From sip-bounces@ietf.org  Tue Sep 21 20:51:36 2004
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From: "Thomas Gal" <ThomasGal@LumenVox.com>
To: "'vikram'" <vikramb@aftek.com>, "'sip-ietf'" <sip@ietf.org>
Subject: RE: [Sip] call stateful ?
Date: Tue, 21 Sep 2004 17:47:06 -0700
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That means a proxy tracks the transaction ID not the session ID. For example
once the invite has completed a proxy may know nothing about the call that
it connected while for a brief period of time it was "aware" (read
maintaining state) that it was in the middle of an invite command.

-Tom 

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of vikram
> Sent: Thursday, September 16, 2004 6:26 AM
> To: sip-ietf
> Subject: [Sip] call stateful ?
> 
> hello all,
>               Could anybody  pls explain following from rfc 3261
> 
>   " *A call stateful proxy is always transaction  stateful, 
> but the converse is not necessarily true"*
> 
> 
> regards
> 
> Vikram
> 
> _______________________________________________
> Sip mailing list  https://www1.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 sip-bounces@ietf.org  Wed Sep 22 00:55:51 2004
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From: "Umesh Sharma" <umesh_sharma@persistent.co.in>
To: <sip-bounces@ietf.org>, "sip" <sip@ietf.org>
Subject: RE: [Sip] Need guidance in call transfer - SIP Based
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pls see inline.
-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Anurag Kabra (by way of Anurag Kabra<anuragk@aftek.com>) (by way of
Anurag Kabra <anuragk@aftek.com>)
Sent: Monday, September 20, 2004 1:24 PM
To: sip@ietf.org
Subject: [Sip] Need guidance in call transfer - SIP Based


I am currently working on VoIP, and I am located in India.
 I have few queries which are bothering me, could you suggest me something
on
 them.  Below are my queries.

1. Do we do retransmission for Refer Request(& Notify) if it doesnot reach
transferee.
>> timeout/retransmission mechanism for refer is same as for bye, non-invite
transactions; when notify times out it is not retransmitted & correp.
subscription should be removed.

2. In a unattended call transfer, when does the dialog end i.e. normally
with a
BYE or Notify(subscriptio-state="terminated").
3. Does the dialog exists even after BYE or subscription-state header in
Notify will terminate the dialog.

2, 3>> my understanding from rfc 3265 bye would not terminate the dialog if
there is subscription on it, subscriber-state="terminated" destroys the
subscription and the dialog if no other application state is associated with
the dialog (more in 3.3.4 rfc 3265). For Call transfer refer goes with in
existing dialog.

4. Is 202 Accepted sent after getting a REFER request, ignored by the
transferor. What is the effect of the response 202 Accepted?
>> it indicates that the subscription has been accepted and that
authorization to proceed with transfer may or may not have been granted.
Also, that a notify will be sent immediately (3.1.4.1). It also creates a
new dialog & subscription if the initial refer was not sent on a
pre-existing dialog otherwise it just creates the new subscription with the
dialog.



This is all for now.  Hope to hear from you soon. Thanks in advance.

Best Regards
 Anurag.


_______________________________________________
Sip mailing list  https://www1.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 sip-bounces@ietf.org  Wed Sep 22 04:35:54 2004
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From: Goeman Stefan <Stefan.Goeman@siemens.com>
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Subject: [Sip]: SIP and SAML 
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Hello,

I was wondering if somebody already reviewed the ID
draft-tschofenig-sip-saml-00.txt.

This draft explicitly mentions that its main goal is not to realize single
sign-on (SSO) functionality as in HTTP. 
However, I wonder if it would not be a good idea to realize SSO
functionality with SAML in SIP. Right now, 3GPP has defined IMS-AKA. This
procedure authenticates the end-user when sending an initial SIP Register
request to the P-CSCF. It must be noted that this is an MNO-centric
approach; the MNO owning both a wireless access network and an IMS domain. 
However, it is not unthinkable that in the future there will exist entities
that will only operate an IMS domain. In this case, end-users will be
authenticated by the (wireless) access network provider. Then, I believe it
makes sense that the access network provider would provide SAML assertions
to the IMS operator providing information on the authentication status of
the end-user. 
How this should be done is unclear to me; I do have some initial ideas
though.

Anybody willing to comment?

Thanks in advance!

Greetings,
Stefan.


Stefan Goeman 
SIEMENS ICM/ICN 
Stefan.Goeman@siemens.com  <mailto:Stefan.Goeman@siemens.atea.be> 
 	


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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2657.73">
<TITLE>[Sip]: SIP and SAML </TITLE>
</HEAD>
<BODY>

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

<P><FONT SIZE=3D2 FACE=3D"Arial">I was wondering if somebody already =
reviewed the ID draft-tschofenig-sip-saml-00.txt.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">This draft explicitly mentions that =
its main goal is not to realize single sign-on (SSO) functionality as =
in HTTP. </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">However, I wonder if it would not be =
a good idea to realize SSO functionality with SAML in SIP. Right now, =
3GPP has defined IMS-AKA. This procedure authenticates the end-user =
when sending an initial SIP Register request to the P-CSCF. It must be =
noted that this is an MNO-centric approach; the MNO owning both a =
wireless access network and an IMS domain. </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">However, it is not unthinkable that in =
the future there will exist entities that will only operate an IMS =
domain. In this case, end-users will be authenticated by the (wireless) =
access network provider. Then, I believe it makes sense that the access =
network provider would provide SAML assertions to the IMS operator =
providing information on the authentication status of the end-user. =
</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">How this should be done is unclear to =
me; I do have some initial ideas though.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Anybody willing to comment?</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Thanks in advance!</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Greetings,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Stefan.</FONT>
</P>
<BR>

<P><FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Arial">Stefan =
Goeman</FONT><BR>
<B></B><B><FONT COLOR=3D"#008080" FACE=3D"Times New =
Roman">SIEMENS</FONT><FONT FACE=3D"Arial"> </FONT><FONT =
COLOR=3D"#008080" SIZE=3D2 FACE=3D"Arial">ICM/ICN</FONT><FONT =
FACE=3D"Arial"></FONT></B><BR>
<A HREF=3D"mailto:Stefan.Goeman@siemens.atea.be"><U></U><U><FONT =
COLOR=3D"#0000FF" SIZE=3D1 =
FACE=3D"Verdana">Stefan.Goeman@siemens.com</FONT></U></A>=20
<BR><FONT FACE=3D"Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</FONT>
</P>

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--===============1042764901==--



From sip-bounces@ietf.org  Wed Sep 22 05:25:31 2004
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	Wed, 22 Sep 2004 02:21:31 PDT
Date: Wed, 22 Sep 2004 02:21:31 -0700 (PDT)
From: brijesh nair <briju31@yahoo.com>
To: sip@ietf.org
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Subject: [Sip] Performance testing tools for SIP entities
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Hi,
  I am looking for good performace testing tools (free
or otherwise) which can be used to test various SIP
entities (e.g B2BUA, Proxy Servers etc.). Can anyone
help on this??

Thanks,
Brijesh




		
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From: "Thomas Johansson G (HF/EAB)" <thomas.g.johansson@ericsson.com>
To: "'brijesh nair'" <briju31@yahoo.com>, sip@ietf.org
Subject: RE: [Sip] Performance testing tools for SIP entities
Date: Wed, 22 Sep 2004 11:30:47 +0200
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Sipp from HP might fulfil your needs (freeware):
http://sourceforge.net/projects/sipp

Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (xml based) scenarios, dynamic adjustement of call-rate and a comprehensive set of real-time statistics.

BRs
/Thomas

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
brijesh nair
Sent: den 22 september 2004 11:22
To: sip@ietf.org
Subject: [Sip] Performance testing tools for SIP entities


Hi,
  I am looking for good performace testing tools (free
or otherwise) which can be used to test various SIP
entities (e.g B2BUA, Proxy Servers etc.). Can anyone
help on this??

Thanks,
Brijesh




		
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_______________________________________________
Sip mailing list  https://www1.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://www1.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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  This message is in MIME format.  The first part should be readable text,
  while the remaining parts are likely unreadable without MIME-aware tools.

---1262393577-1882252819-1095845527=:16071
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Try:
SIPp - sipp.sourceforge.net

//Bj=F6rn

On Wed, 22 Sep 2004, brijesh nair wrote:

> Hi,
>  I am looking for good performace testing tools (free
> or otherwise) which can be used to test various SIP
> entities (e.g B2BUA, Proxy Servers etc.). Can anyone
> help on this??
>
> Thanks,
> Brijesh
>
>
>
>
>
> _______________________________
> Do you Yahoo!?
> Declare Yourself - Register online to vote today!
> http://vote.yahoo.com
>
> _______________________________________________
> Sip mailing list  https://www1.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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_______________________________________________
Sip mailing list  https://www1.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
---1262393577-1882252819-1095845527=:16071--



From sip-bounces@ietf.org  Wed Sep 22 08:58:39 2004
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From: Anurag Kabra <anuragk@aftek.com>
Organization: Aftek Infosys Ltd.
To: sip@ietf.org
Date: Wed, 22 Sep 2004 18:10:31 +0530
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Subject: [Sip] Voice Mail
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Hello Everyone

I need some links detailing Voice Mail Client implementation based on SIP=
=2E =20
Thank You in advance.

Best Regards
  Anurag.

_______________________________________________
Sip mailing list  https://www1.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 sip-bounces@ietf.org  Wed Sep 22 10:19:52 2004
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From: "Thibault Renier" <toubix@kom.auc.dk>
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Date: Wed, 22 Sep 2004 16:13:48 +0200
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Subject: [Sip] SIP and MIP addressing
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Hi all,

I am currently working on integrating SIP in architecture using Mobile IP
for mobility support.
My question concerns the addressing scheme:
At the SIP layer a SIP URI specifies the destination point ("To" field),
going from SIP hop to SIP hop. At the IP layer, the destination address is
usually the same as the current IP address (Care of Address in the MIP
jargon -CoA) that corresponds to the URI used in the "To" (I guess that the
association URI/current IP address of a user is made possible during the
registration procedure). When MIP is used at the IP layer, for
communications from the Correspondent Node (CN) to the Mobile Node (MN), the
destination address is the MN's home address and the packets should be
intercepted by the Home Agent (HA).
1. What is the route in a system combining MIP for mobility support and SIP
for signalling flow? Do the packets go through all the SIP servers/hops
before applying the MIP routing solutions? 
2. What does happen because of the 'conflict' of addresses? (SIP resolves
the URI with the CoA, while IP uses the MN's home address at first)

Thanks for your help
/Tibo

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From: Richard Shockey <richard@shockey.us>
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At 04:30 AM 9/22/2004, Goeman Stefan wrote:

>Hello,
>
>I was wondering if somebody already reviewed the ID 
>draft-tschofenig-sip-saml-00.txt.
>
>This draft explicitly mentions that its main goal is not to realize single 
>sign-on (SSO) functionality as in HTTP.
>However, I wonder if it would not be a good idea to realize SSO 
>functionality with SAML in SIP. Right now, 3GPP has defined IMS-AKA. This 
>procedure authenticates the end-user when sending an initial SIP Register 
>request to the P-CSCF. It must be noted that this is an MNO-centric 
>approach; the MNO owning both a wireless access network and an IMS domain.
>
>However, it is not unthinkable that in the future there will exist 
>entities that will only operate an IMS domain. In this case, end-users 
>will be authenticated by the (wireless) access network provider. Then, I 
>believe it makes sense that the access network provider would provide SAML 
>assertions to the IMS operator providing information on the authentication 
>status of the end-user.

IMHO this is only one useful application of SAML to the SIP environment.

I was personally disappointed this document did not recieve more attention 
in SanDiego I hope it will be given a broader hearing in DC.

There are a variety of issues related to SIP security and Identity that 
SAML could solve.


>How this should be done is unclear to me; I do have some initial ideas 
>though.
>
>Anybody willing to comment?
>
>Thanks in advance!
>
>Greetings,
>Stefan.
>
>Stefan Goeman
>SIEMENS ICM/ICN
><mailto:Stefan.Goeman@siemens.atea.be>Stefan.Goeman@siemens.com
>
>_______________________________________________
>Sip mailing list  https://www1.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


 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Senior Manager, Strategic Technology Initiatives
NeuStar Inc.
46000 Center Oak Plaza  -   Sterling, VA  20166
sip:rshockey(at)iptel.org   sip:57141@fwd.pulver.com
ENUM +87810-13313-31331
PSTN Office +1 571.434.5651 PSTN Mobile: +1 703.593.2683,  Fax: +1 815.333.1237
<mailto:richard(at)shockey.us> or <mailto:richard.shockey(at)neustar.biz>
<http://www.neustar.biz> ; <http://www.enum.org>
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

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<html>
<body>
At 04:30 AM 9/22/2004, Goeman Stefan wrote:<br><br>
<blockquote type=cite class=cite cite=""><font size=2>Hello,</font>
<br><br>
<font size=2>I was wondering if somebody already reviewed the ID draft-tschofenig-sip-saml-00.txt.</font> <br><br>
<font size=2>This draft explicitly mentions that its main goal is not to realize single sign-on (SSO) functionality as in HTTP. </font><br>
<font size=2>However, I wonder if it would not be a good idea to realize SSO functionality with SAML in SIP. Right now, 3GPP has defined IMS-AKA. This procedure authenticates the end-user when sending an initial SIP Register request to the P-CSCF. It must be noted that this is an MNO-centric approach; the MNO owning both a wireless access network and an IMS domain. <br>
</font><br>
<font size=2>However, it is not unthinkable that in the future there will exist entities that will only operate an IMS domain. In this case, end-users will be authenticated by the (wireless) access network provider. Then, I believe it makes sense that the access network provider would provide SAML assertions to the IMS operator providing information on the authentication status of the end-user. </font></blockquote><br>
IMHO this is only one useful application of SAML to the SIP environment.<br><br>
I was personally disappointed this document did not recieve more attention in SanDiego I hope it will be given a broader hearing in DC.<br><br>
There are a variety of issues related to SIP security and Identity that SAML could solve.<br><br>
<br>
<blockquote type=cite class=cite cite=""><font size=2>How this should be done is unclear to me; I do have some initial ideas though.</font> <br><br>
<font size=2>Anybody willing to comment?</font> <br><br>
<font size=2>Thanks in advance!</font> <br><br>
<font size=2>Greetings,</font> <br>
<font size=2>Stefan.</font> <br><br>
<font size=2>Stefan Goeman</font><br>
<font face="Times New Roman, Times" color="#008080"><b>SIEMENS</font> <font size=2 color="#008080">ICM/ICN</b></font><br>
<font face="Verdana" size=1 color="#0000FF"><u><a href="mailto:Stefan.Goeman@siemens.atea.be">Stefan.Goeman@siemens.com</a></u></font> <br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>
_______________________________________________<br>
Sip mailing list&nbsp; <a href="https://www1.ietf.org/mailman/listinfo/sip" eudora="autourl">https://www1.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
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<x-sigsep><p></x-sigsep>
<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
Richard Shockey, Senior Manager, Strategic Technology Initiatives<br>
NeuStar Inc.<br>
46000 Center Oak Plaza&nbsp; -&nbsp;&nbsp; Sterling, VA&nbsp; 20166<br>
sip:rshockey(at)iptel.org&nbsp;&nbsp; sip:57141@fwd.pulver.com<br>
ENUM +87810-13313-31331<br>
PSTN Office +1 571.434.5651 PSTN Mobile: +1 703.593.2683,&nbsp; Fax: +1 815.333.1237<br>
&lt;<a href="mailto:richard(at)shockey.us" eudora="autourl">mailto:richard(at)shockey.us</a>&gt; or &lt;<a href="mailto:richard.shockey(at)neustar.biz" eudora="autourl">mailto:richard.shockey(at)neustar.biz</a>&gt;<br>
&lt;<a href="http://www.neustar.biz/" eudora="autourl">http://www.neustar.biz</a>&gt; ; &lt;<a href="http://www.enum.org/" eudora="autourl">http://www.enum.org</a>&gt;<br>
&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;<br>
</body>
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From sip-bounces@ietf.org  Wed Sep 22 20:06:34 2004
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Subject: Re: [Sip] Re: GRUU and things that rewrite contacts
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From: "Paul Kyzivat" <pkyzivat@cisco.com>
Sent: Thursday, September 16, 2004 6:31 PM
Subject: Re: [Sip] Re: GRUU and things that rewrite contacts

>
> I don't think we can meaningfully comment on the impact on SBCs when 
> we have no definition of such a beast. And trying to generalize to 
> all forms of B2BUA makes things worse - we repeatedly state that 
> about all you can say in general about a B2BUA is that it is two 
> connected UAs - you can't say anything about how the two calls are 
> related.
>
> If this is an important problem for us to mention at all, (and I'm 
> not yet convinced it is), then I think the first step is to define 
> an SBC, along with the kinds of transformations it performs. Since 
> that has never been written down, when it is written it can deal 
> properly with GRUUs.
>
> Paul

Yes, I agree completely. We can learn from the experience of waiting 
several years to figure out BCP for NATs, as we are now apparently 
doing in BEHAVE.

Spencer 



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To: "'Richard Shockey'" <richard@shockey.us>,
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Subject: RE: [Sip]: SIP and SAML 
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# hi richard, 
 
# ~snip~ 

	IMHO this is only one useful application of SAML to the SIP
environment.

# i fully agree with your observation. 
# there is a difference between the usage of saml in http and sip due to the
difference between these two protocols. 
# please take a look at my other mail as well. 

	I was personally disappointed this document did not recieve more
attention in SanDiego I hope it will be given a broader hearing in DC.

# the sip working group is very busy with other working group items. i
thankful that the chairs gave us a few minutes to introduce the document to
the working group. 
	
	There are a variety of issues related to SIP security and Identity
that SAML could solve.	

# true. 
		
~ snip ~ 

ciao
hannes 


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hi stefan, 

thanks for pointing to this important issue. we are currently working on a
draft update. we try to incorporate the issues presented in san diego (see
http://www.softarmor.com/sipwg/meets/ietf60/slides/pres-ietf60-sip-saml-tsch
ofenig.ppt). one of the open issues is related to reference integrity. we
also had a discussion on this issue. since you raised this issue i will give
some more insights in our off-line discussions. 

it is true that we mentioned in
http://www.ietf.org/internet-drafts/draft-tschofenig-sip-saml-00.txt that we
do not aim to create a solution for single sign-on (sso). 

unfortunately, the term 'single sign-on' is somewhat loosely defined. there
i have two answers for you:

first answer: 

in section 4.5 of draft-ietf-sipping-trait-authz-00.txt (trait-based authz
requirements draft) we describe a scenario where different protocols are
linked together using saml assertions. we use an example where we associate
a sip signaling protocol run and a qos signaling protocol. however, the
scenario is more generic in the sense that an arbitrary protocol can be
linked together with sip. if you call this "single sign-on" then this is
certainly in scope of our work. the current sip-saml draft supports such a
scenario. there are, however, some issues with reference integrity. this
issue will be treated in more detail in second answer.  

second answer: 

there is a difference between the usage of saml in http and sip. section 5
talks about a problem where a sip proxy can reuse an obtained saml assertion
to impersonate a user in future protocol runs. this problem has a direct
relationship to your question. 

a few solutions exist to address this problem:

1) you can delete this problem by assuming that all sip proxies are
trustworthy and do not act maliciously. 

2) the current draft talks about binding the saml assertion to a particular
sip session (see paragraphs mentioning the concept of reference integrity).
binding a saml assertion to a sip session means that 
  a) you compute a hash over a number of sip fields (see also
draft-ietf-sip-identity and draft-ietf-sip-authid-body)  
  b) include this hash into the saml assertion
  c) compute a digital signature 
this procedure has to be done by the authentication service. 

now, the main question is which fields to include in the hash. if you
include information such as the call-id then the assertion is only valid for
a particular sip session. if you only include the from,to field then the
assertion is valid for sip sessions between the two indicated entities
(typically for given timeframe - as specified in the validity of the saml
assertion). 

depending what you call single sign-on there might indeed be a problem. 

there is no reason to be disappointed. during a discussion with bob morgan
(one of the saml authors) at the last ietf he pointed me to the
holder-of-the-key assertion to deal with reference integrity in a better
way. the idea is the following: instead of using reference integrity with an
enhancement to the conditions extension point the holder-of-the-key
assertion allows to bind a key to the assertion. as a result, the end host
which was quite passive with the previous approach can be turned into an
active participant. obviously this considerably improves security (with the
drawback of adding some complexity to it).  

although we discussed this issue among the authors we haven't reached a
conclusion on it. 

the usage of saml in sip might heavily be scenario dependend. this means
that one solution might not fit in all cases. hence, i personally see only
the following options: 
- we do not strictly mandate which fields the authentication service has to
include into the hash function in order to gurantee reference integrity.  
- we additionally allow the usage of the holder-of-the-key assertion

ciao
hannes


> -----Original Message-----
> From: Goeman Stefan [mailto:Stefan.Goeman@siemens.com] 
> Sent: Mittwoch, 22. September 2004 10:31
> To: 'sip@ietf.org'
> Subject: [Sip]: SIP and SAML 
> 
> Hello, 
> 
> I was wondering if somebody already reviewed the ID 
> draft-tschofenig-sip-saml-00.txt. 
> 
> This draft explicitly mentions that its main goal is not to 
> realize single sign-on (SSO) functionality as in HTTP. 
> However, I wonder if it would not be a good idea to realize 
> SSO functionality with SAML in SIP. Right now, 3GPP has 
> defined IMS-AKA.
> This procedure authenticates the end-user when sending an 
> initial SIP Register request to the P-CSCF. It must be noted 
> that this is an MNO-centric approach; the MNO owning both a 
> wireless access network and an IMS domain. 
> 
> However, it is not unthinkable that in the future there will 
> exist entities that will only operate an IMS domain. In this 
> case, end-users will be authenticated by the (wireless) 
> access network provider. Then, I believe it makes sense that 
> the access network provider would provide SAML assertions to 
> the IMS operator providing information on the authentication 
> status of the end-user. 
> 
> How this should be done is unclear to me; I do have some 
> initial ideas though. 
> 
> Anybody willing to comment? 
> 
> Thanks in advance! 
> 
> Greetings,
> Stefan. 
> 
> 
> Stefan Goeman
> SIEMENS ICM/ICN
>  <mailto:Stefan.Goeman@siemens.atea.be> Stefan.Goeman@siemens.com 
>         
> 
> 


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Thread-Topic: RFC3515 Refer-To ABNF
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Subject: [Sip] RFC3515 Refer-To ABNF
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Hi,

Refer-To's ABNF:

      Refer-To =3D ("Refer-To" / "r") HCOLON ( name-addr / addr-spec ) *
      (SEMI generic-param)

as well as being incompatible with previous Refer-To draft ABNF (it adds =
optional parameters at the header level), does not seem to be consistent =
on its own.

If I have the header:

Refer-To: sip:user@host;param=3Dvalue

This is RFC3515 ABNF compliant. However, is "param" part of the header, =
or part of the URI? If i use the addr-spec form, i guess the param =
should be part of the URI then. Or is it?

The problem comes from allowing name-addr/addr-spec. This was also =
present for other headers in RFC3261, these are From,To, Contact and =
Reply-To, however for each of these a comment was inserted to clarify =
that the name-addr form (angle quotes) MUST be used if parameters or =
headers are present in the URI:

RFC3261 Sect 20:

   The Contact, From, and To header fields contain a URI.  If the URI
   contains a comma, question mark or semicolon, the URI MUST be
   enclosed in angle brackets (< and >).  Any URI parameters are
   contained within these brackets.  If the URI is not enclosed in angle
   brackets, any semicolon-delimited parameters are header-parameters,
   not URI parameters.

Sect 20.31 Reply-To
   Even if the "display-name" is empty, the "name-addr" form MUST be
   used if the "addr-spec" contains a comma, question mark, or
   semicolon.  Syntax issues are discussed in Section 7.3.1.


My question is: If RFC3261 mandates that the name-addr form be used if =
the URI has parameters/headers, why would it not be the same for =
Refer-To? It is safe to assume that extension headers should also =
conform to this? Should we add the section 20.31 comment Refer-To's =
definition? The examples section in 3515 goes this way...=20

I'm also wondering why the ABNF allows name-addr/addr-spec, when it =
could be simpler by allowing only name-addr, as is the case for Route =
and Record-route? Is this legacy for Pre-Rfc3515 compatibility?

Thanks,

Joel@m5t


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From sip-bounces@ietf.org  Thu Sep 23 11:18:46 2004
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From: Robert Sparks <RjS@xten.com>
Date: Thu, 23 Sep 2004 09:59:25 -0500
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Subject: [Sip] Re: RFC3515 Refer-To ABNF
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The use is the same in Refer-To as it was for From/To/Contact.

Any instance of name-addr | addr-spec in SIP headers will follow this 
same rule.
(and we are being more careful as we go forward to be explicit in each 
document)

If you have a URI parameter you must place it inside angle brackets. 
Any semicolon
delimited parameters appearing outside brackets are header parameters, 
not URI
parameters.

RjS

On Sep 23, 2004, at 9:42 AM, Joel Payeur wrote:

> Hi,
>
> Refer-To's ABNF:
>
>       Refer-To = ("Refer-To" / "r") HCOLON ( name-addr / addr-spec ) *
>       (SEMI generic-param)
>
> as well as being incompatible with previous Refer-To draft ABNF (it 
> adds optional parameters at the header level), does not seem to be 
> consistent on its own.
>
> If I have the header:
>
> Refer-To: sip:user@host;param=value
>
> This is RFC3515 ABNF compliant. However, is "param" part of the 
> header, or part of the URI? If i use the addr-spec form, i guess the 
> param should be part of the URI then. Or is it?
>
> The problem comes from allowing name-addr/addr-spec. This was also 
> present for other headers in RFC3261, these are From,To, Contact and 
> Reply-To, however for each of these a comment was inserted to clarify 
> that the name-addr form (angle quotes) MUST be used if parameters or 
> headers are present in the URI:
>
> RFC3261 Sect 20:
>
>    The Contact, From, and To header fields contain a URI.  If the URI
>    contains a comma, question mark or semicolon, the URI MUST be
>    enclosed in angle brackets (< and >).  Any URI parameters are
>    contained within these brackets.  If the URI is not enclosed in 
> angle
>    brackets, any semicolon-delimited parameters are header-parameters,
>    not URI parameters.
>
> Sect 20.31 Reply-To
>    Even if the "display-name" is empty, the "name-addr" form MUST be
>    used if the "addr-spec" contains a comma, question mark, or
>    semicolon.  Syntax issues are discussed in Section 7.3.1.
>
>
> My question is: If RFC3261 mandates that the name-addr form be used if 
> the URI has parameters/headers, why would it not be the same for 
> Refer-To? It is safe to assume that extension headers should also 
> conform to this? Should we add the section 20.31 comment Refer-To's 
> definition? The examples section in 3515 goes this way...
>
> I'm also wondering why the ABNF allows name-addr/addr-spec, when it 
> could be simpler by allowing only name-addr, as is the case for Route 
> and Record-route? Is this legacy for Pre-Rfc3515 compatibility?
>
> Thanks,
>
> Joel@m5t



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$B@P@n!!0&(B wrote:
> Hi,
> 
> I have a question associated with draft-ietf-sip-join-03.txt 
> and draft-ietf-sip-replaces-05.txt.
> 
> 
> Join document at the tail of Section 7.1  saids:
> 
> For compatibility with dialogs initiated by RFC2543 [11] 
> compliant UAs, a tag of zero matches both tags of zero and null tags
> 
> Examples:
>       Join: 87134@192.0.2.23;to-tag=24796;from-tag=0$B!!(B
> 
> and also said at section 6.1 in Replaces document.
> 
> 
> 
> In inverse case, i.e., there is a dialog initiated by RFC3261 
> compliant UAs to RFC2543 compliant UAs, is the same? 
> In other words, there is a possibility that there is 
> no to-tag in this case. So we can as below?
> 
> For example,
>       Join: 12345@192.1.2.3;to-tag=0;from-tag=56789
> 
> 
> In this example, does tag of zero matches both tags of zero 
> and null tags? And also for Replaces?

That's my reading of the doc, although in retrospect the wording at the
end of joing-7.1 is particularly clumsy, and I wish somebody had noticed
sooner.

I'd like to ask the authors to clarify here, and perhaps we can get an
errata entry with that clarification.

--
Dean

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Question 1:
Accordint to RFC 3261  10.2.2:
Use of the "*" Contact header field value allows a registering UA
to remove all bindings associated with an address-of-record
without knowing their precise values.
So if we use contact * and expire time zero can remove all contact =
bindings for  the user,
is it or not?

Question 2:
For the registrar discovery  mechanism in the RFC 3261 charpter 10.2.6:
we cand send register message with Request-URI "sip.mcast.net" to find =
registrar, and how the registrar reply with it? how we can get the =
registrar information, sucha as domain,ip address and port from the =
registrar response ?

Question 3:
Not like In H323 GK register mechanism the field termination type can be =
used to distinguish the GW or termiantion.  In SIP GW can the telephone =
number prefix be used in the from and to field to send to registrar for =
telephone number routing? if ok, maybe are there several user to =
register with registrar for one GW. if not How SIP GW can register with =
registrar?




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=D6=F7=CC=E2: Sip Digest, Vol 5, Issue 17


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Today's Topics:

   1. RE: call stateful ? (Thomas Gal)
   2. RE: Need guidance in call transfer - SIP Based (Umesh Sharma)
   3. [Sip]: SIP and SAML  (Goeman Stefan)
   4. Performance testing tools for SIP entities (brijesh nair)
   5. RE: Performance testing tools for SIP entities
      (Thomas Johansson G (HF/EAB))
   6. Re: Performance testing tools for SIP entities (Bj?rn Gr?nesj?)
   7. Voice Mail (Anurag Kabra)
   8. SIP and MIP addressing (Thibault Renier)
   9. Re: [Sip]: SIP and SAML  (Richard Shockey)


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

Message: 1
Date: Tue, 21 Sep 2004 17:47:06 -0700
From: "Thomas Gal" <ThomasGal@LumenVox.com>
Subject: RE: [Sip] call stateful ?
To: "'vikram'" <vikramb@aftek.com>, "'sip-ietf'" <sip@ietf.org>
Message-ID: <LV-SVR5P22aoCZjL7av00000660@lv-svr.lumenvox.com>
Content-Type: text/plain;	charset=3D"us-ascii"

That means a proxy tracks the transaction ID not the session ID. For =
example
once the invite has completed a proxy may know nothing about the call =
that
it connected while for a brief period of time it was "aware" (read
maintaining state) that it was in the middle of an invite command.

-Tom=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of vikram
> Sent: Thursday, September 16, 2004 6:26 AM
> To: sip-ietf
> Subject: [Sip] call stateful ?
>=20
> hello all,
>               Could anybody  pls explain following from rfc 3261
>=20
>   " *A call stateful proxy is always transaction  stateful,=20
> but the converse is not necessarily true"*
>=20
>=20
> regards
>=20
> Vikram
>=20
> _______________________________________________
> Sip mailing list  https://www1.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




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

Message: 2
Date: Wed, 22 Sep 2004 09:53:46 +0530
From: "Umesh Sharma" <umesh_sharma@persistent.co.in>
Subject: RE: [Sip] Need guidance in call transfer - SIP Based
To: <sip-bounces@ietf.org>, "sip" <sip@ietf.org>
Message-ID:
	<HNECIFNGLLKHFHHLCABCCELOCCAA.umesh_sharma@persistent.co.in>
Content-Type: text/plain;	charset=3D"iso-8859-1"



pls see inline.
-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Anurag Kabra (by way of Anurag Kabra<anuragk@aftek.com>) (by way of
Anurag Kabra <anuragk@aftek.com>)
Sent: Monday, September 20, 2004 1:24 PM
To: sip@ietf.org
Subject: [Sip] Need guidance in call transfer - SIP Based


I am currently working on VoIP, and I am located in India.
 I have few queries which are bothering me, could you suggest me =
something
on
 them.  Below are my queries.

1. Do we do retransmission for Refer Request(& Notify) if it doesnot =
reach
transferee.
>> timeout/retransmission mechanism for refer is same as for bye, =
non-invite
transactions; when notify times out it is not retransmitted & correp.
subscription should be removed.

2. In a unattended call transfer, when does the dialog end i.e. normally
with a
BYE or Notify(subscriptio-state=3D"terminated").
3. Does the dialog exists even after BYE or subscription-state header in
Notify will terminate the dialog.

2, 3>> my understanding from rfc 3265 bye would not terminate the dialog =
if
there is subscription on it, subscriber-state=3D"terminated" destroys =
the
subscription and the dialog if no other application state is associated =
with
the dialog (more in 3.3.4 rfc 3265). For Call transfer refer goes with =
in
existing dialog.

4. Is 202 Accepted sent after getting a REFER request, ignored by the
transferor. What is the effect of the response 202 Accepted?
>> it indicates that the subscription has been accepted and that
authorization to proceed with transfer may or may not have been granted.
Also, that a notify will be sent immediately (3.1.4.1). It also creates =
a
new dialog & subscription if the initial refer was not sent on a
pre-existing dialog otherwise it just creates the new subscription with =
the
dialog.



This is all for now.  Hope to hear from you soon. Thanks in advance.

Best Regards
 Anurag.


_______________________________________________
Sip mailing list  https://www1.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




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

Message: 3
Date: Wed, 22 Sep 2004 10:30:55 +0200
From: Goeman Stefan <Stefan.Goeman@siemens.com>
Subject: [Sip]: SIP and SAML=20
To: "'sip@ietf.org'" <sip@ietf.org>
Message-ID: <57FD2C3A246F76438CA6FDAD8FE9F195692195@hrtades7.atea.be>
Content-Type: text/plain; charset=3D"us-ascii"

Hello,

I was wondering if somebody already reviewed the ID
draft-tschofenig-sip-saml-00.txt.

This draft explicitly mentions that its main goal is not to realize =
single
sign-on (SSO) functionality as in HTTP.=20
However, I wonder if it would not be a good idea to realize SSO
functionality with SAML in SIP. Right now, 3GPP has defined IMS-AKA. =
This
procedure authenticates the end-user when sending an initial SIP =
Register
request to the P-CSCF. It must be noted that this is an MNO-centric
approach; the MNO owning both a wireless access network and an IMS =
domain.=20
However, it is not unthinkable that in the future there will exist =
entities
that will only operate an IMS domain. In this case, end-users will be
authenticated by the (wireless) access network provider. Then, I believe =
it
makes sense that the access network provider would provide SAML =
assertions
to the IMS operator providing information on the authentication status =
of
the end-user.=20
How this should be done is unclear to me; I do have some initial ideas
though.

Anybody willing to comment?

Thanks in advance!

Greetings,
Stefan.


Stefan Goeman=20
SIEMENS ICM/ICN=20
Stefan.Goeman@siemens.com  <mailto:Stefan.Goeman@siemens.atea.be>=20
 =09

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

Message: 4
Date: Wed, 22 Sep 2004 02:21:31 -0700 (PDT)
From: brijesh nair <briju31@yahoo.com>
Subject: [Sip] Performance testing tools for SIP entities
To: sip@ietf.org
Message-ID: <20040922092131.31780.qmail@web14224.mail.yahoo.com>
Content-Type: text/plain; charset=3Dus-ascii

Hi,
  I am looking for good performace testing tools (free
or otherwise) which can be used to test various SIP
entities (e.g B2BUA, Proxy Servers etc.). Can anyone
help on this??

Thanks,
Brijesh




	=09
_______________________________
Do you Yahoo!?
Declare Yourself - Register online to vote today!
http://vote.yahoo.com



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

Message: 5
Date: Wed, 22 Sep 2004 11:30:47 +0200
From: "Thomas Johansson G (HF/EAB)" <thomas.g.johansson@ericsson.com>
Subject: RE: [Sip] Performance testing tools for SIP entities
To: "'brijesh nair'" <briju31@yahoo.com>, sip@ietf.org
Message-ID:
	<633A2BAC69E8D411819A00508BCF883C0DA5F66D@esealnt452.al.sw.ericsson.se>
=09
Content-Type: text/plain;	charset=3D"iso-8859-1"

Sipp from HP might fulfil your needs (freeware):
http://sourceforge.net/projects/sipp

Sipp is a performance testing tool for the SIP protocol. Its main =
features are basic SIPStone scenarios, TCP/UDP transport, customizable =
(xml based) scenarios, dynamic adjustement of call-rate and a =
comprehensive set of real-time statistics.

BRs
/Thomas

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
brijesh nair
Sent: den 22 september 2004 11:22
To: sip@ietf.org
Subject: [Sip] Performance testing tools for SIP entities


Hi,
  I am looking for good performace testing tools (free
or otherwise) which can be used to test various SIP
entities (e.g B2BUA, Proxy Servers etc.). Can anyone
help on this??

Thanks,
Brijesh




	=09
_______________________________
Do you Yahoo!?
Declare Yourself - Register online to vote today!
http://vote.yahoo.com

_______________________________________________
Sip mailing list  https://www1.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



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

Message: 6
Date: Wed, 22 Sep 2004 11:32:07 +0200 (CEST)
From: Bj?rn Gr?nesj? <bg@ingate.com>
Subject: Re: [Sip] Performance testing tools for SIP entities
To: brijesh nair <briju31@yahoo.com>
Cc: sip@ietf.org
Message-ID: <Pine.LNX.4.61.0409221129380.16071@conar.ingate.se>
Content-Type: text/plain; charset=3D"iso-8859-1"

Try:
SIPp - sipp.sourceforge.net

//Bj=F6rn

On Wed, 22 Sep 2004, brijesh nair wrote:

> Hi,
>  I am looking for good performace testing tools (free
> or otherwise) which can be used to test various SIP
> entities (e.g B2BUA, Proxy Servers etc.). Can anyone
> help on this??
>
> Thanks,
> Brijesh
>
>
>
>
>
> _______________________________
> Do you Yahoo!?
> Declare Yourself - Register online to vote today!
> http://vote.yahoo.com
>
> _______________________________________________
> Sip mailing list  https://www1.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
>

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

Message: 7
Date: Wed, 22 Sep 2004 18:10:31 +0530
From: Anurag Kabra <anuragk@aftek.com>
Subject: [Sip] Voice Mail
To: sip@ietf.org
Message-ID: <200409221810.31676.anuragk@aftek.com>
Content-Type: text/plain;  charset=3D"us-ascii"

Hello Everyone

I need some links detailing Voice Mail Client implementation based on =
SIP. =20
Thank You in advance.

Best Regards
  Anurag.



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

Message: 8
Date: Wed, 22 Sep 2004 16:13:48 +0200
From: "Thibault Renier" <toubix@kom.auc.dk>
Subject: [Sip] SIP and MIP addressing
To: <sip@ietf.org>
Message-ID: <FCEGIMIHPNAAAKBFICFAOEOFCEAA.toubix@kom.auc.dk>
Content-Type: text/plain; charset=3D"iso-8859-1"

Hi all,

I am currently working on integrating SIP in architecture using Mobile =
IP
for mobility support.
My question concerns the addressing scheme:
At the SIP layer a SIP URI specifies the destination point ("To" field),
going from SIP hop to SIP hop. At the IP layer, the destination address =
is
usually the same as the current IP address (Care of Address in the MIP
jargon -CoA) that corresponds to the URI used in the "To" (I guess that =
the
association URI/current IP address of a user is made possible during the
registration procedure). When MIP is used at the IP layer, for
communications from the Correspondent Node (CN) to the Mobile Node (MN), =
the
destination address is the MN's home address and the packets should be
intercepted by the Home Agent (HA).
1. What is the route in a system combining MIP for mobility support and =
SIP
for signalling flow? Do the packets go through all the SIP servers/hops
before applying the MIP routing solutions?=20
2. What does happen because of the 'conflict' of addresses? (SIP =
resolves
the URI with the CoA, while IP uses the MN's home address at first)

Thanks for your help
/Tibo
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------------------------------

Message: 9
Date: Wed, 22 Sep 2004 10:24:46 -0400
From: Richard Shockey <richard@shockey.us>
Subject: Re: [Sip]: SIP and SAML=20
To: Goeman Stefan <Stefan.Goeman@siemens.com>, "'sip@ietf.org'"
	<sip@ietf.org>
Message-ID: <6.1.0.6.2.20040922102253.04e18210@joy.songbird.com>
Content-Type: text/plain; charset=3D"us-ascii"

At 04:30 AM 9/22/2004, Goeman Stefan wrote:

>Hello,
>
>I was wondering if somebody already reviewed the ID=20
>draft-tschofenig-sip-saml-00.txt.
>
>This draft explicitly mentions that its main goal is not to realize =
single=20
>sign-on (SSO) functionality as in HTTP.
>However, I wonder if it would not be a good idea to realize SSO=20
>functionality with SAML in SIP. Right now, 3GPP has defined IMS-AKA. =
This=20
>procedure authenticates the end-user when sending an initial SIP =
Register=20
>request to the P-CSCF. It must be noted that this is an MNO-centric=20
>approach; the MNO owning both a wireless access network and an IMS =
domain.
>
>However, it is not unthinkable that in the future there will exist=20
>entities that will only operate an IMS domain. In this case, end-users=20
>will be authenticated by the (wireless) access network provider. Then, =
I=20
>believe it makes sense that the access network provider would provide =
SAML=20
>assertions to the IMS operator providing information on the =
authentication=20
>status of the end-user.

IMHO this is only one useful application of SAML to the SIP environment.

I was personally disappointed this document did not recieve more =
attention=20
in SanDiego I hope it will be given a broader hearing in DC.

There are a variety of issues related to SIP security and Identity that=20
SAML could solve.


>How this should be done is unclear to me; I do have some initial ideas=20
>though.
>
>Anybody willing to comment?
>
>Thanks in advance!
>
>Greetings,
>Stefan.
>
>Stefan Goeman
>SIEMENS ICM/ICN
><mailto:Stefan.Goeman@siemens.atea.be>Stefan.Goeman@siemens.com
>
>_______________________________________________
>Sip mailing list  https://www1.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


 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Senior Manager, Strategic Technology Initiatives
NeuStar Inc.
46000 Center Oak Plaza  -   Sterling, VA  20166
sip:rshockey(at)iptel.org   sip:57141@fwd.pulver.com
ENUM +87810-13313-31331
PSTN Office +1 571.434.5651 PSTN Mobile: +1 703.593.2683,  Fax: +1 =
815.333.1237
<mailto:richard(at)shockey.us> or =
<mailto:richard.shockey(at)neustar.biz>
<http://www.neustar.biz> ; <http://www.enum.org>
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
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------------------------------

_______________________________________________
Sip mailing list  https://www1.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

End of Sip Digest, Vol 5, Issue 17
**********************************

_______________________________________________
Sip mailing list  https://www1.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 sip-bounces@ietf.org  Fri Sep 24 07:54:54 2004
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Date: Fri, 24 Sep 2004 17:20:33 +0530
From: Vikram Bhuskute <vikramb@aftek.com>
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Subject: [Sip] STUN server Misleads !
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Helo all,
          I am using an open stun client. It  helped me a lot . But 
there is problem.
        one out of  five times my stun gives me wrong type of NAT !! 
When i traced the packets ..
         .i found it is not my client's faul  server itself is 
responisble for that.

         e.g. I was behind port restricted NAT and still i got different 
binding for Test 1.2
          (request to other IP of STUN server)
         and my client  concluded from protocol that it is Symmetric NAT.

          I tried 5-6 STUN server i could found. but same problem

         Is it a known issue ? Is any STUN server is bug free ?


         regards

         vikram
         
          

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org  Fri Sep 24 08:20:48 2004
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From: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
To: "'Roman Shpount'" <roman@atelo.com>
Subject: RE: RE: RE: [Sip] Problem with an intial SIP INVITE server transa
	ction.
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Roman,

I believe you have strayed into a poorly specified area of SIP here.  I have
also found nothing in the specs. that directly answers your question however
I can tell you what is observed:  the duplication INVITEs are simply
ignored.  One option is to keep track of the INVITE transaction until the
ACK arrives.  Although logically there is no transaction layer interaction
(as per section 17, RFC 3261), you can then identify the INVITE as a
duplicate and discard it.

Paul D Smith
Network Protocols Group 
Data Connection Ltd (DCL)
Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com 
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com


-----Original Message-----
From: Roman Shpount [mailto:roman@atelo.com]
Sent: 19 September 2004 08:15
To: SIP Public Folder (E-mail)
Subject: Re: RE: RE: [Sip] Problem with an intial SIP INVITE server
transaction.


Thomas,

The problem is that retransmitted INVITE message does not have the To-tag.
Thus it does not match the dialog and UAC is not even informed about this
message. The only way I could think to deal with this, as I described in my
original email, was to add a timer which keeps the server INVITE transaction
open for T4 after 2XX response is sent. If retransmitted INVITE message
arrives within this time, it matches the transaction and is ignored. By the
way, this discussion is not academic, this is the real problem that we
encountered while dealing with large volumes of SIP transactions over Wi-Fi.
___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. -- www.atelo.com

------ Original message ------

>From: Thomas Gal <ThomasGal@LumenVox.com>
>Subject: RE: RE: [Sip] Problem with an intial SIP INVITE server
transaction.
>Date: 31 Aug 04, 04:44 PM
>To: 'Roman Shpount' <roman@atelo.com>,  <gledgard@iperia.com>,
<sip@ietf.org>

Maybe some people were just trying to help without scouring the spec for all
the details. I have read the spec quite a few times, thanks, but here, maybe
this will help(17):

	In the case of a transaction where the
   request was an INVITE (known as an INVITE transaction), the
   transaction also includes the ACK only if the final response was not
   a 2xx response.  If the response was a 2xx, the ACK is not considered
   part of the transaction.

      The reason for this separation is rooted in the importance of
      delivering all 200 (OK) responses to an INVITE to the UAC.  To
      deliver them all to the UAC, the UAS alone takes responsibility
      for retransmitting them (see Section 13.3.1.4), and the UAC alone
      takes responsibility for acknowledging them with ACK (see Section
      13.2.2.4).  Since this ACK is retransmitted only by the UAC, it is
      effectively considered its own transaction.

You gotta wait til you get the ACK.

-Tom


-----Original Message-----
From: Roman Shpount [mailto:roman@atelo.com] 
Sent: Tuesday, August 31, 2004 8:38 AM
To: gledgard@iperia.com; ThomasGal@LumenVox.com; sip@ietf.org
Subject: Re: RE: [Sip] Problem with an intial SIP INVITE server transaction.

Hi All, 

I got quite a few answers like: "there is a magic key which prevents this"
or "there is a timer that prevents this", or "you cannot get an INVITE once
you sent the 1XX or 2XX response". I would suggest that you would read the
RFC (Section 17.2.1) before rushing to answer my question. RFC clearly
spells out that INVITE server transaction is terminated when 2XX response is
sent. There is even a picture (Figure 7) that shows just that. There are no
timers. There are no special transaction IDs. Transaction is ended, clear
and simple. Once transaction is ended, re-transmitted INVITE message should
start the new transaction.  
___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. -- www.atelo.com

------ Original message ------

>From: Gordon Ledgard <gledgard@iperia.com>
>Subject: RE: [Sip] Problem with an intial SIP INVITE server transaction.
>Date: 31 Aug 04, 12:40 PM
>To:  <ThomasGal@LumenVox.com>, Roman Shpount <roman@atelo.com>,  
><sip@ietf.org>


Read the stuff about timers in the transaction layer. It spells out how you
need to hold on to transactions for messages over unreliable transports for
just this sort of reason.


Gordon



-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of Thomas
Gal
Sent: Monday, August 30, 2004 7:54 PM
To: 'Roman Shpount'; sip@ietf.org
Subject: RE: [Sip] Problem with an intial SIP INVITE server transaction.


I'm not an expert on this protocol (only read through the spec a few times
so it blurs together with some of the other ones) but even without looking
I'm SURE there's supposed to be a unique request-ID generated by the client
to avoid just such a confusion. Of course in this case any stateful UA would
ignore the already processed invite request. Also (sure about this one) if
the client received the provisional(100) response it wouldn't re-transmit
the INVITE either as that's the whole point.

-Tom

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Roman
Shpount
Sent: Tuesday, August 17, 2004 7:05 PM
To: sip@ietf.org
Subject: [Sip] Problem with an intial SIP INVITE server transaction.

I run into the problem with handling of retransmitted SIP INVITE message
based on the RFC 3621. 

Imagine the following situation

INVITE 1
----------------------->
100
<-----------------------
2XX
<-----------------------
re-transmitted INVITE 1
----------------------->

Based on RFC 3261, server INVITE transaction terminates as soon as 2XX
response is sent. This means that re-transmitted INVITE will be treated as a
new transaction. This INVITE message will not match the existing dialog,
since its To tag is empty. This means it will be treated as new dialog
creating message and phone will treat this message as a new call, which is
clearly not intended.

This situation can occur if both 100 and 2XX responses were lost by the
unreliable transport, and INVITE continues to be re-transmitted. The fact
that, 100 and 2XX responses were lost is unknown to the server transaction
and INVITE re-transmit can arrive before the 2XX re-transmit is sent. 

Majority of the transaction state machines in SIP are designed to handle
messages, that are received after the response is sent. It seems that
something simular is required in this case as well. 

In our implementation of SIP stack, server transaction is not terminated
when the 2XX response is sent. Instead, transaction is moved into the
"ConfirmedBis" state and timer, which is set to fire in T4, is started. If
the stack receives an INVITE message which matches the server transaction in
this state, it ignores it. If stack receives a 2XX response, which matches
the server transaction, it passes it to TU. Once the T4 timer is fired, the
transaction is terminated.

There is also an additional issue related to this problem. It looks like,
for the sake of consistency, some type of response should be sent
immediately to the transport layer when INVITE re-transmit is received. If
multiple 2XX responses has been sent, it is unclear what the behavior should
be: should all the 2XX messages be sent, should the last 2XX message be
resent or should the INVITE be plainly ignored. 

___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. -- www.atelo.com






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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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This list is for NEW development of the core SIP Protocol Use
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From sip-bounces@ietf.org  Sat Sep 25 16:01:28 2004
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	Tue, 21 Sep 2004 00:31:12 PDT
Date: Tue, 21 Sep 2004 00:31:12 -0700 (PDT)
From: Rayees Khan <rakhanhss@yahoo.com>
Subject: Re: [Sip] Contact: headers in SIP REGISTER 200 OK responses
To: "Gerke, Tom" <tom.gerke@intel.com>, sip@ietf.org
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=20
I think that the REGISTER messages from both the UAs would be taken by re=
gistrar to be from same UA. As a result the second REGISTER would be take=
n as re-REGISTER. Now the validity of the contacts in the two would depen=
d on the expires parameter in the contact header.=20
These contacts can be overlapping too.=20
In any case the response to the REGISTER message would be sent to the Via=
 address mentioned in the REGISTER rathar than the address specified in c=
ontact.
=20
=20
regards
Rayees
=20


"Gerke, Tom" <tom.gerke@intel.com> wrote:
Hello,

=20

If two different UAs (in the same domain) happen to register a user using=
 the same AOR (but different Contact addresses), to the same Registrar, w=
hat would be in their respective REGISTER 200 OK responses from that Regi=
strar?   i.e. Would they be notified of each others=92 bindings? =20

Section 10.3, step 8, of the SIP spec states that the 200 OK response wil=
l contain =93all current bindings=94.  It doesn=92t mention anything abou=
t the Call-IDs for those bindings having to match up or anything.

=20

So, let=92s say:

First, UA 1 decides to register AOR sip:userA@example.com with the Contac=
t address sip:usr1212@102.103.104.105 to a Registrar.

Then sometime later UA 2 decides to register AOR sip:userA@example.com wi=
th the Contact address sip:m1300@102.103.104.105:5070 to that same Regist=
rar.

=20

Assuming both of these UAs used unique Call-IDs, and there were no author=
ization issues, what Contact: headers would be in the REGISTER 200 OK res=
ponse message from this Registrar:

1) When UA 2 received its initial REGISTER response?

2) Sometime later when UA 1 then refreshes its binding for AOR sip:userA@=
example.com?

=20

Thanks in advance for any clarification on this matter,

-Tom

=20

_______________________________________________
Sip mailing list https://www1.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

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around=20
http://mail.yahoo.com=20
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<DIV>&nbsp;</DIV>
<DIV>I think that the REGISTER messages from both the UAs would be taken =
by registrar to be from same UA. As a result the second REGISTER would be=
 taken as re-REGISTER. Now the validity of the contacts in the two would =
depend on the expires parameter in the contact header. </DIV>
<DIV>These contacts can be overlapping too. </DIV>
<DIV>In any case the response to the REGISTER message would be sent to th=
e Via address mentioned in the REGISTER rathar than the address specified=
 in contact.</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>regards</DIV>
<DIV>Rayees</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR><BR><B><I>"Gerke, Tom" &lt;tom.gerke@intel.com&gt;</I></B> wrote=
:
<BLOCKQUOTE class=3Dreplbq style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #1010ff 2px solid">
<META content=3D"Microsoft Word 11 (filtered medium)" name=3DGenerator>
<STYLE>
<!--
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	{margin:0in;
	margin-bottom:.0001pt;
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a:link, span.MsoHyperlink
	{color:blue;
	text-decoration:underline;}
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	{mso-style-type:personal-compose;
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	margin:1.0in 1.25in 1.0in 1.25in;}
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</STYLE>

<DIV class=3DSection1>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial">Hello,<?xml:namespace prefix =3D o ns =3D "ur=
n:schemas-microsoft-com:office:office" /><o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN 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 style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial">If two different UAs (in the same domain) hap=
pen to register a user using the same AOR (but different Contact addresse=
s), to the same Registrar, what would be in their respective REGISTER 200=
 OK responses from that Registrar?&nbsp; &nbsp;i.e. Would they be notifie=
d of each others=92 bindings?&nbsp; <o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial">Section 10.3, step 8, of the SIP spec states =
that the 200 OK response will contain =93all current bindings=94.&nbsp; I=
t doesn=92t mention anything about the Call-IDs for those bindings having=
 to match up or anything.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN 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 style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial">So, let=92s say:<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial">First, UA 1 decides to register AOR sip:<A hr=
ef=3D"mailto:userA@example.com">userA@example.com</A> with the Contact ad=
dress sip:usr1212@102.103.104.105 to a Registrar.<o:p></o:p></SPAN></FONT=
></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial">Then sometime later UA 2 decides to register =
AOR sip:<A href=3D"mailto:userA@example.com">userA@example.com</A> with t=
he Contact address sip:m1300@102.103.104.105:5070 to that same Registrar.=
<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN 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 style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial">Assuming both of these UAs used unique Call-I=
Ds, and there were no authorization issues, what Contact: headers would b=
e in the REGISTER 200 OK response message from this Registrar:<o:p></o:p>=
</SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial">1) When UA 2 received its initial REGISTER re=
sponse?<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial">2) Sometime later when UA 1 then refreshes it=
s binding for AOR sip:<A href=3D"mailto:userA@example.com">userA@example.=
com</A>?<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN 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 style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial">Thanks in advance for any clarification on th=
is matter,<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial">-Tom<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN style=3D"FONT-SIZE=
: 10pt; FONT-FAMILY: Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P></DIV>____=
___________________________________________<BR>Sip mailing list https://w=
ww1.ietf.org/mailman/listinfo/sip<BR>This list is for NEW development of =
the core SIP Protocol<BR>Use sip-implementors@cs.columbia.edu for questio=
ns on current sip<BR>Use sipping@ietf.org for new developments on the app=
lication of sip</BLOCKQUOTE></DIV><p>____________________________________=
______________<br>Do You Yahoo!?<br>Tired of spam?  Yahoo! Mail has the b=
est spam protection around <br>http://mail.yahoo.com=20
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--===============1074374439==--



From sip-bounces@ietf.org  Sat Sep 25 16:01:55 2004
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Subject: RE: [Sip] Symmetric NAT issue
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From: "Christian Stredicke" <Christian.Stredicke@snom.de>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "vikram" <vikramb@aftek.com>
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IMVHO SBC that use ICE are a smart alternative to using TURN. UA that
are not aware of ICE, STUN, TURN or NAT will run media through the B2BUA
while ICE-aware devices can bypass them. IMEHO ist much easier to
implement STUN/ICE than TURN (well, we did/tried it) and it solves the
problem with the "dumb" devices. Paired with DNS SRV and STUN its
possible to automatically detect the nearest SBC, in case the SBC is
necessary to relay media for symmetrical NAT.

CS

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Jonathan Rosenberg
Sent: Tuesday, September 21, 2004 9:16 AM
To: vikram
Cc: sip@ietf.org
Subject: Re: [Sip] Symmetric NAT issue

There are some TURN implementations. There are security and interop
benefits of TURN over the b2bua-wm approach used by current SBC
products. TURN will work with e2e sip security mechanisms, and does not
require a b2bua function in the network, which can cause problems with
new sip features (see, for example, the recent thread on how sbc's
interfere with gruu operation).

Thanks,
Jonathan R.

vikram wrote:

> Hello All,
>                    We are facing lots of problem with NAT traversal=20
> for SIP .
>      It is not working behind symmetric NAT. I could see various=20
> solutions but none is 100% promising. ??
>       e.g   1) ALGs(intelligent firewall)                2)Session=20
> Border Controller
>               3)TURN
>               4) Tunneling (e.g X-tunnel)
>               5)  RTP proxy ??
>       Which one is best ???
>=20
>       we are looking for a solution which could work by just *changing

> our client* and/or *by using a global server*...
>      ALGs require upgradation of Firewalls, which everybody cant
afford.=20
> Tunneling too require some server entity within Intranet and
>       a global server, which again is not possible to deploy at every=20
> place  .
>=20
>      Now question remains about TURN and SBCs. TURN is still in=20
> process ..has anybody implemented. ?
>      Session border controller seems promising ...?
>      but could i have a SBC for a trial period with whom i can connect

> my client    ??
>      *Dose anybody provide free SBC service as of  SIP proxies ??*
>     =20
>     Please Help !!!!
>=20
>=20
>    regards
>=20
>   Vikram
>                       =20
>=20
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.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
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sip mailing list  https://www1.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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Hi,

The intent of the text is that either the to-tag or the from-tag 
parameter could match a null tag.  Examples are non-normative, so they 
should not limit what you can do.  The normative text intentionally 
says "a tag" and does not specify what kind of tag.

I'm assuming the following text would clarify things appropriately.

    For compatibility with ... RFC 2543 compliant UAs,
    a to-tag of zero matches both a to-tag value of
    zero and a null to-tag.  Likewise, a from-tag of zero
    matches both a to-tag value of zero and a null from-tag.

I might even be able to get this text into Replaces as well.

thanks,
-rohan



On Sep 23, 2004, at 8:03 AM, Dean Willis wrote:

> $B@P@n!!0&(B wrote:
>> Hi,
>>
>> I have a question associated with draft-ietf-sip-join-03.txt
>> and draft-ietf-sip-replaces-05.txt.
>>
>>
>> Join document at the tail of Section 7.1  saids:
>>
>> For compatibility with dialogs initiated by RFC2543 [11]
>> compliant UAs, a tag of zero matches both tags of zero and null tags
>>
>> Examples:
>>       Join: 87134@192.0.2.23;to-tag=24796;from-tag=0$B!!(B
>>
>> and also said at section 6.1 in Replaces document.
>>
>>
>>
>> In inverse case, i.e., there is a dialog initiated by RFC3261
>> compliant UAs to RFC2543 compliant UAs, is the same?
>> In other words, there is a possibility that there is
>> no to-tag in this case. So we can as below?
>>
>> For example,
>>       Join: 12345@192.1.2.3;to-tag=0;from-tag=56789
>>
>>
>> In this example, does tag of zero matches both tags of zero
>> and null tags? And also for Replaces?
>
> That's my reading of the doc, although in retrospect the wording at the
> end of joing-7.1 is particularly clumsy, and I wish somebody had 
> noticed
> sooner.
>
> I'd like to ask the authors to clarify here, and perhaps we can get an
> errata entry with that clarification.
>
> --
> Dean
>
> _______________________________________________
> Sip mailing list  https://www1.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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Hi,


The intent of the text is that either the to-tag or the from-tag
parameter could match a null tag.  Examples are non-normative, so they
should not limit what you can do.  The normative text intentionally
says "a tag" and does not specify what kind of tag.


I'm assuming the following text would clarify things appropriately.


   For compatibility with ... RFC 2543 compliant UAs, 

   a to-tag of zero matches both a to-tag value of

   zero and a null to-tag.  Likewise, a from-tag of zero

   matches both a to-tag value of zero and a null from-tag.


I might even be able to get this text into Replaces as well.


thanks,

-rohan




On Sep 23, 2004, at 8:03 AM, Dean Willis wrote:


<excerpt><fontfamily><param>Hiragino Kaku Gothic Pro</param>$B@P@n!!0&(B</fontfamily>
wrote:

<excerpt>Hi,


I have a question associated with draft-ietf-sip-join-03.txt 

and draft-ietf-sip-replaces-05.txt.



Join document at the tail of Section 7.1  saids:


For compatibility with dialogs initiated by RFC2543 [11] 

compliant UAs, a tag of zero matches both tags of zero and null tags


Examples:

      Join:
87134@192.0.2.23;to-tag=24796;from-tag=0<fontfamily><param>Hiragino Kaku Gothic Pro</param>$B!!(B


</fontfamily>and also said at section 6.1 in Replaces document.




In inverse case, i.e., there is a dialog initiated by RFC3261 

compliant UAs to RFC2543 compliant UAs, is the same? 

In other words, there is a possibility that there is 

no to-tag in this case. So we can as below?


For example,

      Join: 12345@192.1.2.3;to-tag=0;from-tag=56789



In this example, does tag of zero matches both tags of zero 

and null tags? And also for Replaces?

</excerpt>

That's my reading of the doc, although in retrospect the wording at the

end of joing-7.1 is particularly clumsy, and I wish somebody had
noticed

sooner.


I'd like to ask the authors to clarify here, and perhaps we can get an

errata entry with that clarification.


--

Dean


_______________________________________________

Sip mailing list  https://www1.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

</excerpt>
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Use sipping@ietf.org for new developments on the application of sip
--===============0235458443==--




From sip-bounces@ietf.org  Sun Sep 26 22:03:48 2004
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Vikram,

The problem may lie in the fact that RFC 3489's definition of NAT types 
is known to be outdated. In particular, some NATs are known to exhibit 
different behaviors based on whether the source port is in use or not. See:

http://www.ietf.org/internet-drafts/draft-jennings-midcom-stun-results-01.txt
http://www.ietf.org/internet-drafts/draft-audet-nat-behave-00.txt

-Jonathan R.

Vikram Bhuskute wrote:

> Helo all,
>          I am using an open stun client. It  helped me a lot . But there 
> is problem.
>        one out of  five times my stun gives me wrong type of NAT !! When 
> i traced the packets ..
>         .i found it is not my client's faul  server itself is 
> responisble for that.
> 
>         e.g. I was behind port restricted NAT and still i got different 
> binding for Test 1.2
>          (request to other IP of STUN server)
>         and my client  concluded from protocol that it is Symmetric NAT.
> 
>          I tried 5-6 STUN server i could found. but same problem
> 
>         Is it a known issue ? Is any STUN server is bug free ?
> 
> 
>         regards
> 
>         vikram
>                 
> _______________________________________________
> Sip mailing list  https://www1.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
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sip mailing list  https://www1.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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Subject: FW: [Sip-implementors] RE: [Sip] rfc3263 and rfc2915: trying next
	transport upon failure
Date: Mon, 27 Sep 2004 13:28:18 +0100
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Jonathan,

Can I please ask you to comment on the discussion Brett and I have been
having
via the WG regarding the use of NAPTR records?

The question is whether the use of multiple transports, indicated by
multiple
NAPTR records, SRV records or client support for multiple transports should
be attempted or not.

I can see an advantage to defining, say "TLS preferred over TCP over UDP"
but
this would only be useful under the "just try one transport" idea if the
client only supported one of the less preferred options.

However the "try all in order" idea allows for, for example, a separate TLS
connection server to be preferred but for TCP to be acceptable (possibly
with
some restrictions imposed through other means) if the TLS server is
unavailable.

So what is the intended processing?  Also in either case, would you like to 
raise a bugzilla report to ensure this is clarified in a future release of
the
specification?

Thanks,
Paul D Smith
Network Protocols Group 
Data Connection Ltd (DCL)
Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com 
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com


-----Original Message-----
From: Brett Tate [mailto:brett@broadsoft.com]
Sent: 17 September 2004 18:31
To: SIP Public Folder (E-mail)
Subject: RE: [Sip-implementors] RE: [Sip] rfc3263 and rfc2915: trying
next transport upon failure


Hi Jonathan,

Do you agree with Paul's interpretation 
concerning advancement to the next transport
(next NAPTR record or set of SRV records)?

My interpretation was that advancing
between transports has been removed.

There are likely good reasons for and
against advancing between transports.
I am just trying to understand what
rfc3263, rfc2915, and rfc3261 indicate
or should have indicated.

Thanks in advance,
Brett Tate


> RFC3263, section 4.1, contains the following text
> 
>  If no NAPTR records are found, the client 
>  constructs SRV queries for those transport 
>  protocols it supports, and does a query for 
>  each.  Queries are done using the service 
>  identifier "_sip" for SIP URIs and "_sips" 
>  for SIPS URIs.  A particular transport is 
>  supported if the query is successful.  The 
>  client MAY use any transport protocol it 
>  desires which is supported by the server.
> 
> This means that potentially the set of 
> SRV records covers multiple protocols in 
> the case that no NAPTR records are returned.

I interpret the text as indicating that one
transport should be selected instead of
potentially combining the multiple sets of
srv records.

> It therefore seems only sensible to me 
> that if multiple transports are indicated 
> by NAPTR records, that they too should be 
> tried, in the order indicated by the NAPTR 
> records.

Hopefully there would have been a statement 
indicating when to advance to the next NAPTR
record.  Or there could have been a statement
indicating the combining of all the SRV records
for each NAPTR record.


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Thread-Topic: Simple question about ENUM service in SIP solutions ?
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From: =?iso-8859-1?Q?Patrick_Ren=E9_Guillemin?= <Patrick.Guillemin@etsi.org>
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Subject: [Sip] Simple question about ENUM service in SIP solutions ?
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Dear All,

Basic question :

Do you know a SIP product available implementating enum ?

We are preparing enum interop tests and need to identify which SIP
products that already adopted enum ?

We will contact them to set up the test bed for next coming enum =
Plugtests.

Thank you very much I advance for any help.

Patrick GUILLEMIN=20
ETSI - Plugtests Technical Coordinator=20
http://www.etsi.org/plugtests
tel +33(0)4 92 94 43 31=20
fax +33(0)4 92 38 52 31=20
_______________________________________=20



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From: "Elwell, John" <john.elwell@siemens.com>
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Subject: [Sip] Question on offer-answer during establishment
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RFC3261 states (13.2.1, second bullet):

"The UAC MUST treat the first session description it receives as the answer,
and MUST ignore any session descriptions in subsequent responses to the
initial INVITE."

Suppose SDP answer is received in a reliable provisional response. This
would complete the offer-answer sequence. The behaviour mandated here would
seem to preclude the UAS sending a further offer in a subsequent reliable
provisional response or 2xx. Is this intended, and if so for what reason?
Obviously the UAS can send an UPDATE request (or re-INVITE request after
answer) if it wishes to re-negotiate the session, but this would involve
additional messages.

John Elwell (john.elwell@siemens.com)



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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Elwell, John'" <john.elwell@siemens.com>, sip@ietf.org
Subject: RE: [Sip] Question on offer-answer during establishment
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Hi,

>RFC3261 states (13.2.1, second bullet):
> 
>"The UAC MUST treat the first session description it receives 
>as the answer, and MUST ignore any session descriptions in subsequent 
>responses to the initial INVITE."
> 
>Suppose SDP answer is received in a reliable provisional 
>response. This would complete the offer-answer sequence.

Correct.

>The behaviour mandated here would seem to preclude the UAS sending a further offer in a 
>subsequent reliable provisional response or 2xx.

Exactly.

>Is this intended, and if so for what reason?

There has been discussions about this before, but I guess to put it short the answer is that you can only have at-most (sometimes you only send a single offer, or answer, within one transaction, but never more than one complete offer/answer exchange) one offer/answer exchange per SIP transaction.

>Obviously the UAS can send an UPDATE request (or re-INVITE 
>request after answer) if it wishes to re-negotiate the session,

Correct, but that is within a separate SIP transaction.

>but this would involve additional messages.

C'est la vie.

Please also note, that you are only allowed to send SDP (offer or answer) in the FIRST PRACK, so even if you would send new offers in provisional responses you would still have to use a non-PRACK mechanism to send the answers.

Sending SDPs in PRACK then causes other problems, ie how you would reject those, but that's another issue...

Regards,

Christer Holmberg
Ericsson Finland









> 
> John Elwell (john.elwell@siemens.com)
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.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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To: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Question on offer-answer during establishment
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Christer,

Can you point me to the RFC(s) (probably in 3262?) that says a UA is 
only allowed to send SDP in the first PRACK? Thanks

Michael

Christer Holmberg (JO/LMF) wrote:

>Please also note, that you are only allowed to send SDP (offer or answer) in the FIRST PRACK, so even if you would send new offers in provisional responses you would still have to use a non-PRACK mechanism to send the answers.
>
>Sending SDPs in PRACK then causes other problems, ie how you would reject those, but that's another issue...
>

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Subject: Re: [Sip] Question on offer-answer during establishment
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Christer Holmberg (JO/LMF) wrote:
> 
> Please also note, that you are only allowed to send SDP (offer or answer) in the FIRST PRACK, so even if you would send new offers in provisional responses you would still have to use a non-PRACK mechanism to send the answers.
> 
> Sending SDPs in PRACK then causes other problems, ie how you would reject those, but that's another issue...

Christer,

I have to quibble with the above.

If the INVITE is offerless, then an offer must be present in the first 
reliable response. If that is a reliable 1xx, then the 1xx must be 
answered with a PRACK, and that prack must contain the answer. So in 
that case the answer would be in the first PRACK.

I believe that is the *only* case in which an answer can be in a PRACK, 
and that there are *no* cases where an offer can be in a prack.

	Paul


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From sip-bounces@ietf.org  Tue Sep 28 11:34:13 2004
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Subject: Re: [Sip] Contact: headers in SIP REGISTER 200 OK responses
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Rayees Khan wrote:
> =20
> I think that the REGISTER messages from both the UAs would be taken by=20
> registrar to be from same UA.

I disagree. It seems pretty clearcut that these are independent and that=20
the registrar should accept them both. This is exactly how you get=20
multiple contacts associated with an AOR.

More below.

 > As a result the second REGISTER would be
> taken as re-REGISTER. Now the validity of the contacts in the two would=
=20
> depend on the expires parameter in the contact header.
> These contacts can be overlapping too.
> In any case the response to the REGISTER message would be sent to the=20
> Via address mentioned in the REGISTER rathar than the address specified=
=20
> in contact.
> =20
> =20
> regards
> Rayees
> =20
>=20
>=20
> */"Gerke, Tom" <tom.gerke@intel.com>/* wrote:
>=20
>     Hello,
>=20
>     =20
>=20
>     If two different UAs (in the same domain) happen to register a user
>     using the same AOR (but different Contact addresses), to the same
>     Registrar, what would be in their respective REGISTER 200 OK
>     responses from that Registrar?   i.e. Would they be notified of eac=
h
>     others=92 bindings?=20
>=20
>     Section 10.3, step 8, of the SIP spec states that the 200 OK
>     response will contain =93all current bindings=94.  It doesn=92t men=
tion
>     anything about the Call-IDs for those bindings having to match up o=
r
>     anything.
>=20
>     =20
>=20
>     So, let=92s say:
>=20
>     First, UA 1 decides to register AOR sip:userA@example.com
>     <mailto:userA@example.com>

I don't understand the <mailto...> in the above. You have two URLs, and=20
the mailto seems entirely inappropriate. I'm assuming it isn't there.

The presence of this suggests that you have some kind of=20
misunderstanding, but I can't tell what it is.

 >     with the Contact address
>     sip:usr1212@102.103.104.105 to a Registrar.
>=20
>     Then sometime later UA 2 decides to register AOR
>     sip:userA@example.com <mailto:userA@example.com> with the Contact

again ignoring <mailto...>

>     address sip:m1300@102.103.104.105:5070 to that same Registrar.
>=20
>     =20
>=20
>     Assuming both of these UAs used unique Call-IDs, and there were no
>     authorization issues, what Contact: headers would be in the REGISTE=
R
>     200 OK response message from this Registrar:
>=20
>     1) When UA 2 received its initial REGISTER response?

It would see both registered contacts. Each should show the remaining=20
time till expiration in the expires parameter.

>     2) Sometime later when UA 1 then refreshes its binding for AOR
>     sip:userA@example.com <mailto:userA@example.com>?

Assuming the contact from UA2 hasn't expired, there should again be two=20
contacts, as above. If the contact from UA2 had expired without refresh,=20
then only the refreshed contact would be seen.

>     Thanks in advance for any clarification on this matter,
>=20
>     -Tom
>=20
>     =20
>=20
>     _______________________________________________
>     Sip mailing list https://www1.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
> __________________________________________________
> Do You Yahoo!?
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>=20
>=20
> -----------------------------------------------------------------------=
-
>=20
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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A new Request for Comments is now available in online RFC libraries.


        RFC 3891

        Title:      The Session Initiation Protocol (SIP)
                    "Replaces" Header
        Author(s):  R. Mahy, B. Biggs, R. Dean
        Status:     Standards Track
        Date:       September 2004
        Mailbox:    rohan@cisco.com, bbiggs@dumbterm.net, rfc@fdd.com
        Pages:      16
        Characters: 34180
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-sip-replaces-05.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3891.txt


This document defines a new header for use with Session Initiation
Protocol (SIP) multi-party applications and call control.  The
Replaces header is used to logically replace an existing SIP dialog
with a new SIP dialog.  This primitive can be used to enable a variety
of features, for example: "Attended Transfer" and "Call Pickup".  Note
that the definition of these example features is non-normative.

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

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3892

        Title:      The Session Initiation Protocol (SIP)
                    Referred-By Mechanism
        Author(s):  R. Sparks
        Status:     Standards Track
        Date:       September 2004
        Mailbox:    RjS@xten.com
        Pages:      25
        Characters: 52441
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-sip-referredby-05.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3892.txt


The Session Initiation Protocol (SIP) REFER method provides a
mechanism where one party (the referrer) gives a second party (the
referee) an arbitrary URI to reference.  If that URI is a SIP URI, the
referee will send a SIP request, often an INVITE, to that URI (the
refer target).  This document extends the REFER method, allowing the
referrer to provide information about the REFER request to the refer
target using the referee as an intermediary.  This information
includes the identity of the referrer and the URI to which the
referrer referred.  The mechanism utilizes S/MIME to help protect this
information from a malicious intermediary.  This protection is
optional, but a recipient may refuse to accept a request unless it is
present.

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

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
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A new Request for Comments is now available in online RFC libraries.


        RFC 3893

        Title:      Session Initiation Protocol (SIP)
                    Authenticated Identity Body (AIB) Format
        Author(s):  J. Peterson
        Status:     Standards Track
        Date:       September 2004
        Mailbox:    jon.peterson@neustar.biz
        Pages:      13
        Characters: 28500
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-sip-authid-body-03.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3893.txt


RFC 3261 introduces the concept of adding an S/MIME body to a Session
Initiation Protocol (SIP) request or response in order to provide
reference integrity over its headers.  This document provides a more
specific mechanism to derive integrity and authentication properties
from an 'authenticated identity body', a digitally-signed SIP
message, or message fragment.  A standard format for such bodies
(known as Authenticated Identity Bodies, or AIBs) is given in this
document.  Some considerations for the processing of AIBs by
recipients of SIP messages with such bodies are also given.

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

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
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Subject: RE: [Sip] Question on offer-answer during establishment
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Hi,

>>Please also note, that you are only allowed to send SDP 
>>(offer or answer) in the FIRST PRACK, so even if you would 
>>send new offers in provisional responses you would still have 
>>to use a non-PRACK mechanism to send the answers.
>> 
>>Sending SDPs in PRACK then causes other problems, ie how 
>>you would reject those, but that's another issue...
> 
>Christer,
> 
>I have to quibble with the above.
> 
>If the INVITE is offerless, then an offer must be present in 
>the first reliable response. If that is a reliable 1xx, then the 1xx must be 
>answered with a PRACK, and that prack must contain the answer. So in 
>that case the answer would be in the first PRACK.

Correct.

>I believe that is the *only* case in which an answer can be 
>in a PRACK, and that there are *no* cases where an offer can be in a prack.

I would be very happy if that was true :)

However, chapter 5 in RFC 3262 says:

"If the UAC receives a reliable provisional response with an answer, it MAY generate an
additional offer in the PRACK."

Now, I have to admit I did not find the text saying that we're only allowed to use the first PRACK to send a new offer, but my understanding is that that is the case, and one of the reasons for that was to "save" a SIP transaction in pre-condition scenarios (instead of sending the first updated offer in an UPDATE you can use the PRACK). But, correct me if I'm wrong.


At the previous SIPit we had some discussions about the usage of PRACK for offer/answer. The problem, as said before, is that there is currently no standard way to reject an offer received in a PRACK, because a UAS MUST answer with 200 OK, if the PRACK otherwise is ok, matches a reliabel provisional response etc. It has even been suggested that it should be forbidden to send SDPs (at least new offers) in PRACKs, but that would of course have impacts on existing nodes - depending on how many nodes currently support it in the first place, of course.

Regards,

Christer Holmberg
Ericsson Finland






> 
> 	Paul
> 

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From sip-bounces@ietf.org  Wed Sep 29 03:09:47 2004
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Thanks for prodding me to answer this.

I agree this is confusing. RFC 3261 seems to refer to the notion that 
you can change transports on different attempts (end of 8.1.2). However, 
RFC 3263 would seem to indicate that the NAPTR result would be a single 
service, and therefore, a single transport.

The correct behavior is the latter. That is, the NAPTR processing will 
select a single transport, and that's what gets used. Previously, we 
were allowing attempts across transports because we were combining 
addresses from different SRV records and merging them. This was deemed a 
no-no by the DNS folks, due to the operational issues with comparing 
preference values across records.

Once we eliminated this merging, the updated procedures in RFC 3263 
effectively resulted in selection of a single transport, but this wasn't 
  made fully consistent with RFC 3261. I've logged a bug against 3261 to 
clarify (http://bugs.sipit.net/sipwg/show_bug.cgi?id=760).

Thanks,
Jonathan R.

Paul D.Smith wrote:

> Jonathan,
> 
> Can I please ask you to comment on the discussion Brett and I have been
> having
> via the WG regarding the use of NAPTR records?
> 
> The question is whether the use of multiple transports, indicated by
> multiple
> NAPTR records, SRV records or client support for multiple transports should
> be attempted or not.
> 
> I can see an advantage to defining, say "TLS preferred over TCP over UDP"
> but
> this would only be useful under the "just try one transport" idea if the
> client only supported one of the less preferred options.
> 
> However the "try all in order" idea allows for, for example, a separate TLS
> connection server to be preferred but for TCP to be acceptable (possibly
> with
> some restrictions imposed through other means) if the TLS server is
> unavailable.
> 
> So what is the intended processing?  Also in either case, would you like to 
> raise a bugzilla report to ensure this is clarified in a future release of
> the
> specification?
> 
> Thanks,
> Paul D Smith
> Network Protocols Group 
> Data Connection Ltd (DCL)
> Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com 
> Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
> 
> 
> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]
> Sent: 17 September 2004 18:31
> To: SIP Public Folder (E-mail)
> Subject: RE: [Sip-implementors] RE: [Sip] rfc3263 and rfc2915: trying
> next transport upon failure
> 
> 
> Hi Jonathan,
> 
> Do you agree with Paul's interpretation 
> concerning advancement to the next transport
> (next NAPTR record or set of SRV records)?
> 
> My interpretation was that advancing
> between transports has been removed.
> 
> There are likely good reasons for and
> against advancing between transports.
> I am just trying to understand what
> rfc3263, rfc2915, and rfc3261 indicate
> or should have indicated.
> 
> Thanks in advance,
> Brett Tate
> 
> 
> 
>>RFC3263, section 4.1, contains the following text
>>
>> If no NAPTR records are found, the client 
>> constructs SRV queries for those transport 
>> protocols it supports, and does a query for 
>> each.  Queries are done using the service 
>> identifier "_sip" for SIP URIs and "_sips" 
>> for SIPS URIs.  A particular transport is 
>> supported if the query is successful.  The 
>> client MAY use any transport protocol it 
>> desires which is supported by the server.
>>
>>This means that potentially the set of 
>>SRV records covers multiple protocols in 
>>the case that no NAPTR records are returned.
> 
> 
> I interpret the text as indicating that one
> transport should be selected instead of
> potentially combining the multiple sets of
> srv records.
> 
> 
>>It therefore seems only sensible to me 
>>that if multiple transports are indicated 
>>by NAPTR records, that they too should be 
>>tried, in the order indicated by the NAPTR 
>>records.
> 
> 
> Hopefully there would have been a statement 
> indicating when to advance to the next NAPTR
> record.  Or there could have been a statement
> indicating the combining of all the SRV records
> for each NAPTR record.
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From: Vikram Bhuskute <vikramb@aftek.com>
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Subject: [Sip] draft-takeda-symmetric-nat-traversal  ?? 
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Hello all,
            While searching for NAT traversal solution i came across a 
draft  "draft-takeda-symmetric-nat-traversal-00.txt"
            from url ( 
http://cnscenter.future.co.kr/resource/ietf/ind-draft/draft-takeda-symmetric-nat-traversal-00.txt)

           It proposes some solution for Symmetric NAT based on some 
Prediction .....?  It dosent seem reliable ... !!!
          (however it gives  glimps on type of Symmetric NATs )
 
           Is it still a matter of further research ? or the concept  is 
not well accepted ?

          thanks in advance

 vikram

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Hi Robert.

If this rule adjusts to all headers, I think that it is necessary
to allow the use of addr-spec for the Route/Record-Route header.

ex.
Route: sip:proxy.example.com

Jone.

Robert Sparks wrote in <29104F89-0D71-11D9-90F7-000D93326732@xten.com>
>The use is the same in Refer-To as it was for From/To/Contact.
>
>Any instance of name-addr | addr-spec in SIP headers will follow this
>same rule.
>(and we are being more careful as we go forward to be explicit in each
>document)
>
>If you have a URI parameter you must place it inside angle brackets.
>Any semicolon
>delimited parameters appearing outside brackets are header parameters,
>not URI
>parameters.
>
>RjS
>
>On Sep 23, 2004, at 9:42 AM, Joel Payeur wrote:
>
>> Hi,
>>
>> Refer-To's ABNF:
>>
>>       Refer-To = ("Refer-To" / "r") HCOLON ( name-addr / addr-spec ) *
>>       (SEMI generic-param)
>>
>> as well as being incompatible with previous Refer-To draft ABNF (it
>> adds optional parameters at the header level), does not seem to be
>> consistent on its own.
>>
>> If I have the header:
>>
>> Refer-To: sip:user@host;param=value
>>
>> This is RFC3515 ABNF compliant. However, is "param" part of the
>> header, or part of the URI? If i use the addr-spec form, i guess the
>> param should be part of the URI then. Or is it?
>>
>> The problem comes from allowing name-addr/addr-spec. This was also
>> present for other headers in RFC3261, these are From,To, Contact and
>> Reply-To, however for each of these a comment was inserted to clarify
>> that the name-addr form (angle quotes) MUST be used if parameters or
>> headers are present in the URI:
>>
>> RFC3261 Sect 20:
>>
>>    The Contact, From, and To header fields contain a URI.  If the URI
>>    contains a comma, question mark or semicolon, the URI MUST be
>>    enclosed in angle brackets (< and >).  Any URI parameters are
>>    contained within these brackets.  If the URI is not enclosed in
>> angle
>>    brackets, any semicolon-delimited parameters are header-parameters,
>>    not URI parameters.
>>
>> Sect 20.31 Reply-To
>>    Even if the "display-name" is empty, the "name-addr" form MUST be
>>    used if the "addr-spec" contains a comma, question mark, or
>>    semicolon.  Syntax issues are discussed in Section 7.3.1.
>>
>>
>> My question is: If RFC3261 mandates that the name-addr form be used if
>> the URI has parameters/headers, why would it not be the same for
>> Refer-To? It is safe to assume that extension headers should also
>> conform to this? Should we add the section 20.31 comment Refer-To's
>> definition? The examples section in 3515 goes this way...
>>
>> I'm also wondering why the ABNF allows name-addr/addr-spec, when it
>> could be simpler by allowing only name-addr, as is the case for Route
>> and Record-route? Is this legacy for Pre-Rfc3515 compatibility?
>>
>> Thanks,

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To: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
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Christer Holmberg (JO/LMF) wrote:

>At the previous SIPit we had some discussions about the usage of PRACK for offer/answer. The problem, as said before, is that there is currently no standard way to reject an offer received in a PRACK, because a UAS MUST answer with 200 OK, if the PRACK otherwise is ok, matches a reliabel provisional response etc. It has even been suggested that it should be forbidden to send SDPs (at least new offers) in PRACKs, but that would of course have impacts on existing nodes - depending on how many nodes currently support it in the first place, of course.
>  
>
It's not mentioned in RFC 3262 how to reject an offer, but 488 (Not 
Acceptable Here) might be a natural choice according to Section 13.3.1.3 
of RFC 3261, quoted below. In practice, 4xx-6xx respons to PRACK (which 
contains offer) may also be considered rejection of offer.

"

   A UAS rejecting an offer contained in an INVITE SHOULD return a 488
   (Not Acceptable Here) response.  Such a response SHOULD include a
   Warning header field value explaining why the offer was rejected.

"

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Subject: Re: [Sip] removing configuration from the SIP MIB
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here are the proposed access level changes for mib objects
in SIP-COMMON-MIB and SIP-SERVER-MIB modules...

based on...
http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-08.txt

SIP-COMMON-MIB
(currently 21 read-write objects)
OBJECT                     PROPOSED CHANGE
----------------------     ---------------
sipServiceAdminStatus      remove
sipOrganization            make read-only
sipServiceNotifEnable      no change (note 1)
sipCfgTimerA               make read-only
sipCfgTimerB               make read-only
sipCfgTimerC               make read-only
sipCfgTimerD               make read-only
sipCfgTimerE               make read-only
sipCfgTimerF               make read-only
sipCfgTimerG               make read-only
sipCfgTimerH               make read-only
sipCfgTimerI               make read-only
sipCfgTimerJ               make read-only
sipCfgTimerK               make read-only
sipCfgTimerT1              make read-only
sipCfgTimerT2              make read-only
sipCfgTimerT4              make read-only
sipStatusCodeNotifSend     no change (note 1)
sipStatusCodeNotifEmitMode no change (note 1)
sipStatusCodeNotifThresh   no change (note 1)
sipStatusCodeNotifInterval no change (note 1)

SIP-SERVER-MIB
  (9 read-write objects)
OBJECT                          PROPOSED CHANGE
----------------------          ---------------
sipServerHostAddrType           make read-only
sipServerHostAddr               make read-only
sipProxyRecursion               make read-only
sipProxyRecordRoute             make read-only
sipProxyAuthMethod              make read-only
sipProxyAuthRealm               make read-only
sipRegAllowThirdParty           make read-only
sipRegMaxContactExpiryDuration  make read-only
sipRegDfltRegActiveInterval     make read-only

note 1: read-write required to configure snmp notifications.

if there are no objections to this, we'll make it so in
rev -09 of the draft.

kevin

Rohan Mahy wrote:
> Hi Everyone,
> 
> The current SIP MIB  (-08) contains a handful of elements which are 
> related to configuration.  These elements are insufficient to usefully 
> configure a UA or a Proxy.   The SIP community already has a framework 
> for UA Profile configuration, and we have not identified anyone who is 
> planning to implement the SIP MIB for configuration. In addition, the 
> configuration issues directly affects 3 of the 4 open issues (#1, #46, 
> and #79) with the MIB. Finally, the IETF has de-emphasized using MIBs 
> for configuration since last year.
> 
> Unless someone objects (and volunteers to help), the chairs propose that 
> we do not provide  elements for configuration use, and don't make 
> elements write or create enabled which would only use this attribute for 
> configuration.  We would keep many of these same elements for monitoring 
> the actual configuration of elements relevant for monitoring, debugging, 
> or troubleshooting purposes.  If the group does not object, the authors 
> will propose specific changes to the list, edit the document 
> appropriately, and we can send the document to the Area Director.
> 
> Please send mail right away if you disagree with this approach.
> 
> If the group agrees with this proposal, this still leaves issue #87 as 
> an open issue, plus the issue that Bob Penfield recently brought  up on 
> the list. Most issues with the MIB are discussed here:
> 
> http://www1.ietf.org/mail-archive/web/sip/current/msg10184.html
> 
> many thanks,
> -rohan
> 
> 
> _______________________________________________
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> 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		: Enhancements for Authenticated Identity Management 
			  in the Session Initiation Protocol (SIP)
	Author(s)	: J. Peterson, C. Jennings
	Filename	: draft-ietf-sip-identity-03.txt
	Pages		: 32
	Date		: 2004-9-29
	
The existing security mechanisms in the Session Initiation Protocol
   are inadequate for cryptographically assuring the identity of the end
   users that originate SIP requests and responses, especially in an
   interdomain context.  This document recommends practices and
   conventions for identifying end users in SIP messages, and proposes a
   way to distribute cryptographically secure authenticated identities.

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
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	Title		: Update to the Session Initiation Protocol (SIP) 
			  Preconditions Framework
	Author(s)	: G. Camarillo, P. Kyzivat
	Filename	: draft-ietf-sip-rfc3312-update-03.txt
	Pages		: 10
	Date		: 2004-9-29
	
This document updates the framework for preconditions in SIP. We
provide guidelines for authors of new precondition types and describe
how to use SIP preconditions in situations that involve session
mobility.

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From sip-bounces@ietf.org  Wed Sep 29 19:12:35 2004
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From: "Shan Lu" <shanlu@sentito.com>
To: "'IETF SIP Mailing List'" <sip@ietf.org>
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Subject: [Sip] Question on RFC3261 Merged Requests
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Hi,

RFC3261 has following text:

8.2.2.2 Merged Requests

   If the request has no tag in the To header field, the UAS core MUST
   check the request against ongoing transactions.  If the From tag,
   Call-ID, and CSeq exactly match those associated with an ongoing
   transaction, but the request does not match that transaction (based
   on the matching rules in Section 17.2.3), the UAS core SHOULD
   generate a 482 (Loop Detected) response and pass it to the server
   transaction.

Assume that calls to support@example.com is forked to two phone numbers on
PSTN, say, 555-1111 and 555-2222. Further assume that there is only one
SIP-PSTN gateway. The two INVITEs arriving at the gw are identical except
for R-URI userinfo and top VIA branch (assuming forking proxy is the only
proxy). 

My question is: why should GW 482 second INVITE? Isn't that the purpose of
forking? Should 8.2.2.2 be modified as:

... If request URI, the From tag, Call-ID, and CSeq exactly match ...
       ^^^^^^^^^^^^ 

Regards, 

Shan Lu
sentitO Networks


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From sip-bounces@ietf.org  Wed Sep 29 19:43:48 2004
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As promised, I've released a new version of the SIP identity draft that
removes all text relating to response identity - and as such, there's no
more talk in the draft about restricting retargeting or what have you. Given
the many dependencies on request identity, I think it's worth getting this
part done first, and then revisiting response identity at a later time.

I also made a few more changes to the draft. The most obvious will be the
addition of examples and a corresponding bit-exact archive of the exact
messages (following the practice of Robert Sparks' torture test draft). The
hope is that they will significantly increase the likelihood that the draft
will be implemented. I also added new sections giving considerations on
privacy and the use of telephone numbers as identities. There are also some
subtler changes to the mechanism itself to address a few lingering threats.

I think that the draft is in a fairly mature state now, and I solicited a
number of detailed reviews that were factored into this version. I'd
appreciate it if interested parties could give it a read.

Thanks,

Jon Peterson
NeuStar, Inc.


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From sip-bounces@ietf.org  Thu Sep 30 00:13:52 2004
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From: "Dan Wing" <dwing@cisco.com>
To: "Vikram Bhuskute" <vikramb@aftek.com>, <sip@ietf.org>
Subject: RE: [Sip] draft-takeda-symmetric-nat-traversal  ?? 
Date: Wed, 29 Sep 2004 21:09:30 -0700
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Thanks for bringing that I-D to the group's attention.  I, for one, hadn't
seen it.

Section 8.1 is where the meat is, and it says:

   ...
   If
   the port allocation rule of the other endpoint is a symmetric type,
   the endpoint needs either to record the source port number of an
   incoming packet or to predict a port number that the symmetric NAT
   will assign, with a MAPPED-ADDRESS obtained from the other endpoint.
   ...

which I agree isn't a good mechanism, especially with a large NAT.  It
could work just fine on a small, residential NAT which isn't processing
much new traffic (and thus it is, in theory, possible to guess the next
port number it will assign for the next mapping).

This technique is too fragile for my blood.


There is a lot of other good information in this Internet Draft, though;
I hope we can incorporate some of it into the BEHAVE work if the author
is willing.

-d

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
> Vikram Bhuskute
> Sent: Wednesday, September 29, 2004 4:12 AM
> To: sip@ietf.org
> Subject: [Sip] draft-takeda-symmetric-nat-traversal ??
>
>
> Hello all,
>             While searching for NAT traversal solution i came across a
> draft  "draft-takeda-symmetric-nat-traversal-00.txt"
>             from url (
> http://cnscenter.future.co.kr/resource/ietf/ind-draft/draft-takeda
> -symmetric-nat-traversal-00.txt)
>
>            It proposes some solution for Symmetric NAT based on some
> Prediction .....?  It dosent seem reliable ... !!!
>           (however it gives  glimps on type of Symmetric NATs )
>
>            Is it still a matter of further research ? or the concept  is
> not well accepted ?
>
>           thanks in advance
>
>  vikram
>
> _______________________________________________
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> 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 sip-bounces@ietf.org  Thu Sep 30 01:34:12 2004
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Michael Zhou'" <xmzhou@cisco.com>
Subject: RE: [Sip] Question on offer-answer during establishment
Date: Thu, 30 Sep 2004 07:29:54 +0200
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Hi,

>>At the previous SIPit we had some discussions about the 
>>usage of PRACK for offer/answer. The problem, as said before, 
>>is that there is currently no standard way to reject an offer 
>>received in a PRACK, because a UAS MUST answer with 200 OK, 
>>if the PRACK otherwise is ok, matches a reliabel provisional 
>>response etc. It has even been suggested that it should be 
>>forbidden to send SDPs (at least new offers) in PRACKs, but 
>>that would of course have impacts on existing nodes - 
>>depending on how many nodes currently support it in the first 
>>place, of course.
>>  
>>
>It's not mentioned in RFC 3262 how to reject an offer, but 488 (Not 
>Acceptable Here) might be a natural choice according to 
>Section 13.3.1.3 of RFC 3261, quoted below. In practice, 4xx-6xx respons to 
>PRACK (which contains offer) may also be considered rejection of offer.
> 
>"
> 
>    A UAS rejecting an offer contained in an INVITE SHOULD return a 488
>    (Not Acceptable Here) response.  Such a response SHOULD include a
>    Warning header field value explaining why the offer was rejected.
> 
>"

You are right. But, at the same time the PRACK RFC says that 200 MUST be sent, and I think is also the conclusion from previous discussions on the list.

I know about implementations that reject the offer by simply don't inserting an answer in the 200 OK. No, you don't have to tell me it's non-standard behaviour :)

Regards,

Christer Holmberg
Ericsson Finland 

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From sip-bounces@ietf.org  Thu Sep 30 01:51:22 2004
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Shan Lu'" <shanlu@sentito.com>, "'IETF SIP Mailing List'" <sip@ietf.org>
Subject: RE: [Sip] Question on RFC3261 Merged Requests
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Hi,

Interesting point.

One could argue that the existing text is ok for a normal SIP phone, because there is really no idea in receiving multiple "calls" (multiple "instances" of the same forked INVITE) from the same caller (in most cases the SIP phone can handle only one call at a time anyway, no matter if the incoming INVITEs origin from the same user or different users).

However, for a gateway that may not be a valid case. Even if the gateway from a SIP perspective is a single SIP endpoint there are multiple end-users behind it, so IF we want to allow "forking" into the PSTN network (or whatever is on the other side) you are correct, I think.

The problem I have with merged requests is that it takes away the advantages of only using the RFC3261 Via branch cookie from transaction identification. A node compliant with RFC3261 will basically still have to use the "old" transaction mapping mechanism, in order the to the merge check...

Regards,

Christer Holmberg
Ericsson Finland




> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
> Shan Lu
> Sent: 30. syyskuuta 2004 1:22
> To: 'IETF SIP Mailing List'
> Subject: [Sip] Question on RFC3261 Merged Requests
> 
> 
> Hi,
> 
> RFC3261 has following text:
> 
> 8.2.2.2 Merged Requests
> 
>    If the request has no tag in the To header field, the UAS core MUST
>    check the request against ongoing transactions.  If the From tag,
>    Call-ID, and CSeq exactly match those associated with an ongoing
>    transaction, but the request does not match that transaction (based
>    on the matching rules in Section 17.2.3), the UAS core SHOULD
>    generate a 482 (Loop Detected) response and pass it to the server
>    transaction.
> 
> Assume that calls to support@example.com is forked to two 
> phone numbers on
> PSTN, say, 555-1111 and 555-2222. Further assume that there 
> is only one
> SIP-PSTN gateway. The two INVITEs arriving at the gw are 
> identical except
> for R-URI userinfo and top VIA branch (assuming forking proxy 
> is the only
> proxy). 
> 
> My question is: why should GW 482 second INVITE? Isn't that 
> the purpose of
> forking? Should 8.2.2.2 be modified as:
> 
> ... If request URI, the From tag, Call-ID, and CSeq exactly match ...
>        ^^^^^^^^^^^^ 
> 
> Regards, 
> 
> Shan Lu
> sentitO Networks
> 
> 
> _______________________________________________
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> 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 sip-bounces@ietf.org  Thu Sep 30 09:40:43 2004
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Also in RFC and draft, there is a similar example.

draft-ietf-sip-session-timer-15
	Route: sips:p1.atlanta.example.com;lr

RFC 3329
	Route: sip:callee@domain.com

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From sip-bounces@ietf.org  Thu Sep 30 12:59:19 2004
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From: "Elwell, John" <john.elwell@siemens.com>
To: "'sip@ietf.org'" <sip@ietf.org>
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Subject: [Sip] Proposed bug fix to RFC3261 arising from
	draft-elwell-sip-state-u pdate-02
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At the SIP meeting in San Diego, I was asked to raise on the mailing list
the individual bug fixes proposed in the state-update draft, so that each
can be given due scrutiny. I decided to start with RFC3261. Here only a few
changes are proposed and it does not seem sensible to consider these
separately, so I propose them here as a single bug fix. If we can gain
consensus on those, I will follow it up with messages proposing bug fixes to
the other documents (mainly RFC3311).

As a result of comments by Paul Kyzivat raised on the list recently, I have
modified the proposed changes slightly. In particular, I have dropped any
headers that are not obviously dialog state changes, including those that
might not be considered state at all (e.g., Alert-Info) and those that might
be considered call state rather than dialog state (e.g., Call-Info,
Subject). If anyone wants these re-instating, please say so.

Proposed bug fix to RFC3261:

Add the following paragraph to the end of 12 (prior to 12.1):

"A dialog can also have certain other state information that can change
during the lifetime of the dialog. This includes the session description (as
negotiated by SDP offer/answer), information from certain other headers and
information from certain other bodies. There are no examples from this RFC.
However, extensions may define other headers or bodies containing
information that contributes to the state of a dialog and can change during
the lifetime of the dialog."

Add two new paragraphs to the end of 12.2 (prior to 12.2.1):

"A request, whether or not it is a target refresh request, MAY update
certain other information transmitted at the time of dialog establishment.
Although there are no examples in this RFC, other extensions may define
headers or bodies that are appropriate for updating during a dialog.

"In this RFC, the only request defined that is suitable for updating
information during a dialog is re-INVITE (see section 14). Other extensions
may define different requests suitable for updating information during a
dialog."

Add to the end of the first paragraph of 14.1:

"A UAC MAY send a session description that is unchanged, if the purpose of
the re-INVITE is solely for updating dialog-related information."

John Elwell (john.elwell@siemens.com)

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Date: Thu, 30 Sep 2004 22:54:05 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: "Elwell, John" <john.elwell@siemens.com>
Subject: Re: [Sip] Proposed bug fix to RFC3261 arising
	from	draft-elwell-sip-state-u pdate-02
References: <50B1CBA96870A34799A506B2313F26670252D417@ntht201e>
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John,

I agree with what you are trying to do. But one rather significant 
detail with what you propose is that I don't believe the session 
description (SDP) is dialog state. It is call state. (A dialog that 
exists for the purpose of a SUBSCRIBE would not have this.)

That in some sense guts this proposal, since now there are no examples 
in this document of additional dialog state. However I still think the 
clarification is needed.

In addition, 3261 sweeps the distinction between dialog state and call 
state under the rug. Once alternative uses of dialogs are in play it is 
important to make this distinction clear, and to identify those items 
that are part of call state. This includes the session description info 
(the negotiated offer and answer as well as potentially another proposed 
change to offer and answer). It also includes the data associated with 
session timer, and possibly Call-Info, Alert-Info, Subject. (Which if 
any of these is truly call state remains to be discussed.)

	Paul

Elwell, John wrote:
> At the SIP meeting in San Diego, I was asked to raise on the mailing list
> the individual bug fixes proposed in the state-update draft, so that each
> can be given due scrutiny. I decided to start with RFC3261. Here only a few
> changes are proposed and it does not seem sensible to consider these
> separately, so I propose them here as a single bug fix. If we can gain
> consensus on those, I will follow it up with messages proposing bug fixes to
> the other documents (mainly RFC3311).
> 
> As a result of comments by Paul Kyzivat raised on the list recently, I have
> modified the proposed changes slightly. In particular, I have dropped any
> headers that are not obviously dialog state changes, including those that
> might not be considered state at all (e.g., Alert-Info) and those that might
> be considered call state rather than dialog state (e.g., Call-Info,
> Subject). If anyone wants these re-instating, please say so.
> 
> Proposed bug fix to RFC3261:
> 
> Add the following paragraph to the end of 12 (prior to 12.1):
> 
> "A dialog can also have certain other state information that can change
> during the lifetime of the dialog. This includes the session description (as
> negotiated by SDP offer/answer), information from certain other headers and
> information from certain other bodies. There are no examples from this RFC.
> However, extensions may define other headers or bodies containing
> information that contributes to the state of a dialog and can change during
> the lifetime of the dialog."
> 
> Add two new paragraphs to the end of 12.2 (prior to 12.2.1):
> 
> "A request, whether or not it is a target refresh request, MAY update
> certain other information transmitted at the time of dialog establishment.
> Although there are no examples in this RFC, other extensions may define
> headers or bodies that are appropriate for updating during a dialog.
> 
> "In this RFC, the only request defined that is suitable for updating
> information during a dialog is re-INVITE (see section 14). Other extensions
> may define different requests suitable for updating information during a
> dialog."
> 
> Add to the end of the first paragraph of 14.1:
> 
> "A UAC MAY send a session description that is unchanged, if the purpose of
> the re-INVITE is solely for updating dialog-related information."
> 
> John Elwell (john.elwell@siemens.com)
> 
> _______________________________________________
> Sip mailing list  https://www1.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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