail. Please notify the sender
>> immediately and destroy all copies of this message and any attachments.
>>
>> WARNING: Computer viruses can be transmitted via email. The recipient
>> should check this email and any attachments for the presence of
>> viruses. The company accepts no liability for any damage caused by any
>> virus transmitted by this email.
>>
>> www.wipro.com
>>
>> _______________________________________________
>> Sip mailing list =A0https://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
>>
>
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@lists.cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
> _______________________________________________
> 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
>


--=20
aayush
----------------------------------------------------
"A philosopher always knows what to do...until it happens to him"

From abhatnagar192006@gmail.com  Mon Aug 24 19:53:18 2009
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Date: Tue, 25 Aug 2009 02:53:21 +0000
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From: aayush bhatnagar <abhatnagar192006@gmail.com>
To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
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Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold withdifferent media capabilities
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Once the media streams(which are encoded in a set of codecs negotiated
earlier) have requested to be put on hold, then the answer should not
try to change any media attributes or codecs (that were negotiated
earlier).

By providing a different codec set in response to a HOLD request would
be equavalent to modifing the media characteristics of the current
session, not putting the media on hold.


On 25/08/2009, Avasarala Ranjit-A20990 <ranjit@motorola.com> wrote:
> Hi Vikram
>
> Gateway can respond with media capabilities to a offer less INVITE if it =
is
> acting as a 3PCC, not otherwise.
>
>
> Regards
> Ranjit
>
> -----Original Message-----
> From: sip-implementors-bounces@lists.cs.columbia.edu
> [mailto:sip-implementors-bounces@lists.cs.columbia.edu] On Behalf Of Vikr=
am
> Chhibber
> Sent: Monday, August 24, 2009 11:35 PM
> To: sunil.bhagat@wipro.com
> Cc: sip@ietf.org; sip@core3.amsl.com; sip-implementors@lists.cs.columbia.=
edu
> Subject: Re: [Sip-implementors] [Sip] 200 OK response for hold withdiffer=
ent
> media capabilities
>
> The gateway can respond with different media capabilities in 200 OK respo=
nse
> for offer-less INVITE not for offer that we view as for hold.
> The gateway should create response as per rules defined in RFC 3264.
> As far as SIP signalling is concerned, hold INVITE request is treated as =
a
> regular re-INVITE request.
>
> On Mon, Aug 24, 2009 at 2:55 AM, <sunil.bhagat@wipro.com> wrote:
>> Hi All,
>>
>>
>>
>> I have a small query related to HOLD request being sent out.
>>
>>
>>
>> For a hold request, can gateway respond with different media
>> capabilities in 200 OK response?
>>
>>
>>
>> Is the HOLD treated as any normal Re-INVITE?
>>
>>
>>
>> Regards,
>>
>> Sunil.
>>
>>
>>
>> Please do not print this email unless it is absolutely necessary.
>>
>> The information contained in this electronic message and any
>> attachments to this message are intended for the exclusive use of the
>> addressee(s) and may contain proprietary, confidential or privileged
>> information. If you are not the intended recipient, you should not
>> disseminate, distribute or copy this e-mail. Please notify the sender
>> immediately and destroy all copies of this message and any attachments.
>>
>> WARNING: Computer viruses can be transmitted via email. The recipient
>> should check this email and any attachments for the presence of
>> viruses. The company accepts no liability for any damage caused by any
>> virus transmitted by this email.
>>
>> www.wipro.com
>>
>> _______________________________________________
>> Sip mailing list =A0https://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
>>
>
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@lists.cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
> _______________________________________________
> 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
>


--=20
aayush
----------------------------------------------------
"A philosopher always knows what to do...until it happens to him"

From vikram.chhibber@gmail.com  Mon Aug 24 20:59:56 2009
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References: <E45B59E8A17574419150DA25A71AED070699DDC1@BLR-EC-MBX02.wipro.com> <ac8baa2c0908241105p4e2626f8vbbf37748a208c565@mail.gmail.com> <750BBC72E178114F9DC4872EBFF29A5B08E1606E@ZMY16EXM66.ds.mot.com>
Date: Mon, 24 Aug 2009 20:59:49 -0700
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From: Vikram Chhibber <vikram.chhibber@gmail.com>
To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
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Cc: sip@ietf.org, sunil.bhagat@wipro.com, sip@core3.amsl.com, sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold withdifferent media capabilities
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On Mon, Aug 24, 2009 at 6:44 PM, Avasarala
Ranjit-A20990<ranjit@motorola.com> wrote:
> Hi Vikram
>
> Gateway can respond with media capabilities to a offer less INVITE if it =
is acting as a 3PCC, not otherwise.
>
This is exactly what I wrote in my mail.
I would again emphasize Paul=92s comment. =93Hold=94 is application=92s
interpretation of some values in SDP such as offer with a =3D
sendonly/inactive. There is no such SDP called as hold SDP.
Furthermore,  this offer should be processed as a normal offer as
there could be a change in IP/port, codecs or codecs attributes.
>
> Regards
> Ranjit
>
> -----Original Message-----
> From: sip-implementors-bounces@lists.cs.columbia.edu [mailto:sip-implemen=
tors-bounces@lists.cs.columbia.edu] On Behalf Of Vikram Chhibber
> Sent: Monday, August 24, 2009 11:35 PM
> To: sunil.bhagat@wipro.com
> Cc: sip@ietf.org; sip@core3.amsl.com; sip-implementors@lists.cs.columbia.=
edu
> Subject: Re: [Sip-implementors] [Sip] 200 OK response for hold withdiffer=
ent media capabilities
>
> The gateway can respond with different media capabilities in 200 OK respo=
nse for offer-less INVITE not for offer that we view as for hold.
> The gateway should create response as per rules defined in RFC 3264.
> As far as SIP signalling is concerned, hold INVITE request is treated as =
a regular re-INVITE request.
>
> On Mon, Aug 24, 2009 at 2:55 AM, <sunil.bhagat@wipro.com> wrote:
>> Hi All,
>>
>>
>>
>> I have a small query related to HOLD request being sent out.
>>
>>
>>
>> For a hold request, can gateway respond with different media
>> capabilities in 200 OK response?
>>
>>
>>
>> Is the HOLD treated as any normal Re-INVITE?
>>
>>
>>
>> Regards,
>>
>> Sunil.
>>
>>
>>
>> Please do not print this email unless it is absolutely necessary.
>>
>> The information contained in this electronic message and any
>> attachments to this message are intended for the exclusive use of the
>> addressee(s) and may contain proprietary, confidential or privileged
>> information. If you are not the intended recipient, you should not
>> disseminate, distribute or copy this e-mail. Please notify the sender
>> immediately and destroy all copies of this message and any attachments.
>>
>> WARNING: Computer viruses can be transmitted via email. The recipient
>> should check this email and any attachments for the presence of
>> viruses. The company accepts no liability for any damage caused by any
>> virus transmitted by this email.
>>
>> www.wipro.com
>>
>> _______________________________________________
>> Sip mailing list =A0https://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
>>
>
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@lists.cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>

From vikram.chhibber@gmail.com  Mon Aug 24 21:03:13 2009
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Date: Mon, 24 Aug 2009 21:03:17 -0700
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From: Vikram Chhibber <vikram.chhibber@gmail.com>
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Cc: sip@ietf.org, sunil.bhagat@wipro.com, sip@core3.amsl.com, sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold withdifferent media capabilities
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>> Gateway can respond with media capabilities to a offer less INVITE if it=
 is acting as a 3PCC, not otherwise.
A small correction here. I do not know what Gateway you are talking
about here. For me, a gateway is a SIP end-point and would respond
with its media capabilities for offer-less INVITE.

On Mon, Aug 24, 2009 at 8:59 PM, Vikram
Chhibber<vikram.chhibber@gmail.com> wrote:
> On Mon, Aug 24, 2009 at 6:44 PM, Avasarala
> Ranjit-A20990<ranjit@motorola.com> wrote:
>> Hi Vikram
>>
>> Gateway can respond with media capabilities to a offer less INVITE if it=
 is acting as a 3PCC, not otherwise.
>>
> This is exactly what I wrote in my mail.
> I would again emphasize Paul=92s comment. =93Hold=94 is application=92s
> interpretation of some values in SDP such as offer with a =3D
> sendonly/inactive. There is no such SDP called as hold SDP.
> Furthermore, =A0this offer should be processed as a normal offer as
> there could be a change in IP/port, codecs or codecs attributes.
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: sip-implementors-bounces@lists.cs.columbia.edu [mailto:sip-impleme=
ntors-bounces@lists.cs.columbia.edu] On Behalf Of Vikram Chhibber
>> Sent: Monday, August 24, 2009 11:35 PM
>> To: sunil.bhagat@wipro.com
>> Cc: sip@ietf.org; sip@core3.amsl.com; sip-implementors@lists.cs.columbia=
.edu
>> Subject: Re: [Sip-implementors] [Sip] 200 OK response for hold withdiffe=
rent media capabilities
>>
>> The gateway can respond with different media capabilities in 200 OK resp=
onse for offer-less INVITE not for offer that we view as for hold.
>> The gateway should create response as per rules defined in RFC 3264.
>> As far as SIP signalling is concerned, hold INVITE request is treated as=
 a regular re-INVITE request.
>>
>> On Mon, Aug 24, 2009 at 2:55 AM, <sunil.bhagat@wipro.com> wrote:
>>> Hi All,
>>>
>>>
>>>
>>> I have a small query related to HOLD request being sent out.
>>>
>>>
>>>
>>> For a hold request, can gateway respond with different media
>>> capabilities in 200 OK response?
>>>
>>>
>>>
>>> Is the HOLD treated as any normal Re-INVITE?
>>>
>>>
>>>
>>> Regards,
>>>
>>> Sunil.
>>>
>>>
>>>
>>> Please do not print this email unless it is absolutely necessary.
>>>
>>> The information contained in this electronic message and any
>>> attachments to this message are intended for the exclusive use of the
>>> addressee(s) and may contain proprietary, confidential or privileged
>>> information. If you are not the intended recipient, you should not
>>> disseminate, distribute or copy this e-mail. Please notify the sender
>>> immediately and destroy all copies of this message and any attachments.
>>>
>>> WARNING: Computer viruses can be transmitted via email. The recipient
>>> should check this email and any attachments for the presence of
>>> viruses. The company accepts no liability for any damage caused by any
>>> virus transmitted by this email.
>>>
>>> www.wipro.com
>>>
>>> _______________________________________________
>>> Sip mailing list =A0https://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
>>>
>>
>> _______________________________________________
>> Sip-implementors mailing list
>> Sip-implementors@lists.cs.columbia.edu
>> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>>
>

From vikram.chhibber@gmail.com  Mon Aug 24 21:03:15 2009
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Date: Mon, 24 Aug 2009 21:03:17 -0700
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From: Vikram Chhibber <vikram.chhibber@gmail.com>
To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
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Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold withdifferent media capabilities
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>> Gateway can respond with media capabilities to a offer less INVITE if it=
 is acting as a 3PCC, not otherwise.
A small correction here. I do not know what Gateway you are talking
about here. For me, a gateway is a SIP end-point and would respond
with its media capabilities for offer-less INVITE.

On Mon, Aug 24, 2009 at 8:59 PM, Vikram
Chhibber<vikram.chhibber@gmail.com> wrote:
> On Mon, Aug 24, 2009 at 6:44 PM, Avasarala
> Ranjit-A20990<ranjit@motorola.com> wrote:
>> Hi Vikram
>>
>> Gateway can respond with media capabilities to a offer less INVITE if it=
 is acting as a 3PCC, not otherwise.
>>
> This is exactly what I wrote in my mail.
> I would again emphasize Paul=92s comment. =93Hold=94 is application=92s
> interpretation of some values in SDP such as offer with a =3D
> sendonly/inactive. There is no such SDP called as hold SDP.
> Furthermore, =A0this offer should be processed as a normal offer as
> there could be a change in IP/port, codecs or codecs attributes.
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: sip-implementors-bounces@lists.cs.columbia.edu [mailto:sip-impleme=
ntors-bounces@lists.cs.columbia.edu] On Behalf Of Vikram Chhibber
>> Sent: Monday, August 24, 2009 11:35 PM
>> To: sunil.bhagat@wipro.com
>> Cc: sip@ietf.org; sip@core3.amsl.com; sip-implementors@lists.cs.columbia=
.edu
>> Subject: Re: [Sip-implementors] [Sip] 200 OK response for hold withdiffe=
rent media capabilities
>>
>> The gateway can respond with different media capabilities in 200 OK resp=
onse for offer-less INVITE not for offer that we view as for hold.
>> The gateway should create response as per rules defined in RFC 3264.
>> As far as SIP signalling is concerned, hold INVITE request is treated as=
 a regular re-INVITE request.
>>
>> On Mon, Aug 24, 2009 at 2:55 AM, <sunil.bhagat@wipro.com> wrote:
>>> Hi All,
>>>
>>>
>>>
>>> I have a small query related to HOLD request being sent out.
>>>
>>>
>>>
>>> For a hold request, can gateway respond with different media
>>> capabilities in 200 OK response?
>>>
>>>
>>>
>>> Is the HOLD treated as any normal Re-INVITE?
>>>
>>>
>>>
>>> Regards,
>>>
>>> Sunil.
>>>
>>>
>>>
>>> Please do not print this email unless it is absolutely necessary.
>>>
>>> The information contained in this electronic message and any
>>> attachments to this message are intended for the exclusive use of the
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>>> disseminate, distribute or copy this e-mail. Please notify the sender
>>> immediately and destroy all copies of this message and any attachments.
>>>
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>>> should check this email and any attachments for the presence of
>>> viruses. The company accepts no liability for any damage caused by any
>>> virus transmitted by this email.
>>>
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>>>
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>>> This list is for NEW development of the core SIP Protocol Use
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>>> sipping@ietf.org for new developments on the application of sip
>>>
>>
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>

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Date: Mon, 24 Aug 2009 20:59:49 -0700
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From: Vikram Chhibber <vikram.chhibber@gmail.com>
To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
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Cc: sip@ietf.org, sunil.bhagat@wipro.com, sip@core3.amsl.com, sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold withdifferent media capabilities
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On Mon, Aug 24, 2009 at 6:44 PM, Avasarala
Ranjit-A20990<ranjit@motorola.com> wrote:
> Hi Vikram
>
> Gateway can respond with media capabilities to a offer less INVITE if it =
is acting as a 3PCC, not otherwise.
>
This is exactly what I wrote in my mail.
I would again emphasize Paul=92s comment. =93Hold=94 is application=92s
interpretation of some values in SDP such as offer with a =3D
sendonly/inactive. There is no such SDP called as hold SDP.
Furthermore,  this offer should be processed as a normal offer as
there could be a change in IP/port, codecs or codecs attributes.
>
> Regards
> Ranjit
>
> -----Original Message-----
> From: sip-implementors-bounces@lists.cs.columbia.edu [mailto:sip-implemen=
tors-bounces@lists.cs.columbia.edu] On Behalf Of Vikram Chhibber
> Sent: Monday, August 24, 2009 11:35 PM
> To: sunil.bhagat@wipro.com
> Cc: sip@ietf.org; sip@core3.amsl.com; sip-implementors@lists.cs.columbia.=
edu
> Subject: Re: [Sip-implementors] [Sip] 200 OK response for hold withdiffer=
ent media capabilities
>
> The gateway can respond with different media capabilities in 200 OK respo=
nse for offer-less INVITE not for offer that we view as for hold.
> The gateway should create response as per rules defined in RFC 3264.
> As far as SIP signalling is concerned, hold INVITE request is treated as =
a regular re-INVITE request.
>
> On Mon, Aug 24, 2009 at 2:55 AM, <sunil.bhagat@wipro.com> wrote:
>> Hi All,
>>
>>
>>
>> I have a small query related to HOLD request being sent out.
>>
>>
>>
>> For a hold request, can gateway respond with different media
>> capabilities in 200 OK response?
>>
>>
>>
>> Is the HOLD treated as any normal Re-INVITE?
>>
>>
>>
>> Regards,
>>
>> Sunil.
>>
>>
>>
>> Please do not print this email unless it is absolutely necessary.
>>
>> The information contained in this electronic message and any
>> attachments to this message are intended for the exclusive use of the
>> addressee(s) and may contain proprietary, confidential or privileged
>> information. If you are not the intended recipient, you should not
>> disseminate, distribute or copy this e-mail. Please notify the sender
>> immediately and destroy all copies of this message and any attachments.
>>
>> WARNING: Computer viruses can be transmitted via email. The recipient
>> should check this email and any attachments for the presence of
>> viruses. The company accepts no liability for any damage caused by any
>> virus transmitted by this email.
>>
>> www.wipro.com
>>
>> _______________________________________________
>> Sip mailing list =A0https://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
>>
>
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@lists.cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>

From pkyzivat@cisco.com  Tue Aug 25 08:28:18 2009
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Cc: sunil.bhagat@wipro.com, sip@core3.amsl.com, sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip] [Sip-implementors] 200 OK response for hold with	different media capabilities
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I'm removing the cross posting to the sip list.
Cross posting is unpolite to those who subscribe to both lists.
And this topic is appropriate to sip-implementors.

Avasarala Ranjit-A20990 wrote:
> Hi Paul
> 
> The HOLD request is nothing but a re INVITE request indicating HOLD.

There is no such thing as a reINVITE request that indicates *HOLD*.
There are indications for sendrecv, sendonly, recvonly, and inactive.
They are often used by UAs that have a HOLD feature, but they are also 
used for other purposes unrelated to HOLD features.

When a UAC implements a HOLD feature by sending a reINVITE with 
a=sendonly, the UAS does not know that the reason for this change is 
because of a HOLD feature. It can only take the signaling at face value:

"My peer wants to send me media, but doesn't want to receive it. Given 
that constraint, what do I want to do?"

There can be many answers to that question. There are no rules that say: 
thou shalt not change media attributes in this case.

> Could you give a use case where such a thing may happen? I have not come
> across any scenario where in the process of putting remote party on
> HOLD, media parameters like IP/codec getting changed.  If this happens,
> then it cannot be considered a HOLD request.

Its probably good advice that in general when being presented with an 
offered change you should respond with an answer that makes the minimal 
changes on your part that are consistent with what was offered. But that 
behavior isn't required.

I have heard of implementations that change media port on every 
reINVITE. Doesn't make sense to me, but maybe it did to them, and its 
within the rights of the implementor to do so.

> So how do u distinguish a HOLD request from a non HOLD one if in both
> the cases the media parameters (other than a=.....) change?

As the recipient, you *don't* distinguish a HOLD request from a non HOLD 
one. You simply take things at face value without attempting to ascribe 
a reason for *why* they have happened.

	Thanks,
	Paul

> Regards
> Ranjit
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Tuesday, August 25, 2009 12:07 AM
> To: Avasarala Ranjit-A20990
> Cc: sunil.bhagat@wipro.com; sip@ietf.org;
> sip-implementors@lists.cs.columbia.edu; sip@core3.amsl.com
> Subject: Re: [Sip-implementors] [Sip] 200 OK response for hold with
> different media capabilities
> 
> 
> 
> Avasarala Ranjit-A20990 wrote:
>> no it should not. For a Hold request, you do not send any media 
>> capabilities. U only change the existing media description lines to 
>> indicate HOLD.
> 
> I beg to differ.
> 
> Technically there is no HOLD request.
> All there is is an offer that offers to change a media attribute, from
> sendrecv to sendonly or inactive, or from sendonly or recvonly to
> inactive.
> 
> The response may change anything that is permitted to be changed in the
> answer, given the offer. So it may not add m-lines, but it can certainly
> change media addresses, and within limits may change media formats
> (codecs).
> 
> 	Thanks,
> 	Paul
> 
>> Regards
>> Ranjit
>>
>>  
>>
>> ________________________________
>>
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of 
>> sunil.bhagat@wipro.com
>> Sent: Monday, August 24, 2009 3:25 PM
>> To: sip@ietf.org; sip-implementors@lists.cs.columbia.edu;
>> sip@core3.amsl.com
>> Subject: [Sip] 200 OK response for hold with different media 
>> capabilities
>>
>>
>>
>> Hi All,
>>
>>  
>>
>> I have a small query related to HOLD request being sent out.
>>
>>  
>>
>> For a hold request, can gateway respond with different media 
>> capabilities in 200 OK response?
>>
>>  
>>
>> Is the HOLD treated as any normal Re-INVITE?
>>
>>  
>>
>> Regards,
>>
>> Sunil.
>>
>>  
>>
>> Please do not print this email unless it is absolutely necessary. 
>>
>> The information contained in this electronic message and any 
>> attachments to this message are intended for the exclusive use of the 
>> addressee(s) and may contain proprietary, confidential or privileged 
>> information. If you are not the intended recipient, you should not 
>> disseminate, distribute or copy this e-mail. Please notify the sender 
>> immediately and destroy all copies of this message and any
> attachments.
>> WARNING: Computer viruses can be transmitted via email. The recipient 
>> should check this email and any attachments for the presence of
> viruses.
>> The company accepts no liability for any damage caused by any virus 
>> transmitted by this email.
>>
>> www.wipro.com
>>
>> _______________________________________________
>> Sip-implementors mailing list
>> Sip-implementors@lists.cs.columbia.edu
>> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>>
> 

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Hi
=20
As specified in section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, If
a UA is configured with a outbound-proxy-set with two proxy uris:
"sip:primary.example.com" and "sip:secondary.example.com" And for
example, "sip:primary.example.com" got DNS resolved in to two SRV
records "Host1" and "Host2" and "sip:secondary.example.com" got DNS
resolved in to "Host3" and "Host4", Should UA create four FLOWs, One
with each host OR just two FLOWs, one with each proxy URI(i.e one flow
with either Host1 or Host2 and second flow with Host3 or Host4)?
=20
Please clarify.
=20
Regards
Srikanth

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<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D100403814-27082009>Hi</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D100403814-27082009>As =
specified in=20
section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, If a UA is =
configured with=20
a outbound-proxy-set with two proxy uris: "sip:primary.example.com" and=20
"sip:secondary.example.com" And for example, "sip:primary.example.com" =
got DNS=20
resolved in to two SRV records "Host1" and "Host2" and=20
"sip:secondary.example.com" got DNS resolved in to "Host3" and "Host4", =
Should=20
UA create four FLOWs, One with each host&nbsp;OR just two FLOWs, one =
with each=20
proxy URI(i.e one flow with either Host1 or Host2 and second flow with =
Host3 or=20
Host4)?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D100403814-27082009>Please =

clarify.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D100403814-27082009>Regards</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D100403814-27082009>Srikanth</SPAN></FONT></DIV></BODY></HTML>

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Thread-Topic: [Sip] Question on draft-ietf-sip-outbound-20 draft: multiple flowcreation
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References: <608D7BFF0D57794785CBC2C440BC4FA1058116D7@ZMY16EXM67.ds.mot.com>
From: "Francois Audet" <audet@nortel.com>
To: "Vavilapalli Srikanth-A19563" <srikanthv@motorola.com>, <sip@ietf.org>
Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multiple flowcreation
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You would have 2. One for primary, and one for secondary. For each one =
of those, you would use normal RFC 3263 procedures for determining which =
SRV record to use.


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Vavilapalli Srikanth-A19563
	Sent: Thursday, August 27, 2009 07:57
	To: sip@ietf.org
	Subject: [Sip] Question on draft-ietf-sip-outbound-20 draft: multiple =
flowcreation
=09
=09
	Hi
	=20
	As specified in section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, If =
a UA is configured with a outbound-proxy-set with two proxy uris: =
"sip:primary.example.com" and "sip:secondary.example.com" And for =
example, "sip:primary.example.com" got DNS resolved in to two SRV =
records "Host1" and "Host2" and "sip:secondary.example.com" got DNS =
resolved in to "Host3" and "Host4", Should UA create four FLOWs, One =
with each host OR just two FLOWs, one with each proxy URI(i.e one flow =
with either Host1 or Host2 and second flow with Host3 or Host4)?
	=20
	Please clarify.
	=20
	Regards
	Srikanth


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
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<BODY>
<DIV><SPAN class=3D393564715-27082009><FONT face=3DArial color=3D#800000 =
size=3D2>You=20
would have 2. </FONT></SPAN><SPAN class=3D393564715-27082009><FONT =
face=3DArial=20
color=3D#800000 size=3D2>One for primary, and one for secondary. For =
each one of=20
those, you would use normal RFC 3263 procedures for determining which =
SRV record=20
to use.</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Vavilapalli=20
  Srikanth-A19563<BR><B>Sent:</B> Thursday, August 27, 2009 =
07:57<BR><B>To:</B>=20
  sip@ietf.org<BR><B>Subject:</B> [Sip] Question on =
draft-ietf-sip-outbound-20=20
  draft: multiple flowcreation<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009>Hi</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D100403814-27082009>As =
specified in=20
  section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, If a UA is =
configured=20
  with a outbound-proxy-set with two proxy uris: =
"sip:primary.example.com" and=20
  "sip:secondary.example.com" And for example, "sip:primary.example.com" =
got DNS=20
  resolved in to two SRV records "Host1" and "Host2" and=20
  "sip:secondary.example.com" got DNS resolved in to "Host3" and =
"Host4", Should=20
  UA create four FLOWs, One with each host&nbsp;OR just two FLOWs, one =
with each=20
  proxy URI(i.e one flow with either Host1 or Host2 and second flow with =
Host3=20
  or Host4)?</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D100403814-27082009>Please=20
  clarify.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009>Regards</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  =
class=3D100403814-27082009>Srikanth</SPAN></FONT></DIV></BLOCKQUOTE></BOD=
Y></HTML>

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Thread-Topic: [Sip] Question on draft-ietf-sip-outbound-20 draft: multiple flowcreation
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From: "Vavilapalli Srikanth-A19563" <srikanthv@motorola.com>
To: "Francois Audet" <audet@nortel.com>, <sip@ietf.org>, <sip@core3.amsl.com>,  <sip-implementors@lists.cs.columbia.edu>
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Hi
=20
Thanks for the reply..One more of clarification required..
=20
So If UA wishes to create multiple transport flows with
"sip:primary.example.com"(As specified in
draft-ietf-sip-outbound-20#section-13 to honor RFC3261 requirement for
message size greater than MTU), would it be OK it one FLOW lands on
Host1 and other FLOW lands on Host2?
=20
i.e for example if "sip:primary.example.com" got resolved into (IP1,
Port1, TCP) -> (IP2, Port2, TCP) -> (IP2, Port2, UDP) -> (IP1, Port1,
UDP) in that order, and if UA wises to create both TCP and UDP FLOWs
with "sip:primary.example.com", then it starts creating TCP FLOW with
(IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, UDP) based
on RFC 3263 procedures. Is my understanding correct here?=20
=20
In the above example, if (IP1) is down/crashed for some reasons, then
"sip:primary.example.com", can never use the TCP FLOW for transporting
inbound messages to that UA (Even though the IP1 and IP2 are state
synchronized) until User Agent detects the FLOW failure with (IP1) and
re-create the TCP FLOW with (IP1 or IP2). Is my understanding correct
here?
=20
Please clarify..
=20
Regard
Srikanth

________________________________

From: Francois Audet [mailto:audet@nortel.com]=20
Sent: Thursday, August 27, 2009 9:22 PM
To: Vavilapalli Srikanth-A19563; sip@ietf.org
Subject: RE: [Sip] Question on draft-ietf-sip-outbound-20 draft:
multiple flowcreation


You would have 2. One for primary, and one for secondary. For each one
of those, you would use normal RFC 3263 procedures for determining which
SRV record to use.


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Vavilapalli Srikanth-A19563
	Sent: Thursday, August 27, 2009 07:57
	To: sip@ietf.org
	Subject: [Sip] Question on draft-ietf-sip-outbound-20 draft:
multiple flowcreation
=09
=09
	Hi
	=20
	As specified in section 3.3/3.4 of draft-ietf-sip-outbound-20
draft, If a UA is configured with a outbound-proxy-set with two proxy
uris: "sip:primary.example.com" and "sip:secondary.example.com" And for
example, "sip:primary.example.com" got DNS resolved in to two SRV
records "Host1" and "Host2" and "sip:secondary.example.com" got DNS
resolved in to "Host3" and "Host4", Should UA create four FLOWs, One
with each host OR just two FLOWs, one with each proxy URI(i.e one flow
with either Host1 or Host2 and second flow with Host3 or Host4)?
	=20
	Please clarify.
	=20
	Regards
	Srikanth


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<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>Hi</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>Thanks for the reply..One more of =
clarification=20
required..</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>So If UA wishes to create multiple transport =
flows with=20
"sip:primary.example.com"(As specified in =
draft-ietf-sip-outbound-20#section-13=20
to honor RFC3261 requirement for message size&nbsp;greater =
than&nbsp;MTU), would=20
it be OK it one FLOW lands on Host1 and other FLOW lands on=20
Host2?</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>i.e for example if "sip:primary.example.com" =
got=20
resolved into (IP1, Port1, TCP) -&gt; (IP2, Port2, TCP) -&gt; (IP2, =
Port2, UDP)=20
-&gt; (IP1, Port1, UDP) in that order, and if UA wises to create both =
TCP and=20
UDP FLOWs with "sip:primary.example.com", then it starts creating TCP =
FLOW with=20
(IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, UDP) based =
on RFC=20
3263 procedures. Is my understanding correct here? </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>In the above example, if (IP1) is =
down/crashed for some=20
reasons, then "sip:primary.example.com", can never use the TCP FLOW for=20
transporting inbound messages to that UA (Even though the IP1 and IP2 =
are state=20
synchronized) until User Agent detects the FLOW failure with (IP1) and =
re-create=20
the TCP FLOW with (IP1 or IP2). Is my understanding correct=20
here?</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>Please clarify..</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>Regard</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>Srikanth</SPAN></FONT></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Francois Audet =
[mailto:audet@nortel.com]=20
<BR><B>Sent:</B> Thursday, August 27, 2009 9:22 PM<BR><B>To:</B> =
Vavilapalli=20
Srikanth-A19563; sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Question on=20
draft-ietf-sip-outbound-20 draft: multiple =
flowcreation<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=3D393564715-27082009><FONT face=3DArial color=3D#800000 =
size=3D2>You=20
would have 2. </FONT></SPAN><SPAN class=3D393564715-27082009><FONT =
face=3DArial=20
color=3D#800000 size=3D2>One for primary, and one for secondary. For =
each one of=20
those, you would use normal RFC 3263 procedures for determining which =
SRV record=20
to use.</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Vavilapalli=20
  Srikanth-A19563<BR><B>Sent:</B> Thursday, August 27, 2009 =
07:57<BR><B>To:</B>=20
  sip@ietf.org<BR><B>Subject:</B> [Sip] Question on =
draft-ietf-sip-outbound-20=20
  draft: multiple flowcreation<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009>Hi</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D100403814-27082009>As =
specified in=20
  section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, If a UA is =
configured=20
  with a outbound-proxy-set with two proxy uris: =
"sip:primary.example.com" and=20
  "sip:secondary.example.com" And for example, "sip:primary.example.com" =
got DNS=20
  resolved in to two SRV records "Host1" and "Host2" and=20
  "sip:secondary.example.com" got DNS resolved in to "Host3" and =
"Host4", Should=20
  UA create four FLOWs, One with each host&nbsp;OR just two FLOWs, one =
with each=20
  proxy URI(i.e one flow with either Host1 or Host2 and second flow with =
Host3=20
  or Host4)?</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D100403814-27082009>Please=20
  clarify.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009>Regards</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  =
class=3D100403814-27082009>Srikanth</SPAN></FONT></DIV></BLOCKQUOTE></BOD=
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Thread-Topic: [Sip] Question on draft-ietf-sip-outbound-20 draft: multiple flowcreation
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From: "Vavilapalli Srikanth-A19563" <srikanthv@motorola.com>
To: "Francois Audet" <audet@nortel.com>, <sip@ietf.org>, <sip@core3.amsl.com>,  <sip-implementors@lists.cs.columbia.edu>
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Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multiple flowcreation
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Hi
=20
Thanks for the reply..One more of clarification required..
=20
So If UA wishes to create multiple transport flows with
"sip:primary.example.com"(As specified in
draft-ietf-sip-outbound-20#section-13 to honor RFC3261 requirement for
message size greater than MTU), would it be OK it one FLOW lands on
Host1 and other FLOW lands on Host2?
=20
i.e for example if "sip:primary.example.com" got resolved into (IP1,
Port1, TCP) -> (IP2, Port2, TCP) -> (IP2, Port2, UDP) -> (IP1, Port1,
UDP) in that order, and if UA wises to create both TCP and UDP FLOWs
with "sip:primary.example.com", then it starts creating TCP FLOW with
(IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, UDP) based
on RFC 3263 procedures. Is my understanding correct here?=20
=20
In the above example, if (IP1) is down/crashed for some reasons, then
"sip:primary.example.com", can never use the TCP FLOW for transporting
inbound messages to that UA (Even though the IP1 and IP2 are state
synchronized) until User Agent detects the FLOW failure with (IP1) and
re-create the TCP FLOW with (IP1 or IP2). Is my understanding correct
here?
=20
Please clarify..
=20
Regard
Srikanth

________________________________

From: Francois Audet [mailto:audet@nortel.com]=20
Sent: Thursday, August 27, 2009 9:22 PM
To: Vavilapalli Srikanth-A19563; sip@ietf.org
Subject: RE: [Sip] Question on draft-ietf-sip-outbound-20 draft:
multiple flowcreation


You would have 2. One for primary, and one for secondary. For each one
of those, you would use normal RFC 3263 procedures for determining which
SRV record to use.


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Vavilapalli Srikanth-A19563
	Sent: Thursday, August 27, 2009 07:57
	To: sip@ietf.org
	Subject: [Sip] Question on draft-ietf-sip-outbound-20 draft:
multiple flowcreation
=09
=09
	Hi
	=20
	As specified in section 3.3/3.4 of draft-ietf-sip-outbound-20
draft, If a UA is configured with a outbound-proxy-set with two proxy
uris: "sip:primary.example.com" and "sip:secondary.example.com" And for
example, "sip:primary.example.com" got DNS resolved in to two SRV
records "Host1" and "Host2" and "sip:secondary.example.com" got DNS
resolved in to "Host3" and "Host4", Should UA create four FLOWs, One
with each host OR just two FLOWs, one with each proxy URI(i.e one flow
with either Host1 or Host2 and second flow with Host3 or Host4)?
	=20
	Please clarify.
	=20
	Regards
	Srikanth


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3603" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>Hi</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>Thanks for the reply..One more of =
clarification=20
required..</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>So If UA wishes to create multiple transport =
flows with=20
"sip:primary.example.com"(As specified in =
draft-ietf-sip-outbound-20#section-13=20
to honor RFC3261 requirement for message size&nbsp;greater =
than&nbsp;MTU), would=20
it be OK it one FLOW lands on Host1 and other FLOW lands on=20
Host2?</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>i.e for example if "sip:primary.example.com" =
got=20
resolved into (IP1, Port1, TCP) -&gt; (IP2, Port2, TCP) -&gt; (IP2, =
Port2, UDP)=20
-&gt; (IP1, Port1, UDP) in that order, and if UA wises to create both =
TCP and=20
UDP FLOWs with "sip:primary.example.com", then it starts creating TCP =
FLOW with=20
(IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, UDP) based =
on RFC=20
3263 procedures. Is my understanding correct here? </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>In the above example, if (IP1) is =
down/crashed for some=20
reasons, then "sip:primary.example.com", can never use the TCP FLOW for=20
transporting inbound messages to that UA (Even though the IP1 and IP2 =
are state=20
synchronized) until User Agent detects the FLOW failure with (IP1) and =
re-create=20
the TCP FLOW with (IP1 or IP2). Is my understanding correct=20
here?</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>Please clarify..</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>Regard</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D791390009-28082009>Srikanth</SPAN></FONT></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Francois Audet =
[mailto:audet@nortel.com]=20
<BR><B>Sent:</B> Thursday, August 27, 2009 9:22 PM<BR><B>To:</B> =
Vavilapalli=20
Srikanth-A19563; sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Question on=20
draft-ietf-sip-outbound-20 draft: multiple =
flowcreation<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=3D393564715-27082009><FONT face=3DArial color=3D#800000 =
size=3D2>You=20
would have 2. </FONT></SPAN><SPAN class=3D393564715-27082009><FONT =
face=3DArial=20
color=3D#800000 size=3D2>One for primary, and one for secondary. For =
each one of=20
those, you would use normal RFC 3263 procedures for determining which =
SRV record=20
to use.</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Vavilapalli=20
  Srikanth-A19563<BR><B>Sent:</B> Thursday, August 27, 2009 =
07:57<BR><B>To:</B>=20
  sip@ietf.org<BR><B>Subject:</B> [Sip] Question on =
draft-ietf-sip-outbound-20=20
  draft: multiple flowcreation<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009>Hi</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D100403814-27082009>As =
specified in=20
  section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, If a UA is =
configured=20
  with a outbound-proxy-set with two proxy uris: =
"sip:primary.example.com" and=20
  "sip:secondary.example.com" And for example, "sip:primary.example.com" =
got DNS=20
  resolved in to two SRV records "Host1" and "Host2" and=20
  "sip:secondary.example.com" got DNS resolved in to "Host3" and =
"Host4", Should=20
  UA create four FLOWs, One with each host&nbsp;OR just two FLOWs, one =
with each=20
  proxy URI(i.e one flow with either Host1 or Host2 and second flow with =
Host3=20
  or Host4)?</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D100403814-27082009>Please=20
  clarify.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D100403814-27082009>Regards</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  =
class=3D100403814-27082009>Srikanth</SPAN></FONT></DIV></BLOCKQUOTE></BOD=
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Thread-Topic: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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From: "Francois Audet" <audet@nortel.com>
To: "Vavilapalli Srikanth-A19563" <srikanthv@motorola.com>, <sip@ietf.org>, <sip@core3.amsl.com>, <sip-implementors@lists.cs.columbia.edu>
Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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I honestly don't know why anybody would want to do both UDP and TCP.
=20
(In other words, if you set up a TCP flow, I don't see a benefit in also =
setting up a UDP flow which you will have to maintain for keep alives).
=20


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Vavilapalli Srikanth-A19563
	Sent: Friday, August 28, 2009 07:46
	To: Audet, Francois (SC100:3055); sip@ietf.org; sip@core3.amsl.com; =
sip-implementors@lists.cs.columbia.edu
	Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: =
multipleflowcreation
=09
=09
	Hi
	=20
	Thanks for the reply..One more of clarification required..
	=20
	So If UA wishes to create multiple transport flows with =
"sip:primary.example.com"(As specified in =
draft-ietf-sip-outbound-20#section-13 to honor RFC3261 requirement for =
message size greater than MTU), would it be OK it one FLOW lands on =
Host1 and other FLOW lands on Host2?
	=20
	i.e for example if "sip:primary.example.com" got resolved into (IP1, =
Port1, TCP) -> (IP2, Port2, TCP) -> (IP2, Port2, UDP) -> (IP1, Port1, =
UDP) in that order, and if UA wises to create both TCP and UDP FLOWs =
with "sip:primary.example.com", then it starts creating TCP FLOW with =
(IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, UDP) based =
on RFC 3263 procedures. Is my understanding correct here?=20
	=20
	In the above example, if (IP1) is down/crashed for some reasons, then =
"sip:primary.example.com", can never use the TCP FLOW for transporting =
inbound messages to that UA (Even though the IP1 and IP2 are state =
synchronized) until User Agent detects the FLOW failure with (IP1) and =
re-create the TCP FLOW with (IP1 or IP2). Is my understanding correct =
here?
	=20
	Please clarify..
	=20
	Regard
	Srikanth

________________________________

	From: Francois Audet [mailto:audet@nortel.com]=20
	Sent: Thursday, August 27, 2009 9:22 PM
	To: Vavilapalli Srikanth-A19563; sip@ietf.org
	Subject: RE: [Sip] Question on draft-ietf-sip-outbound-20 draft: =
multiple flowcreation
=09
=09
	You would have 2. One for primary, and one for secondary. For each one =
of those, you would use normal RFC 3263 procedures for determining which =
SRV record to use.


________________________________

		From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Vavilapalli Srikanth-A19563
		Sent: Thursday, August 27, 2009 07:57
		To: sip@ietf.org
		Subject: [Sip] Question on draft-ietf-sip-outbound-20 draft: multiple =
flowcreation
	=09
	=09
		Hi
		=20
		As specified in section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, =
If a UA is configured with a outbound-proxy-set with two proxy uris: =
"sip:primary.example.com" and "sip:secondary.example.com" And for =
example, "sip:primary.example.com" got DNS resolved in to two SRV =
records "Host1" and "Host2" and "sip:secondary.example.com" got DNS =
resolved in to "Host3" and "Host4", Should UA create four FLOWs, One =
with each host OR just two FLOWs, one with each proxy URI(i.e one flow =
with either Host1 or Host2 and second flow with Host3 or Host4)?
		=20
		Please clarify.
		=20
		Regards
		Srikanth


------_=_NextPart_001_01CA2810.B82636C3
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.3603" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =
size=3D2>I=20
honestly don't know why anybody would want to do both UDP and=20
TCP.</FONT></SPAN></DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =
size=3D2>(In=20
other words, if you set up a TCP flow, I don't see a benefit in also =
setting up=20
a UDP flow which you will have to maintain for keep =
alives).</FONT></SPAN></DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Vavilapalli=20
  Srikanth-A19563<BR><B>Sent:</B> Friday, August 28, 2009 =
07:46<BR><B>To:</B>=20
  Audet, Francois (SC100:3055); sip@ietf.org; sip@core3.amsl.com;=20
  sip-implementors@lists.cs.columbia.edu<BR><B>Subject:</B> Re: [Sip] =
Question=20
  on draft-ietf-sip-outbound-20 draft: =
multipleflowcreation<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Hi</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Thanks for the reply..One more of =
clarification=20
  required..</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>So If UA wishes to create multiple =
transport flows=20
  with "sip:primary.example.com"(As specified in=20
  draft-ietf-sip-outbound-20#section-13 to honor RFC3261 requirement for =
message=20
  size&nbsp;greater than&nbsp;MTU), would it be OK it one FLOW lands on =
Host1=20
  and other FLOW lands on Host2?</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>i.e for example if =
"sip:primary.example.com" got=20
  resolved into (IP1, Port1, TCP) -&gt; (IP2, Port2, TCP) -&gt; (IP2, =
Port2,=20
  UDP) -&gt; (IP1, Port1, UDP) in that order, and if UA wises to create =
both TCP=20
  and UDP FLOWs with "sip:primary.example.com", then it starts creating =
TCP FLOW=20
  with (IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, =
UDP) based=20
  on RFC 3263 procedures. Is my understanding correct here? =
</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>In the above example, if (IP1) is =
down/crashed for=20
  some reasons, then "sip:primary.example.com", can never use the TCP =
FLOW for=20
  transporting inbound messages to that UA (Even though the IP1 and IP2 =
are=20
  state synchronized) until User Agent detects the FLOW failure with =
(IP1) and=20
  re-create the TCP FLOW with (IP1 or IP2). Is my understanding correct=20
  here?</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Please clarify..</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Regard</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Srikanth</SPAN></FONT></DIV><BR>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Francois Audet =
[mailto:audet@nortel.com]=20
  <BR><B>Sent:</B> Thursday, August 27, 2009 9:22 PM<BR><B>To:</B> =
Vavilapalli=20
  Srikanth-A19563; sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Question on =

  draft-ietf-sip-outbound-20 draft: multiple =
flowcreation<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><SPAN class=3D393564715-27082009><FONT face=3DArial =
color=3D#800000 size=3D2>You=20
  would have 2. </FONT></SPAN><SPAN class=3D393564715-27082009><FONT =
face=3DArial=20
  color=3D#800000 size=3D2>One for primary, and one for secondary. For =
each one of=20
  those, you would use normal RFC 3263 procedures for determining which =
SRV=20
  record to use.</FONT></SPAN></DIV><BR>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
    [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Vavilapalli=20
    Srikanth-A19563<BR><B>Sent:</B> Thursday, August 27, 2009=20
    07:57<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] Question =
on=20
    draft-ietf-sip-outbound-20 draft: multiple =
flowcreation<BR></FONT><BR></DIV>
    <DIV></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009>Hi</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN class=3D100403814-27082009>As =
specified in=20
    section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, If a UA is =
configured=20
    with a outbound-proxy-set with two proxy uris: =
"sip:primary.example.com" and=20
    "sip:secondary.example.com" And for example, =
"sip:primary.example.com" got=20
    DNS resolved in to two SRV records "Host1" and "Host2" and=20
    "sip:secondary.example.com" got DNS resolved in to "Host3" and =
"Host4",=20
    Should UA create four FLOWs, One with each host&nbsp;OR just two =
FLOWs, one=20
    with each proxy URI(i.e one flow with either Host1 or Host2 and =
second flow=20
    with Host3 or Host4)?</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D100403814-27082009>Please=20
    clarify.</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009>Regards</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    =
class=3D100403814-27082009>Srikanth</SPAN></FONT></DIV></BLOCKQUOTE></BLO=
CKQUOTE></BODY></HTML>

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Thread-Topic: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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From: "Francois Audet" <audet@nortel.com>
To: "Vavilapalli Srikanth-A19563" <srikanthv@motorola.com>, <sip@ietf.org>, <sip@core3.amsl.com>, <sip-implementors@lists.cs.columbia.edu>
Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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I honestly don't know why anybody would want to do both UDP and TCP.
=20
(In other words, if you set up a TCP flow, I don't see a benefit in also =
setting up a UDP flow which you will have to maintain for keep alives).
=20


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Vavilapalli Srikanth-A19563
	Sent: Friday, August 28, 2009 07:46
	To: Audet, Francois (SC100:3055); sip@ietf.org; sip@core3.amsl.com; =
sip-implementors@lists.cs.columbia.edu
	Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: =
multipleflowcreation
=09
=09
	Hi
	=20
	Thanks for the reply..One more of clarification required..
	=20
	So If UA wishes to create multiple transport flows with =
"sip:primary.example.com"(As specified in =
draft-ietf-sip-outbound-20#section-13 to honor RFC3261 requirement for =
message size greater than MTU), would it be OK it one FLOW lands on =
Host1 and other FLOW lands on Host2?
	=20
	i.e for example if "sip:primary.example.com" got resolved into (IP1, =
Port1, TCP) -> (IP2, Port2, TCP) -> (IP2, Port2, UDP) -> (IP1, Port1, =
UDP) in that order, and if UA wises to create both TCP and UDP FLOWs =
with "sip:primary.example.com", then it starts creating TCP FLOW with =
(IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, UDP) based =
on RFC 3263 procedures. Is my understanding correct here?=20
	=20
	In the above example, if (IP1) is down/crashed for some reasons, then =
"sip:primary.example.com", can never use the TCP FLOW for transporting =
inbound messages to that UA (Even though the IP1 and IP2 are state =
synchronized) until User Agent detects the FLOW failure with (IP1) and =
re-create the TCP FLOW with (IP1 or IP2). Is my understanding correct =
here?
	=20
	Please clarify..
	=20
	Regard
	Srikanth

________________________________

	From: Francois Audet [mailto:audet@nortel.com]=20
	Sent: Thursday, August 27, 2009 9:22 PM
	To: Vavilapalli Srikanth-A19563; sip@ietf.org
	Subject: RE: [Sip] Question on draft-ietf-sip-outbound-20 draft: =
multiple flowcreation
=09
=09
	You would have 2. One for primary, and one for secondary. For each one =
of those, you would use normal RFC 3263 procedures for determining which =
SRV record to use.


________________________________

		From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Vavilapalli Srikanth-A19563
		Sent: Thursday, August 27, 2009 07:57
		To: sip@ietf.org
		Subject: [Sip] Question on draft-ietf-sip-outbound-20 draft: multiple =
flowcreation
	=09
	=09
		Hi
		=20
		As specified in section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, =
If a UA is configured with a outbound-proxy-set with two proxy uris: =
"sip:primary.example.com" and "sip:secondary.example.com" And for =
example, "sip:primary.example.com" got DNS resolved in to two SRV =
records "Host1" and "Host2" and "sip:secondary.example.com" got DNS =
resolved in to "Host3" and "Host4", Should UA create four FLOWs, One =
with each host OR just two FLOWs, one with each proxy URI(i.e one flow =
with either Host1 or Host2 and second flow with Host3 or Host4)?
		=20
		Please clarify.
		=20
		Regards
		Srikanth


------_=_NextPart_001_01CA2810.B82636C3
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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.3603" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =
size=3D2>I=20
honestly don't know why anybody would want to do both UDP and=20
TCP.</FONT></SPAN></DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =
size=3D2>(In=20
other words, if you set up a TCP flow, I don't see a benefit in also =
setting up=20
a UDP flow which you will have to maintain for keep =
alives).</FONT></SPAN></DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Vavilapalli=20
  Srikanth-A19563<BR><B>Sent:</B> Friday, August 28, 2009 =
07:46<BR><B>To:</B>=20
  Audet, Francois (SC100:3055); sip@ietf.org; sip@core3.amsl.com;=20
  sip-implementors@lists.cs.columbia.edu<BR><B>Subject:</B> Re: [Sip] =
Question=20
  on draft-ietf-sip-outbound-20 draft: =
multipleflowcreation<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Hi</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Thanks for the reply..One more of =
clarification=20
  required..</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>So If UA wishes to create multiple =
transport flows=20
  with "sip:primary.example.com"(As specified in=20
  draft-ietf-sip-outbound-20#section-13 to honor RFC3261 requirement for =
message=20
  size&nbsp;greater than&nbsp;MTU), would it be OK it one FLOW lands on =
Host1=20
  and other FLOW lands on Host2?</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>i.e for example if =
"sip:primary.example.com" got=20
  resolved into (IP1, Port1, TCP) -&gt; (IP2, Port2, TCP) -&gt; (IP2, =
Port2,=20
  UDP) -&gt; (IP1, Port1, UDP) in that order, and if UA wises to create =
both TCP=20
  and UDP FLOWs with "sip:primary.example.com", then it starts creating =
TCP FLOW=20
  with (IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, =
UDP) based=20
  on RFC 3263 procedures. Is my understanding correct here? =
</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>In the above example, if (IP1) is =
down/crashed for=20
  some reasons, then "sip:primary.example.com", can never use the TCP =
FLOW for=20
  transporting inbound messages to that UA (Even though the IP1 and IP2 =
are=20
  state synchronized) until User Agent detects the FLOW failure with =
(IP1) and=20
  re-create the TCP FLOW with (IP1 or IP2). Is my understanding correct=20
  here?</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Please clarify..</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Regard</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Srikanth</SPAN></FONT></DIV><BR>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Francois Audet =
[mailto:audet@nortel.com]=20
  <BR><B>Sent:</B> Thursday, August 27, 2009 9:22 PM<BR><B>To:</B> =
Vavilapalli=20
  Srikanth-A19563; sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Question on =

  draft-ietf-sip-outbound-20 draft: multiple =
flowcreation<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><SPAN class=3D393564715-27082009><FONT face=3DArial =
color=3D#800000 size=3D2>You=20
  would have 2. </FONT></SPAN><SPAN class=3D393564715-27082009><FONT =
face=3DArial=20
  color=3D#800000 size=3D2>One for primary, and one for secondary. For =
each one of=20
  those, you would use normal RFC 3263 procedures for determining which =
SRV=20
  record to use.</FONT></SPAN></DIV><BR>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
    [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Vavilapalli=20
    Srikanth-A19563<BR><B>Sent:</B> Thursday, August 27, 2009=20
    07:57<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] Question =
on=20
    draft-ietf-sip-outbound-20 draft: multiple =
flowcreation<BR></FONT><BR></DIV>
    <DIV></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009>Hi</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN class=3D100403814-27082009>As =
specified in=20
    section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, If a UA is =
configured=20
    with a outbound-proxy-set with two proxy uris: =
"sip:primary.example.com" and=20
    "sip:secondary.example.com" And for example, =
"sip:primary.example.com" got=20
    DNS resolved in to two SRV records "Host1" and "Host2" and=20
    "sip:secondary.example.com" got DNS resolved in to "Host3" and =
"Host4",=20
    Should UA create four FLOWs, One with each host&nbsp;OR just two =
FLOWs, one=20
    with each proxy URI(i.e one flow with either Host1 or Host2 and =
second flow=20
    with Host3 or Host4)?</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D100403814-27082009>Please=20
    clarify.</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009>Regards</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    =
class=3D100403814-27082009>Srikanth</SPAN></FONT></DIV></BLOCKQUOTE></BLO=
CKQUOTE></BODY></HTML>

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Thread-Topic: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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From: "Vavilapalli Srikanth-A19563" <srikanthv@motorola.com>
To: "Francois Audet" <audet@nortel.com>, <sip@ietf.org>, <sip@core3.amsl.com>,  <sip-implementors@lists.cs.columbia.edu>
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Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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True, UDP flow may not be required if TCP flow has been setup to route
incoming traffic towards UA. But Won't it be an additional bonus if UA
creates a UDP FLOW in addition to TCP FLOW, so that network can still
reach the UA over the other transport FLOW when one FLOW is down (Unless
if both FLOWs got created with same host and all the FLOWs connected to
that host is down)?=20
=20
Also assume a UA has some local policy to prefer UDP transport over TCP
to transmit the outgoing SIP messages of size less than MTU. In such
scenarios, UA might want to establish both UDP and TCP FLOWs with the
same edge proxy and keep alive both the connections and use the
corresponding FLOW based on DNS lookup preferences and User's local
policy preferences to transport outgoing messages.

________________________________

From: Francois Audet [mailto:audet@nortel.com]=20
Sent: Saturday, August 29, 2009 12:23 AM
To: Vavilapalli Srikanth-A19563; sip@ietf.org; sip@core3.amsl.com;
sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip] Question on draft-ietf-sip-outbound-20 draft:
multipleflowcreation


I honestly don't know why anybody would want to do both UDP and TCP.
=20
(In other words, if you set up a TCP flow, I don't see a benefit in also
setting up a UDP flow which you will have to maintain for keep alives).
=20


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Vavilapalli Srikanth-A19563
	Sent: Friday, August 28, 2009 07:46
	To: Audet, Francois (SC100:3055); sip@ietf.org;
sip@core3.amsl.com; sip-implementors@lists.cs.columbia.edu
	Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft:
multipleflowcreation
=09
=09
	Hi
	=20
	Thanks for the reply..One more of clarification required..
	=20
	So If UA wishes to create multiple transport flows with
"sip:primary.example.com"(As specified in
draft-ietf-sip-outbound-20#section-13 to honor RFC3261 requirement for
message size greater than MTU), would it be OK it one FLOW lands on
Host1 and other FLOW lands on Host2?
	=20
	i.e for example if "sip:primary.example.com" got resolved into
(IP1, Port1, TCP) -> (IP2, Port2, TCP) -> (IP2, Port2, UDP) -> (IP1,
Port1, UDP) in that order, and if UA wises to create both TCP and UDP
FLOWs with "sip:primary.example.com", then it starts creating TCP FLOW
with (IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, UDP)
based on RFC 3263 procedures. Is my understanding correct here?=20
	=20
	In the above example, if (IP1) is down/crashed for some reasons,
then "sip:primary.example.com", can never use the TCP FLOW for
transporting inbound messages to that UA (Even though the IP1 and IP2
are state synchronized) until User Agent detects the FLOW failure with
(IP1) and re-create the TCP FLOW with (IP1 or IP2). Is my understanding
correct here?
	=20
	Please clarify..
	=20
	Regard
	Srikanth

________________________________

	From: Francois Audet [mailto:audet@nortel.com]=20
	Sent: Thursday, August 27, 2009 9:22 PM
	To: Vavilapalli Srikanth-A19563; sip@ietf.org
	Subject: RE: [Sip] Question on draft-ietf-sip-outbound-20 draft:
multiple flowcreation
=09
=09
	You would have 2. One for primary, and one for secondary. For
each one of those, you would use normal RFC 3263 procedures for
determining which SRV record to use.


________________________________

		From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]
On Behalf Of Vavilapalli Srikanth-A19563
		Sent: Thursday, August 27, 2009 07:57
		To: sip@ietf.org
		Subject: [Sip] Question on draft-ietf-sip-outbound-20
draft: multiple flowcreation
	=09
	=09
		Hi
		=20
		As specified in section 3.3/3.4 of
draft-ietf-sip-outbound-20 draft, If a UA is configured with a
outbound-proxy-set with two proxy uris: "sip:primary.example.com" and
"sip:secondary.example.com" And for example, "sip:primary.example.com"
got DNS resolved in to two SRV records "Host1" and "Host2" and
"sip:secondary.example.com" got DNS resolved in to "Host3" and "Host4",
Should UA create four FLOWs, One with each host OR just two FLOWs, one
with each proxy URI(i.e one flow with either Host1 or Host2 and second
flow with Host3 or Host4)?
		=20
		Please clarify.
		=20
		Regards
		Srikanth


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3603" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D105283913-31082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>True, UDP flow may not be required if TCP flow =
has been=20
setup to route incoming traffic towards UA. But Won't it be an =
additional bonus=20
if UA creates a UDP FLOW in addition to TCP FLOW, so that network can =
still=20
reach the UA over the other transport FLOW when one FLOW is down (Unless =
if both=20
FLOWs got created with same host and&nbsp;all the FLOWs connected to=20
that&nbsp;host is down)?&nbsp;</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D105283913-31082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D105283913-31082009></SPAN><SPAN=20
class=3D105283913-31082009><FONT face=3DArial color=3D#0000ff =
size=3D2>Also&nbsp;assume=20
a&nbsp;UA has some local policy to prefer UDP transport&nbsp;over TCP to =

transmit the outgoing&nbsp;SIP messages of&nbsp;size less than MTU. In =
such=20
scenarios, UA&nbsp;might want&nbsp;to establish both UDP and TCP FLOWs =
with the=20
same edge proxy and keep alive both the connections and use the =
corresponding=20
FLOW based on DNS lookup preferences and User's local policy preferences =
to=20
transport outgoing messages.</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Francois Audet =
[mailto:audet@nortel.com]=20
<BR><B>Sent:</B> Saturday, August 29, 2009 12:23 AM<BR><B>To:</B> =
Vavilapalli=20
Srikanth-A19563; sip@ietf.org; sip@core3.amsl.com;=20
sip-implementors@lists.cs.columbia.edu<BR><B>Subject:</B> RE: [Sip] =
Question on=20
draft-ietf-sip-outbound-20 draft: =
multipleflowcreation<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =
size=3D2>I=20
honestly don't know why anybody would want to do both UDP and=20
TCP.</FONT></SPAN></DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =
size=3D2>(In=20
other words, if you set up a TCP flow, I don't see a benefit in also =
setting up=20
a UDP flow which you will have to maintain for keep =
alives).</FONT></SPAN></DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Vavilapalli=20
  Srikanth-A19563<BR><B>Sent:</B> Friday, August 28, 2009 =
07:46<BR><B>To:</B>=20
  Audet, Francois (SC100:3055); sip@ietf.org; sip@core3.amsl.com;=20
  sip-implementors@lists.cs.columbia.edu<BR><B>Subject:</B> Re: [Sip] =
Question=20
  on draft-ietf-sip-outbound-20 draft: =
multipleflowcreation<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Hi</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Thanks for the reply..One more of =
clarification=20
  required..</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>So If UA wishes to create multiple =
transport flows=20
  with "sip:primary.example.com"(As specified in=20
  draft-ietf-sip-outbound-20#section-13 to honor RFC3261 requirement for =
message=20
  size&nbsp;greater than&nbsp;MTU), would it be OK it one FLOW lands on =
Host1=20
  and other FLOW lands on Host2?</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>i.e for example if =
"sip:primary.example.com" got=20
  resolved into (IP1, Port1, TCP) -&gt; (IP2, Port2, TCP) -&gt; (IP2, =
Port2,=20
  UDP) -&gt; (IP1, Port1, UDP) in that order, and if UA wises to create =
both TCP=20
  and UDP FLOWs with "sip:primary.example.com", then it starts creating =
TCP FLOW=20
  with (IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, =
UDP) based=20
  on RFC 3263 procedures. Is my understanding correct here? =
</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>In the above example, if (IP1) is =
down/crashed for=20
  some reasons, then "sip:primary.example.com", can never use the TCP =
FLOW for=20
  transporting inbound messages to that UA (Even though the IP1 and IP2 =
are=20
  state synchronized) until User Agent detects the FLOW failure with =
(IP1) and=20
  re-create the TCP FLOW with (IP1 or IP2). Is my understanding correct=20
  here?</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Please clarify..</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Regard</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Srikanth</SPAN></FONT></DIV><BR>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Francois Audet =
[mailto:audet@nortel.com]=20
  <BR><B>Sent:</B> Thursday, August 27, 2009 9:22 PM<BR><B>To:</B> =
Vavilapalli=20
  Srikanth-A19563; sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Question on =

  draft-ietf-sip-outbound-20 draft: multiple =
flowcreation<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><SPAN class=3D393564715-27082009><FONT face=3DArial =
color=3D#800000 size=3D2>You=20
  would have 2. </FONT></SPAN><SPAN class=3D393564715-27082009><FONT =
face=3DArial=20
  color=3D#800000 size=3D2>One for primary, and one for secondary. For =
each one of=20
  those, you would use normal RFC 3263 procedures for determining which =
SRV=20
  record to use.</FONT></SPAN></DIV><BR>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
    [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Vavilapalli=20
    Srikanth-A19563<BR><B>Sent:</B> Thursday, August 27, 2009=20
    07:57<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] Question =
on=20
    draft-ietf-sip-outbound-20 draft: multiple =
flowcreation<BR></FONT><BR></DIV>
    <DIV></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009>Hi</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN class=3D100403814-27082009>As =
specified in=20
    section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, If a UA is =
configured=20
    with a outbound-proxy-set with two proxy uris: =
"sip:primary.example.com" and=20
    "sip:secondary.example.com" And for example, =
"sip:primary.example.com" got=20
    DNS resolved in to two SRV records "Host1" and "Host2" and=20
    "sip:secondary.example.com" got DNS resolved in to "Host3" and =
"Host4",=20
    Should UA create four FLOWs, One with each host&nbsp;OR just two =
FLOWs, one=20
    with each proxy URI(i.e one flow with either Host1 or Host2 and =
second flow=20
    with Host3 or Host4)?</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D100403814-27082009>Please=20
    clarify.</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009>Regards</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    =
class=3D100403814-27082009>Srikanth</SPAN></FONT></DIV></BLOCKQUOTE></BLO=
CKQUOTE></BODY></HTML>

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Thread-Topic: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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From: "Vavilapalli Srikanth-A19563" <srikanthv@motorola.com>
To: "Francois Audet" <audet@nortel.com>, <sip@ietf.org>, <sip@core3.amsl.com>,  <sip-implementors@lists.cs.columbia.edu>
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Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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True, UDP flow may not be required if TCP flow has been setup to route
incoming traffic towards UA. But Won't it be an additional bonus if UA
creates a UDP FLOW in addition to TCP FLOW, so that network can still
reach the UA over the other transport FLOW when one FLOW is down (Unless
if both FLOWs got created with same host and all the FLOWs connected to
that host is down)?=20
=20
Also assume a UA has some local policy to prefer UDP transport over TCP
to transmit the outgoing SIP messages of size less than MTU. In such
scenarios, UA might want to establish both UDP and TCP FLOWs with the
same edge proxy and keep alive both the connections and use the
corresponding FLOW based on DNS lookup preferences and User's local
policy preferences to transport outgoing messages.

________________________________

From: Francois Audet [mailto:audet@nortel.com]=20
Sent: Saturday, August 29, 2009 12:23 AM
To: Vavilapalli Srikanth-A19563; sip@ietf.org; sip@core3.amsl.com;
sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip] Question on draft-ietf-sip-outbound-20 draft:
multipleflowcreation


I honestly don't know why anybody would want to do both UDP and TCP.
=20
(In other words, if you set up a TCP flow, I don't see a benefit in also
setting up a UDP flow which you will have to maintain for keep alives).
=20


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Vavilapalli Srikanth-A19563
	Sent: Friday, August 28, 2009 07:46
	To: Audet, Francois (SC100:3055); sip@ietf.org;
sip@core3.amsl.com; sip-implementors@lists.cs.columbia.edu
	Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft:
multipleflowcreation
=09
=09
	Hi
	=20
	Thanks for the reply..One more of clarification required..
	=20
	So If UA wishes to create multiple transport flows with
"sip:primary.example.com"(As specified in
draft-ietf-sip-outbound-20#section-13 to honor RFC3261 requirement for
message size greater than MTU), would it be OK it one FLOW lands on
Host1 and other FLOW lands on Host2?
	=20
	i.e for example if "sip:primary.example.com" got resolved into
(IP1, Port1, TCP) -> (IP2, Port2, TCP) -> (IP2, Port2, UDP) -> (IP1,
Port1, UDP) in that order, and if UA wises to create both TCP and UDP
FLOWs with "sip:primary.example.com", then it starts creating TCP FLOW
with (IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, UDP)
based on RFC 3263 procedures. Is my understanding correct here?=20
	=20
	In the above example, if (IP1) is down/crashed for some reasons,
then "sip:primary.example.com", can never use the TCP FLOW for
transporting inbound messages to that UA (Even though the IP1 and IP2
are state synchronized) until User Agent detects the FLOW failure with
(IP1) and re-create the TCP FLOW with (IP1 or IP2). Is my understanding
correct here?
	=20
	Please clarify..
	=20
	Regard
	Srikanth

________________________________

	From: Francois Audet [mailto:audet@nortel.com]=20
	Sent: Thursday, August 27, 2009 9:22 PM
	To: Vavilapalli Srikanth-A19563; sip@ietf.org
	Subject: RE: [Sip] Question on draft-ietf-sip-outbound-20 draft:
multiple flowcreation
=09
=09
	You would have 2. One for primary, and one for secondary. For
each one of those, you would use normal RFC 3263 procedures for
determining which SRV record to use.


________________________________

		From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]
On Behalf Of Vavilapalli Srikanth-A19563
		Sent: Thursday, August 27, 2009 07:57
		To: sip@ietf.org
		Subject: [Sip] Question on draft-ietf-sip-outbound-20
draft: multiple flowcreation
	=09
	=09
		Hi
		=20
		As specified in section 3.3/3.4 of
draft-ietf-sip-outbound-20 draft, If a UA is configured with a
outbound-proxy-set with two proxy uris: "sip:primary.example.com" and
"sip:secondary.example.com" And for example, "sip:primary.example.com"
got DNS resolved in to two SRV records "Host1" and "Host2" and
"sip:secondary.example.com" got DNS resolved in to "Host3" and "Host4",
Should UA create four FLOWs, One with each host OR just two FLOWs, one
with each proxy URI(i.e one flow with either Host1 or Host2 and second
flow with Host3 or Host4)?
		=20
		Please clarify.
		=20
		Regards
		Srikanth


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3603" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D105283913-31082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>True, UDP flow may not be required if TCP flow =
has been=20
setup to route incoming traffic towards UA. But Won't it be an =
additional bonus=20
if UA creates a UDP FLOW in addition to TCP FLOW, so that network can =
still=20
reach the UA over the other transport FLOW when one FLOW is down (Unless =
if both=20
FLOWs got created with same host and&nbsp;all the FLOWs connected to=20
that&nbsp;host is down)?&nbsp;</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D105283913-31082009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D105283913-31082009></SPAN><SPAN=20
class=3D105283913-31082009><FONT face=3DArial color=3D#0000ff =
size=3D2>Also&nbsp;assume=20
a&nbsp;UA has some local policy to prefer UDP transport&nbsp;over TCP to =

transmit the outgoing&nbsp;SIP messages of&nbsp;size less than MTU. In =
such=20
scenarios, UA&nbsp;might want&nbsp;to establish both UDP and TCP FLOWs =
with the=20
same edge proxy and keep alive both the connections and use the =
corresponding=20
FLOW based on DNS lookup preferences and User's local policy preferences =
to=20
transport outgoing messages.</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Francois Audet =
[mailto:audet@nortel.com]=20
<BR><B>Sent:</B> Saturday, August 29, 2009 12:23 AM<BR><B>To:</B> =
Vavilapalli=20
Srikanth-A19563; sip@ietf.org; sip@core3.amsl.com;=20
sip-implementors@lists.cs.columbia.edu<BR><B>Subject:</B> RE: [Sip] =
Question on=20
draft-ietf-sip-outbound-20 draft: =
multipleflowcreation<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =
size=3D2>I=20
honestly don't know why anybody would want to do both UDP and=20
TCP.</FONT></SPAN></DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =
size=3D2>(In=20
other words, if you set up a TCP flow, I don't see a benefit in also =
setting up=20
a UDP flow which you will have to maintain for keep =
alives).</FONT></SPAN></DIV>
<DIV><SPAN class=3D210035118-28082009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Vavilapalli=20
  Srikanth-A19563<BR><B>Sent:</B> Friday, August 28, 2009 =
07:46<BR><B>To:</B>=20
  Audet, Francois (SC100:3055); sip@ietf.org; sip@core3.amsl.com;=20
  sip-implementors@lists.cs.columbia.edu<BR><B>Subject:</B> Re: [Sip] =
Question=20
  on draft-ietf-sip-outbound-20 draft: =
multipleflowcreation<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Hi</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Thanks for the reply..One more of =
clarification=20
  required..</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>So If UA wishes to create multiple =
transport flows=20
  with "sip:primary.example.com"(As specified in=20
  draft-ietf-sip-outbound-20#section-13 to honor RFC3261 requirement for =
message=20
  size&nbsp;greater than&nbsp;MTU), would it be OK it one FLOW lands on =
Host1=20
  and other FLOW lands on Host2?</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>i.e for example if =
"sip:primary.example.com" got=20
  resolved into (IP1, Port1, TCP) -&gt; (IP2, Port2, TCP) -&gt; (IP2, =
Port2,=20
  UDP) -&gt; (IP1, Port1, UDP) in that order, and if UA wises to create =
both TCP=20
  and UDP FLOWs with "sip:primary.example.com", then it starts creating =
TCP FLOW=20
  with (IP1, Port1, TCP) and can create a UDP FLOW with (IP2, Port2, =
UDP) based=20
  on RFC 3263 procedures. Is my understanding correct here? =
</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>In the above example, if (IP1) is =
down/crashed for=20
  some reasons, then "sip:primary.example.com", can never use the TCP =
FLOW for=20
  transporting inbound messages to that UA (Even though the IP1 and IP2 =
are=20
  state synchronized) until User Agent detects the FLOW failure with =
(IP1) and=20
  re-create the TCP FLOW with (IP1 or IP2). Is my understanding correct=20
  here?</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Please clarify..</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Regard</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D791390009-28082009>Srikanth</SPAN></FONT></DIV><BR>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Francois Audet =
[mailto:audet@nortel.com]=20
  <BR><B>Sent:</B> Thursday, August 27, 2009 9:22 PM<BR><B>To:</B> =
Vavilapalli=20
  Srikanth-A19563; sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Question on =

  draft-ietf-sip-outbound-20 draft: multiple =
flowcreation<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><SPAN class=3D393564715-27082009><FONT face=3DArial =
color=3D#800000 size=3D2>You=20
  would have 2. </FONT></SPAN><SPAN class=3D393564715-27082009><FONT =
face=3DArial=20
  color=3D#800000 size=3D2>One for primary, and one for secondary. For =
each one of=20
  those, you would use normal RFC 3263 procedures for determining which =
SRV=20
  record to use.</FONT></SPAN></DIV><BR>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
    [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Vavilapalli=20
    Srikanth-A19563<BR><B>Sent:</B> Thursday, August 27, 2009=20
    07:57<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] Question =
on=20
    draft-ietf-sip-outbound-20 draft: multiple =
flowcreation<BR></FONT><BR></DIV>
    <DIV></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009>Hi</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN class=3D100403814-27082009>As =
specified in=20
    section 3.3/3.4 of draft-ietf-sip-outbound-20 draft, If a UA is =
configured=20
    with a outbound-proxy-set with two proxy uris: =
"sip:primary.example.com" and=20
    "sip:secondary.example.com" And for example, =
"sip:primary.example.com" got=20
    DNS resolved in to two SRV records "Host1" and "Host2" and=20
    "sip:secondary.example.com" got DNS resolved in to "Host3" and =
"Host4",=20
    Should UA create four FLOWs, One with each host&nbsp;OR just two =
FLOWs, one=20
    with each proxy URI(i.e one flow with either Host1 or Host2 and =
second flow=20
    with Host3 or Host4)?</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D100403814-27082009>Please=20
    clarify.</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    class=3D100403814-27082009>Regards</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN=20
    =
class=3D100403814-27082009>Srikanth</SPAN></FONT></DIV></BLOCKQUOTE></BLO=
CKQUOTE></BODY></HTML>

------_=_NextPart_001_01CA2A48.0C9B6CCA--

From scott.lawrence@nortel.com  Mon Aug 31 09:31:05 2009
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Cc: sip@ietf.org, Francois Audet <audet@nortel.com>, sip@core3.amsl.com, sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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On Mon, 2009-08-31 at 22:33 +0800, Vavilapalli Srikanth-A19563 wrote:
> True, UDP flow may not be required if TCP flow has been setup to route
> incoming traffic towards UA. But Won't it be an additional bonus if UA
> creates a UDP FLOW in addition to TCP FLOW, so that network can still
> reach the UA over the other transport FLOW when one FLOW is down
> (Unless if both FLOWs got created with same host and all the FLOWs
> connected to that host is down)? 
>  
> Also assume a UA has some local policy to prefer UDP transport over
> TCP to transmit the outgoing SIP messages of size less than MTU. In
> such scenarios, UA might want to establish both UDP and TCP FLOWs with
> the same edge proxy and keep alive both the connections and use the
> corresponding FLOW based on DNS lookup preferences and User's local
> policy preferences to transport outgoing messages.

That would not be a very sensible proxy if the phone is behind a NAT and
supports TCP.  




From scott.lawrence@nortel.com  Mon Aug 31 09:45:18 2009
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To: Vavilapalli Srikanth-A19563 <srikanthv@motorola.com>
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Cc: sip@ietf.org, Francois Audet <audet@nortel.com>, sip@core3.amsl.com, sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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On Mon, 2009-08-31 at 22:33 +0800, Vavilapalli Srikanth-A19563 wrote:
> True, UDP flow may not be required if TCP flow has been setup to route
> incoming traffic towards UA. But Won't it be an additional bonus if UA
> creates a UDP FLOW in addition to TCP FLOW, so that network can still
> reach the UA over the other transport FLOW when one FLOW is down
> (Unless if both FLOWs got created with same host and all the FLOWs
> connected to that host is down)? 
>  
> Also assume a UA has some local policy to prefer UDP transport over
> TCP to transmit the outgoing SIP messages of size less than MTU. In
> such scenarios, UA might want to establish both UDP and TCP FLOWs with
> the same edge proxy and keep alive both the connections and use the
> corresponding FLOW based on DNS lookup preferences and User's local
> policy preferences to transport outgoing messages.

That would not be a very sensible proxy if the phone is behind a NAT and
supports TCP.  




From scott.lawrence@nortel.com  Mon Aug 31 09:55:48 2009
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References: <608D7BFF0D57794785CBC2C440BC4FA1058116D7@ZMY16EXM67.ds.mot.com> <1ECE0EB50388174790F9694F77522CCF1FA69CD0@zrc2hxm0.corp.nortel.com> <608D7BFF0D57794785CBC2C440BC4FA1058119A2@ZMY16EXM67.ds.mot.com> <1ECE0EB50388174790F9694F77522CCF1FB064B1@zrc2hxm0.corp.nortel.com> <608D7BFF0D57794785CBC2C440BC4FA105811D7C@ZMY16EXM67.ds.mot.com> <1251736265.19992.106.camel@scott>
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Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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On Mon, 2009-08-31 at 12:31 -0400, Scott Lawrence wrote:
> On Mon, 2009-08-31 at 22:33 +0800, Vavilapalli Srikanth-A19563 wrote:
> > True, UDP flow may not be required if TCP flow has been setup to route
> > incoming traffic towards UA. But Won't it be an additional bonus if UA
> > creates a UDP FLOW in addition to TCP FLOW, so that network can still
> > reach the UA over the other transport FLOW when one FLOW is down
> > (Unless if both FLOWs got created with same host and all the FLOWs
> > connected to that host is down)? 
> >  
> > Also assume a UA has some local policy to prefer UDP transport over
> > TCP to transmit the outgoing SIP messages of size less than MTU. In
> > such scenarios, UA might want to establish both UDP and TCP FLOWs with
> > the same edge proxy and keep alive both the connections and use the
> > corresponding FLOW based on DNS lookup preferences and User's local
> > policy preferences to transport outgoing messages.
> 
> That would not be a very sensible proxy if the phone is behind a NAT and
> supports TCP.  

s/proxy/policy/




From scott.lawrence@nortel.com  Mon Aug 31 09:55:56 2009
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References: <608D7BFF0D57794785CBC2C440BC4FA1058116D7@ZMY16EXM67.ds.mot.com> <1ECE0EB50388174790F9694F77522CCF1FA69CD0@zrc2hxm0.corp.nortel.com> <608D7BFF0D57794785CBC2C440BC4FA1058119A2@ZMY16EXM67.ds.mot.com> <1ECE0EB50388174790F9694F77522CCF1FB064B1@zrc2hxm0.corp.nortel.com> <608D7BFF0D57794785CBC2C440BC4FA105811D7C@ZMY16EXM67.ds.mot.com> <1251736265.19992.106.camel@scott>
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Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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On Mon, 2009-08-31 at 12:31 -0400, Scott Lawrence wrote:
> On Mon, 2009-08-31 at 22:33 +0800, Vavilapalli Srikanth-A19563 wrote:
> > True, UDP flow may not be required if TCP flow has been setup to route
> > incoming traffic towards UA. But Won't it be an additional bonus if UA
> > creates a UDP FLOW in addition to TCP FLOW, so that network can still
> > reach the UA over the other transport FLOW when one FLOW is down
> > (Unless if both FLOWs got created with same host and all the FLOWs
> > connected to that host is down)? 
> >  
> > Also assume a UA has some local policy to prefer UDP transport over
> > TCP to transmit the outgoing SIP messages of size less than MTU. In
> > such scenarios, UA might want to establish both UDP and TCP FLOWs with
> > the same edge proxy and keep alive both the connections and use the
> > corresponding FLOW based on DNS lookup preferences and User's local
> > policy preferences to transport outgoing messages.
> 
> That would not be a very sensible proxy if the phone is behind a NAT and
> supports TCP.  

s/proxy/policy/




From AUDET@nortel.com  Mon Aug 31 10:11:02 2009
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Thread-Topic: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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From: "Francois Audet" <audet@nortel.com>
To: "Scott Lawrence" <scott.lawrence@nortel.com>, "Vavilapalli Srikanth-A19563" <srikanthv@motorola.com>
Cc: sip@ietf.org, sip@core3.amsl.com, sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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I would tend to disagree that a policy of "Use UDP preferably
and switch to TCP if message is too big" is a good policy. Especially
considering that you'd have to keep the TCP connection alive anyways.

I believe that always using the TCP connection in this case would
make more sense.

But it's definitively legal.

> -----Original Message-----
> From: Lawrence, Scott (BL60:9D30)=20
> Sent: Monday, August 31, 2009 09:56
> To: Vavilapalli Srikanth-A19563
> Cc: Audet, Francois (SC100:3055); sip@ietf.org;=20
> sip@core3.amsl.com; sip-implementors@lists.cs.columbia.edu
> Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20=20
> draft: multipleflowcreation
>=20
> On Mon, 2009-08-31 at 12:31 -0400, Scott Lawrence wrote:
> > On Mon, 2009-08-31 at 22:33 +0800, Vavilapalli=20
> Srikanth-A19563 wrote:
> > > True, UDP flow may not be required if TCP flow has been setup to=20
> > > route incoming traffic towards UA. But Won't it be an additional=20
> > > bonus if UA creates a UDP FLOW in addition to TCP FLOW, so that=20
> > > network can still reach the UA over the other transport FLOW when=20
> > > one FLOW is down (Unless if both FLOWs got created with same host=20
> > > and all the FLOWs connected to that host is down)?
> > > =20
> > > Also assume a UA has some local policy to prefer UDP=20
> transport over=20
> > > TCP to transmit the outgoing SIP messages of size less=20
> than MTU. In=20
> > > such scenarios, UA might want to establish both UDP and TCP FLOWs=20
> > > with the same edge proxy and keep alive both the=20
> connections and use=20
> > > the corresponding FLOW based on DNS lookup preferences and User's=20
> > > local policy preferences to transport outgoing messages.
> >=20
> > That would not be a very sensible proxy if the phone is=20
> behind a NAT=20
> > and supports TCP.
>=20
> s/proxy/policy/
>=20
>=20
>=20
>=20

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Thread-Topic: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
Thread-Index: AcoqW+jNRTPQ1zUnQEC3VhV8Ai+kcwAAd9Qg
References: <608D7BFF0D57794785CBC2C440BC4FA1058116D7@ZMY16EXM67.ds.mot.com> <1ECE0EB50388174790F9694F77522CCF1FA69CD0@zrc2hxm0.corp.nortel.com> <608D7BFF0D57794785CBC2C440BC4FA1058119A2@ZMY16EXM67.ds.mot.com> <1ECE0EB50388174790F9694F77522CCF1FB064B1@zrc2hxm0.corp.nortel.com> <608D7BFF0D57794785CBC2C440BC4FA105811D7C@ZMY16EXM67.ds.mot.com> <1251736265.19992.106.camel@scott> <1251737751.19992.132.camel@scott>
From: "Francois Audet" <audet@nortel.com>
To: "Scott Lawrence" <scott.lawrence@nortel.com>, "Vavilapalli Srikanth-A19563" <srikanthv@motorola.com>
Cc: sip@ietf.org, sip@core3.amsl.com, sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20 draft: multipleflowcreation
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I would tend to disagree that a policy of "Use UDP preferably
and switch to TCP if message is too big" is a good policy. Especially
considering that you'd have to keep the TCP connection alive anyways.

I believe that always using the TCP connection in this case would
make more sense.

But it's definitively legal.

> -----Original Message-----
> From: Lawrence, Scott (BL60:9D30)=20
> Sent: Monday, August 31, 2009 09:56
> To: Vavilapalli Srikanth-A19563
> Cc: Audet, Francois (SC100:3055); sip@ietf.org;=20
> sip@core3.amsl.com; sip-implementors@lists.cs.columbia.edu
> Subject: Re: [Sip] Question on draft-ietf-sip-outbound-20=20
> draft: multipleflowcreation
>=20
> On Mon, 2009-08-31 at 12:31 -0400, Scott Lawrence wrote:
> > On Mon, 2009-08-31 at 22:33 +0800, Vavilapalli=20
> Srikanth-A19563 wrote:
> > > True, UDP flow may not be required if TCP flow has been setup to=20
> > > route incoming traffic towards UA. But Won't it be an additional=20
> > > bonus if UA creates a UDP FLOW in addition to TCP FLOW, so that=20
> > > network can still reach the UA over the other transport FLOW when=20
> > > one FLOW is down (Unless if both FLOWs got created with same host=20
> > > and all the FLOWs connected to that host is down)?
> > > =20
> > > Also assume a UA has some local policy to prefer UDP=20
> transport over=20
> > > TCP to transmit the outgoing SIP messages of size less=20
> than MTU. In=20
> > > such scenarios, UA might want to establish both UDP and TCP FLOWs=20
> > > with the same edge proxy and keep alive both the=20
> connections and use=20
> > > the corresponding FLOW based on DNS lookup preferences and User's=20
> > > local policy preferences to transport outgoing messages.
> >=20
> > That would not be a very sensible proxy if the phone is=20
> behind a NAT=20
> > and supports TCP.
>=20
> s/proxy/policy/
>=20
>=20
>=20
>=20
