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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: [Sip] Agenda requests for SF
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Folks,

we are going to start gathering agenda requests for the next IETF
meeting in San Francisco. Please, let the chairs know if you would like
to request a slot.

Thanks,

Gonzalo
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Subject: [Sip] I-D ACTION:draft-ietf-sip-content-indirect-mech-02.txt
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--NextPart

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

	Title		: A Mechanism for Content Indirection in Session 
                          Initiation Protocol (SIP) Messages
	Author(s)	: S. Olson
	Filename	: draft-ietf-sip-content-indirect-mech-02.txt
	Pages		: 15
	Date		: 2003-3-3
	
This document proposes an extension to the URL MIME External- Body
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These extensions are aimed at allowing any MIME part in a SIP message
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From mailnull@www1.ietf.org  Tue Mar  4 09:14:25 2003
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Hello!

Is it possible to use multicast for sending NOTIFY requests to the
subscribers? I want to send the notifications very fast and it takes
some time to do the sending one after another.

Regards,
Klaus Darilion
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From mailnull@www1.ietf.org  Tue Mar  4 10:40:31 2003
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From: "Valerie Kenneally" <vkenneally@cit.ie>
To: <sip@ietf.org>, <sipping@ietf.org>
Cc: <alan.johnston@wcom.com>, <sdonovan@dynamicsoft.com>,
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Subject: [Sip] Question on Basic Call flows document section 3.6
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Hi All,

	In document at the below link

http://www.ietf.org/internet-drafts/draft-ietf-sipping-basic-call-flows-01.t
xt
section 3.6 "Session via Redirect and Proxy Servers with the SDP in ACK".

I am wondering if someone can explain to me how the INVITE (F4) is directed
from User A to Proxy 3 without any Route header being included in the
message.

I have been trying to figure this out for a while and I'd really appreciate
any feedback asap

Thanks in advance,
Val

------------------------
Val Kenneally,
Post-Graduate,
Electronic Eng. Dept.,
CIT,
Rossa Ave.,
Bishopstown,
Cork,
Ireland.
------------------------
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Fax: +353 21 4326625


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From mailnull@www1.ietf.org  Tue Mar  4 14:27:24 2003
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Date: Tue, 4 Mar 2003 11:25:43 -0800 (PST)
From: Sean Olson <seancolson@yahoo.com>
Subject: Re: [Sip] Event Notifications with Multicast
To: Klaus Darilion <darilion@ict.tuwien.ac.at>, sip@ietf.org
In-Reply-To: <1FE28144CBED9F4C8959A2D25CDDF02706D1E3@idefix.ict.tuwien.ac.at>
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Intriguing idea, but it has some issues. Namely,
what would you put in the From and Call-ID
for such a NOTIFY request that was destined to
multiple subscribers (who had issued SUBSCRIBE 
requests with different From and Call-ID values)

Note that there is no inherent reason you could
not send these NOTIFY requests in parallel. That
is more of an implementation issue than a protocol
one.

Regards,
Sean Olson
Microsoft


--- Klaus Darilion <darilion@ict.tuwien.ac.at> wrote:
> Hello!
> 
> Is it possible to use multicast for sending NOTIFY
> requests to the
> subscribers? I want to send the notifications very
> fast and it takes
> some time to do the sending one after another.
> 
> Regards,
> Klaus Darilion
> _______________________________________________
> 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 mailnull@www1.ietf.org  Tue Mar  4 18:34:17 2003
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Date: Wed, 5 Mar 2003 00:33:06 +0100 (CET)
From: Jiri Kuthan <jiri@iptel.org>
To: <sip@ietf.org>
cc: <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sip] manyfolks comment
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I think it would be useful to mention the special-case of forking in the
manyfolks draft. I think it is not entirely obvious and readers would
be certainly pleased if the document included a how-to on that.

imho, it takes initiating a PRACK transaction (and assuring preconditions)
to each destination which replied with 18x.

I'm not yet sure, what CSeq to set for all the PRACKs. Obviously, all UASs 
deserve CSeq(invite)+1 but the idea of sending multiple "parallel" transactions 
with the same CSeq is still surprising to me. 

Comments?

Thanks,

-Jiri

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Subject: [Sip] RFC 3485 on The Session Initiation Protocol (SIP) and Session Description Protocol (SDP) Static Dictionary for Signaling Compression (SigComp)
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--NextPart


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


        RFC 3485

        Title:      The Session Initiation Protocol (SIP) and Session
                    Description Protocol (SDP) Static Dictionary for
                    Signaling Compression (SigComp)
        Author(s):  M. Garcia-Martin, C. Bormann, J. Ott, R. Price,
                    A. B. Roach
        Status:     Standards Track
        Date:       February 2003
        Mailbox:    miguel.a.garcia@ericsson.com, cabo@tzi.org,
                    jo@tzi.org, richard.price@roke.co.uk,
                    adam@dynamicsoft.com
        Pages:      30
        Characters: 80195
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-sipping-sigcomp-sip-dictionary-05.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3485.txt


The Session Initiation Protocol (SIP) is a text-based protocol for
initiating and managing communication sessions.  The protocol can be
compressed by using Signaling Compression (SigComp).  Similarly, the
Session Description Protocol (SDP) is a text-based protocol intended
for describing multimedia sessions for the purposes of session
announcement, session invitation, and other forms of multimedia
session initiation.  This memo defines the SIP/SDP-specific static
dictionary that SigComp may use in order to achieve higher efficiency.
The dictionary is compression algorithm independent.

This document is a product of the Session Initiation Proposal
Investigation Working Group and the Session Initiation 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
of this memo is unlimited.

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Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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Subject: [Sip] Re: manyfolks comment
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Jiri,

> imho, it takes initiating a PRACK transaction (and assuring preconditions)
> to each destination which replied with 18x.

Yes, if you want to use preconditions, this is what you will most likely
do (assuming that you can reserve so much resources).

 
> I'm not yet sure, what CSeq to set for all the PRACKs. Obviously, all UASs
> deserve CSeq(invite)+1 but the idea of sending multiple "parallel" transactions
> with the same CSeq is still surprising to me.

The Cseq number spaces of different dialogs are independent. Even if all
the dialogs share the same Call-ID, they are still independent.
Therefore, there is nothing weird in sending the same Cseq to different
destinations.

If, in the future, we review the manyfolks spec, it may be worthwhile
mentioning that forking can interact with manyfolks... although I do not
see any big technical issue there.

Thanks for the comments,

Gonzalo
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From mailnull@www1.ietf.org  Wed Mar  5 04:54:03 2003
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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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At 10:04 AM 3/5/2003, Gonzalo Camarillo wrote:
>If, in the future, we review the manyfolks spec, it may be worthwhile
>mentioning that forking can interact with manyfolks... although I do not
>see any big technical issue there.

I agree -- it just did not seem so intuitively understandable and thus
worth mentioning to me.

-Jiri 

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From mailnull@www1.ietf.org  Wed Mar  5 13:59:46 2003
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From: "Klaus Darilion" <darilion@ict.tuwien.ac.at>
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I know that won't an easy solution. My first idea was to use unicast for
the subscription request, the subscription respond and the notification
respond, and multicast only for the NOTIFY request. In the subscription
response, the notifiyer has to tell the subscriber which multicast
address it should bind to.

The From: headers in the SUBSCRIBE request need not to be changed. The
To: header can have some generic name like
"subscriber@multicast-IP-address". The Call-ID will probably be
different as the one in the subscription. So, subscribers has to use the
Call-ID they receive in the first notification (I think this is similar
to subscriptions without a SUBSCRIBE transaction). I found 2 solutions
to differentiate the responses: Either the subscriber rewrites the To:
header to its own sip address or the subscriber inserts an header with
its sip-uri.

NOTIFY retransmission should be sent by unicast to avoid annoying of the
subscribers which had responded to the notification.

I had the idea while I analyzed the presence capability of KPhone.
KPhone (respectively dissipate) takes 2ms to send a notification (1GHz
PC), but I want to notify a lot of subscribers (50 ore more) within a
very short time (~20ms). So I thought of multicast. (I know, another way
would be to use a faster SIP stack.)

Regards,
Klaus Darilion


> -----Original Message-----
> From: Sean Olson [mailto:seancolson@yahoo.com] 
> Sent: Tuesday, March 04, 2003 8:26 PM
> To: Klaus Darilion; sip@ietf.org
> Subject: Re: [Sip] Event Notifications with Multicast
> 
> 
> Intriguing idea, but it has some issues. Namely,
> what would you put in the From and Call-ID
> for such a NOTIFY request that was destined to
> multiple subscribers (who had issued SUBSCRIBE 
> requests with different From and Call-ID values)
> 
> Note that there is no inherent reason you could
> not send these NOTIFY requests in parallel. That
> is more of an implementation issue than a protocol
> one.
> 
> Regards,
> Sean Olson
> Microsoft
> 
> 
> --- Klaus Darilion <darilion@ict.tuwien.ac.at> wrote:
> > Hello!
> > 
> > Is it possible to use multicast for sending NOTIFY
> > requests to the
> > subscribers? I want to send the notifications very
> > fast and it takes
> > some time to do the sending one after another.
> > 
> > Regards,
> > Klaus Darilion _______________________________________________
> > 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: "paopai" <swifthorse@hotgroup.org>
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sip��艇挫��

	

　　　　　　　　崑
撰��
 				

　　　　　　　　paopai
　　　　　　　　swifthorse@hotgroup.org
　　　　　　　　　　2003-03-06



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Subject: [Sip] I-D ACTION:draft-ietf-sip-mib-05.txt
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--NextPart

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

	Title		: Management Information Base for Session Initiation 
                          Protocol
	Author(s)	: K. Lingle, J. Maeng, J. Mule, D. Walker
	Filename	: draft-ietf-sip-mib-05.txt
	Pages		: 97
	Date		: 2003-3-6
	
This memo defines a portion of the Management Information Base (MIB) 
for use with network management protocols in the Internet community.  
In particular, it describes a set of managed objects that are used 
to manage Session Initiation Protocol (SIP) entities, which include 
User Agents, Proxy servers, Redirect servers and Registrars.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-05.txt

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_______________________________________________
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From mailnull@www1.ietf.org  Fri Mar  7 07:39:40 2003
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: Robert Sparks <rsparks@dynamicsoft.com>
CC: sip@ietf.org
Subject: Re: [Sip] I-D ACTION:draft-sparks-sip-noninvite-00.txt
References: <200302111144.GAA28328@ietf.org>
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Robert, 

a couple of comments on the draft.

At the end of Section 5.1, you say that SUBSCRIBE "proofs that another
option exists". However, SUBSCRIBE/NOTIFY does not resolve the HERP
problem (e.g., one of the UASs wants to challenge the request and the
other does not).

In Section 6, you mention: 

"ACK is not needed for this pending non-INVITE because we have removed
   unreliable transports.  (ACK was originally needed for INVITE because
   reliability over UDP became the server's responsibility after its
   first response and the server needed to know when to stop
   retransmitting."

Reliability was not the only (original) purpose of ACK. If you receive a
request and respond immediately, you can assume that the client is still
there. However, if you receive a request, take a long time doing whaever
processing is needed, and then answer, you want to be sure that the
client is still there. The ACK confirms that the client has not died or
given up in the meantime. It may be good to document this in the draft.

Thanks,

Gonzalo



Internet-Drafts@ietf.org wrote:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> 
>         Title           : Considerations for the Session Initiation Protocol's
>                           non-INVITE Transaction
>         Author(s)       : R. Sparks
>         Filename        : draft-sparks-sip-noninvite-00.txt
>         Pages           : 14
>         Date            : 2003-2-10
> 
> This draft explores several issues with the Session Initiation
> Protocol's non-INVITE transaction.  It focuses on the use of
> provisional responses and on problems related to transaction
> timeouts.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-sparks-sip-noninvite-00.txt
> 
> To remove yourself from the IETF Announcement list, send a message to
> ietf-announce-request with the word unsubscribe in the body of the message.
> 
> Internet-Drafts are also available by anonymous FTP. Login with the username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
>         "get draft-sparks-sip-noninvite-00.txt".
> 
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
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> 
> NOTE:   The mail server at ietf.org can return the document in
>         MIME-encoded form by using the "mpack" utility.  To use this
>         feature, insert the command "ENCODING mime" before the "FILE"
>         command.  To decode the response(s), you will need "munpack" or
>         a MIME-compliant mail reader.  Different MIME-compliant mail readers
>         exhibit different behavior, especially when dealing with
>         "multipart" MIME messages (i.e. documents which have been split
>         up into multiple messages), so check your local documentation on
>         how to manipulate these messages.
> 
> 
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> 
>   ------------------------------------------------------------------------

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
_______________________________________________
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--NextPart

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

	Title		: The Session Inititation Protocol (SIP) 'Join' Header
	Author(s)	: R. Mahy, D. Petrie
	Filename	: draft-ietf-sip-join-01.txt
	Pages		: 17
	Date		: 2003-3-6
	
This document defines a new header for use with SIP multi-party
applications and call control.  The Join header is used to logically
join an existing SIP dialog with a new SIP dialog.  This primitive
can be used to enable a variety of features, for example: 'Barge-In',
answering-machine-style 'Message Screening' and 'Call Center
Monitoring'.  Note that definition of these example features is non-
normative.

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--NextPart

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

	Title		: The Session Inititation Protocol (SIP) 'Replaces' Header
	Author(s)	: B. Biggs, R. Dean, R. Mahy
	Filename	: draft-ietf-sip-replaces-03.txt
	Pages		: 17
	Date		: 2003-3-6
	
This document defines a new header for use with 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 'Retrieve from Call Park'.  Note that
definition of these example features is non-normative.

A URL for this Internet-Draft is:
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Below is the data which will enable a MIME compliant mail reader
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Date: Fri, 7 Mar 2003 14:25:11 -0600
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Subject: [Sip] SIP IETF56 Web Site live and agenda requests posted
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See http://www.softarmor.com/sipwg/meets/ietf56/requests.html

For the list of requested agenda topics so far (and send me anything I'm
missing)

And of course, general meeting info will be at

http://www.softarmor.com/sipwg/meets/ietf56/


Thanks,


--
Dean



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Fri Mar  7 15:42:46 2003
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From: Internet-Drafts@ietf.org
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Date: Fri, 07 Mar 2003 15:38:45 -0500
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--NextPart

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

	Title		: SIP Authenticated Identity Body (AIB) Format
	Author(s)	: J. Peterson
	Filename	: draft-ietf-sip-authid-body-01.txt
	Pages		: 11
	Date		: 2003-3-7
	
RFC3261 introduces the concept of adding an S/MIME body to a 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.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-authid-body-01.txt

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	"multipart" MIME messages (i.e. documents which have been split
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Below is the data which will enable a MIME compliant mail reader
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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



From mailnull@www1.ietf.org  Fri Mar  7 16:00:03 2003
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From: Internet-Drafts@ietf.org
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Date: Fri, 07 Mar 2003 15:38:28 -0500
Subject: [Sip] I-D ACTION:draft-ietf-sip-callerprefs-08.txt
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--NextPart

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

	Title		: Caller Preferences and Callee Capabilities 
			  for the Session Initiation Protocol (SIP)
	Author(s)	: H. Schulzrinne, J. Rosenberg, P. Kyzivat
	Filename	: draft-ietf-sip-callerprefs-08.txt
	Pages		: 58
	Date		: 2003-3-7
	
This document describes a set of extensions to the Session Initiation
Protocol (SIP) which allow a caller to express preferences about
request handling in servers. These preferences include the ability to
select which URIs a request gets routed to, and to specify certain
request handling directives in proxies and redirect servers. It does
so by defining four new request headers, Accept-Contact, Reject-
Contact, Require-Contact and Request-Disposition, which specify the
caller's preferences. The extension also defines new parameters for
the Contact header that describe the capabilities and characteristics
of a UA.

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

To remove yourself from the IETF Announcement list, send a message to 
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NOTE:	The mail server at ietf.org can return the document in
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	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
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Internet-Draft.

--NextPart
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--OtherAccess
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Content-Type: text/plain
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ENCODING mime
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--OtherAccess
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	name="draft-ietf-sip-callerprefs-08.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
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--OtherAccess--

--NextPart--


_______________________________________________
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 mailnull@www1.ietf.org  Fri Mar  7 16:00:48 2003
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To: IETF-Announce: ;
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From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Fri, 07 Mar 2003 15:38:36 -0500
Subject: [Sip] I-D ACTION:draft-ietf-sip-identity-01.txt
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--NextPart

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

	Title		: Enhancements for Authenticated Identity Management 
			  in the Session Initiation Protocol (SIP)
	Author(s)	: J. Peterson
	Filename	: draft-ietf-sip-identity-01.txt
	Pages		: 16
	Date		: 2003-3-7
	
The existing mechanisms for expressing identity in the Session
Initiation Protocol oftentimes do not permit an administrative domain
to verify securely the identity of the originator of a request.  This
document recommends practices and conventions for authenticating end
users, and proposes a way to distribute cryptographically secure
authenticated identities within SIP messages.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-identity-01.txt

To remove yourself from the IETF Announcement list, send a message to 
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	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
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FILE /internet-drafts/draft-ietf-sip-identity-01.txt

--OtherAccess
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	name="draft-ietf-sip-identity-01.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<2003-3-7151928.I-D@ietf.org>

--OtherAccess--

--NextPart--


_______________________________________________
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 mailnull@www1.ietf.org  Mon Mar 10 07:17:05 2003
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Date: Mon, 10 Mar 2003 11:32:57 +0100
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Thread-Topic: Push-To-Talk with SIP
Thread-Index: AcLm8Gpzm0q0+z4GSfm02npdYLWDfg==
From: "Klaus Darilion" <darilion@ict.tuwien.ac.at>
To: <sip@ietf.org>
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Subject: [Sip] Push-To-Talk with SIP
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Hello!

Nokia, Siemens and Ericsson have announced that they are developing a
push-to-talk application for GSM networks with GPRS and for future 3G
networks. They want to use SIP for signaling but I couldn't find any
more detailed information.

Does someone of you know some details about the PTT signaling with SIP?
Do they care about arbitration if two people want to talk to the same
group at the same time? Are there some drafts or discussions about this
topics?

With kind regards,
Klaus Darilion
_______________________________________________
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 mailnull@www1.ietf.org  Mon Mar 10 08:52:27 2003
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From: "Burnside, Andrew" <Andrew.Burnside@thalesgroup.com>
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Hello

I am currently looking at options for providing QoS managed inter-domain
connections using SIP. This will probably be in a DiffServ architecture,
with aggregated flows between domains where possible. Has there been much
work done in this area with SIP? 

I saw that Henry Sinnreich had an Internet Draft out on this topic back in
2000, though I can't find any references to subsequent work.  

Regards

Andrew Burnside
_______________________________________________
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: Li Li <Li.Li@SS8.com>
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Subject: [Sip] SIP Privacy draft
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Can anybody help me to understand what happened to the 
privay draft that had the remote-party-ID header? Is 
it completely obsoleted? Or is it moved to some other 
draft? RFC 3323 does not include such header operation 
mechanism. 

Thanks,

Li Li


> -----Original Message-----
> From: Burnside, Andrew [mailto:Andrew.Burnside@thalesgroup.com]
> Sent: Monday, March 10, 2003 8:49 AM
> To: 'sip@ietf.org'
> Subject: [Sip] Inter-domain QoS with SIP
> 
> 
> Hello
> 
> I am currently looking at options for providing QoS managed 
> inter-domain
> connections using SIP. This will probably be in a DiffServ 
> architecture,
> with aggregated flows between domains where possible. Has 
> there been much
> work done in this area with SIP? 
> 
> I saw that Henry Sinnreich had an Internet Draft out on this 
> topic back in
> 2000, though I can't find any references to subsequent work.  
> 
> Regards
> 
> Andrew Burnside
> _______________________________________________
> 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



From mailnull@www1.ietf.org  Mon Mar 10 09:49:03 2003
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Date: Mon, 10 Mar 2003 08:22:53 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] Inter-domain QoS with SIP
In-reply-to: <51BF576D5A02CC4CB2591F50994FD76614E7E6@NTS013.uk.trt.thales>
To: "'Burnside, Andrew'" <Andrew.Burnside@thalesgroup.com>, sip@ietf.org
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Please see for interdomain QOS:

http://www.ietf.org/internet-drafts/draft-johnston-sip-osp-token-04.txt

It also has the updated interdomain model.

Thanks, Henry

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Burnside, Andrew
> Sent: Monday, March 10, 2003 7:49 AM
> To: 'sip@ietf.org'
> Subject: [Sip] Inter-domain QoS with SIP
> 
> 
> Hello
> 
> I am currently looking at options for providing QoS managed 
> inter-domain connections using SIP. This will probably be in 
> a DiffServ architecture, with aggregated flows between 
> domains where possible. Has there been much work done in this 
> area with SIP? 
> 
> I saw that Henry Sinnreich had an Internet Draft out on this 
> topic back in 2000, though I can't find any references to 
> subsequent work.  
> 
> Regards
> 
> Andrew Burnside
> _______________________________________________
> 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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From: "Cullen Jennings" <fluffy@cisco.com>
To: "'Li Li'" <Li.Li@ss8.com>, <sip@ietf.org>
Subject: RE: [Sip] SIP Privacy draft
Date: Mon, 10 Mar 2003 08:22:51 -0800
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It became RFC 3325

http://www.ietf.org/rfc/rfc3325.txt


> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Li Li
> Sent: Monday, March 10, 2003 6:39 AM
> To: 'sip@ietf.org'
> Subject: [Sip] SIP Privacy draft
> 
> 
> Can anybody help me to understand what happened to the 
> privay draft that had the remote-party-ID header? Is 
> it completely obsoleted? Or is it moved to some other 
> draft? RFC 3323 does not include such header operation 
> mechanism. 
> 
> Thanks,
> 
> Li Li
> 
> 
> > -----Original Message-----
> > From: Burnside, Andrew [mailto:Andrew.Burnside@thalesgroup.com]
> > Sent: Monday, March 10, 2003 8:49 AM
> > To: 'sip@ietf.org'
> > Subject: [Sip] Inter-domain QoS with SIP
> > 
> > 
> > Hello
> > 
> > I am currently looking at options for providing QoS managed
> > inter-domain
> > connections using SIP. This will probably be in a DiffServ 
> > architecture,
> > with aggregated flows between domains where possible. Has 
> > there been much
> > work done in this area with SIP? 
> > 
> > I saw that Henry Sinnreich had an Internet Draft out on this
> > topic back in
> > 2000, though I can't find any references to subsequent work.  
> > 
> > Regards
> > 
> > Andrew Burnside _______________________________________________
> > 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
> 

_______________________________________________
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: Adam Roach <adam@dynamicsoft.com>
To: "'Jiri Kuthan'" <jiri@iptel.org>, sip@ietf.org
Cc: Gonzalo.Camarillo@ericsson.com
Subject: RE: [Sip] manyfolks comment
Date: Mon, 10 Mar 2003 11:35:02 -0600
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> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri@iptel.org]
> 
> I'm not yet sure, what CSeq to set for all the PRACKs. 
> Obviously, all UASs 
> deserve CSeq(invite)+1 but the idea of sending multiple 
> "parallel" transactions 
> with the same CSeq is still surprising to me. 

Keep in mind:

- CSeqs are scoped to dialogs
- Dialogs are identified by the set (call-id, local-tag, remote-tag)
- 1xx responses from different sources to a forked request will
  have remote-tags unique to the source

/a
_______________________________________________
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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
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   SIP                                                      H. Khartabil 
   Internet Draft                                                  Nokia 
   Expires: Septemper 2, 2003                              March 3, 2003 
                                                                         
    
    
               Congestion safety and Content Indirection 
              draft-khartabil-sip-congestionsafe-ci-02.txt 
    
    
STATUS OF THIS MEMO 
    
   This document is an Internet-Draft and is in full conformance with 
   all provisions of Section 10 of RFC2026. 
    
   Internet-Drafts are working documents of the Internet Engineering 
   Task Force (IETF), its areas, and its working groups.  Note that 
   other groups may also distribute working documents as Internet-
   Drafts. 
    
   Internet-Drafts are draft documents valid for a maximum of six 
   months and may be updated, replaced, or obsoleted by other documents 
   at any time.  It is inappropriate to use Internet- Drafts a 
   reference material or to cite them other than as work in progress. 
    
   The list of current Internet-Drafts can be accessed at 
   http://www.ietf.org/ietf/1id-abstracts.txt 
    
   The list of Internet-Draft Shadow Directories can be accessed at 
   http://www.ietf.org/shadow.html. 
    
   This Internet Draft will expire on April 23, 2003. 
    
    
Copyright Notice 
    
   Copyright (C) The Internet Society (2002). All Rights Reserved. 
    
    
Abstract 
    
   The Content-Indirection service has many uses. This document 
   describes this service by combining congestion safely and Content-
   Indirection Mechanism into the one document and presents scenarios 
   where the 2 could be combined. It also introduces extensions to SIP 
   that allows end points to specify their maximum acceptable message 
   size. 
 
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Table of Contents 
 
1.0 Terminology.......................................................3 
2.0 Introduction......................................................3 
3.0 Overview of Operation.............................................4 
4.0 UA Behaviour......................................................5 
4.1 UAC Sending Requests..............................................5 
4.2 UAC Behaviour when receiving Response.............................5 
4.2.1 �413� or �513� Error Responses..................................5 
4.3 UA Registering Desired Maximum SIP Message Size...................6 
4.4 UAS Receiving A SIP Message With Large Contents...................7 
4.5 UAS Receiving a Request with Indirected Content...................8 
5.0 Proxy Behaviour...................................................8 
5.1 Congestion Safe Proxy.............................................8 
5.2 Proxy Receiving A SIP Message With Large Content..................8 
5.3 Proxy Refusing To Handle Large SIP Requests......................10 
5.4 Proxy Performing Content Indirection.............................10 
5.5 Proxy Receiving �413� Response From Downstream...................11 
5.6 Proxy Receiving a Request With Indirected Content................11 
6.0 Content Storage Server (CSS) Behaviour...........................12 
7.0 Syntax for Extensions............................................12 
7.1 Max-Size Header..................................................12 
7.2 Max-size parameter...............................................12 
8.0 Security considerations..........................................13 
9.0 Examples.........................................................13 
9.1 UAC Posting Contents to CSS Before Sending SIP MESSAGE...........13 
9.2 Receiving A MESSAGE With Large Contents after a REGISTER, Home 
Proxy Performs Content-Indirection...................................15 
9.3 Receiving A NOTIFY With Large Contents, UAS Refuses To Handle 
Request..............................................................17 
10.0 IANA Considerations.............................................19 
11.0 Acknowledgements................................................19 
12.0 References......................................................19 
Authors' Addresses...................................................19 
Full Copyright Statement.............................................20 
Acknowledgement......................................................20 
 
 
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1.0 Terminology 
    
   In this document, the key words "MUST", "MUST NOT", "REQUIRED", 
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED" "MAY", 
   and "OPTIONAL" are to be interpreted as described in RFC 2119 [5]. 
    
    
2.0 Introduction 
    
   The content-indirection service has gaind a lot of support from the 
   SIP community lately. There have been many scenarios where content-
   indirection service can be useful. This document discusses this 
   service and describes how it can be deployed in a network. This 
   document covers, in more detail, the scenarios where senders, 
   intermediaries and receivers of SIP messages may want to make use of 
   the content-indirection service. 
    
   A SIP UA originating a SIP request may not want to send SIP requests 
   into the network, but instead chooses to publish a link to a URI 
   where it has posted such content, using the mechanism described in 
   [3]. This restriction is maybe due to charging, network 
   configuration, or merely SIP message size constraints (the UA 
   limiting its SIP transport to message oriented protocol such as 
   UDP). Good examples of such SIP messages are MESSAGE [2] and PUBLISH 
   [6]. 
    
   The Session Initiation Protocol [1] allows the use of UDP for 
   transport of SIP messages.  The use of UDP with large payloads risks 
   network congestion problems, as UDP itself does not define 
   congestion prevention, detection, or correction mechanisms.  Large 
   SIP messages may cause UDP datagram fragmentation, something not 
   desired by a network of SIP entities.  
    
   The root of the problem lies with the SIP proxies. SIP proxies are 
   able to convert the transport protocol from reliable to unreliable 
   on hop-by-hop basis, therefore end-to-end congestion-safe path for 
   SIP messages cannot be guaranteed. 
    
   Conventional SIP payloads carry signalling information about media, 
   but not media itself.  There is potential that when a SIP 
   infrastructure is shared between call signalling and instant 
   messaging [2], the IM traffic will interfere with call signalling 
   traffic. This also applies to other types of SIP Requests that have 
   potential to carry large contents (NOTIFY is an example). 
    
   Furthermore, mobile terminals might want to limit the size of a SIP 
   message received regardless of the underlying transport protocol 
   (e.g. TCP). This may be due to memory limitations or the fact that 
   the radio link that the terminal is accessing at this instant is 
   very slow and the user wants to postpone collecting data until a 
   better link is acquired. 
    
    
 
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3.0 Overview of Operation 
    
   The basic scenario is when a SIP request is being sent from a UA1 to 
   a UA2 that has a home proxy. That SIP request could carry large 
   contents. 
 
    
    
    
                                    * 
                                    * 
              UA2 Domain            *    UA1 Domain 
                                    * 
                                    * 
                                    * 
                                    * 
                        +-------+   *   +-------+ 
                        | UA2   |   *   | UA1   | 
                        / CSS   |   *   | CSS   \ 
                      //|       |   *   |       |\\ 
                     /  +---+---+   *   +---+---+  \ 
                   //       |       *       |       \ 
                  /         |       *       |        \\ 
                 /          |       *       |          \ 
     +-------+ //       +---+---+   *   +---+---+       \  +-------+ 
     |       |/         | UA2   |   *   | UA1   |        \\|       | 
     | UA2   +----------+ Home  +---*---+ Home  +----------* UA1   | 
     |       |          | Proxy |   *   | Proxy |          |       | 
     +-------+          +-------+   *   +-------+          +-------+ 
                                    * 
                                    * 
                                    * 
                                    * 
                                    * 
                                    * 
                                    * 
    
   UA2 earlier has registered at the registrar co-located with the home 
   proxy. UA2 may have indicated its maximum acceptable message size 
   using the extensions defined in this document. 
    
   UA1 has a choice to either send requests with large contents 
   directly to its home proxy, or it can post the contents to its 
   Content Storage Server (CSS) and then send the link to the posted 
   content in the SIP request. Note that posting to the CSS does not 
   need the precondition of large contents of a SIP message. 
    
   In the first scenario, UA1 home proxy or UA2 home proxy receive the 
   request with the large contents. They have the option of rejecting 
   the request with a �513 Message Too Large� error response 
   (congestion safety). UA2 home Proxy also has the option of 
   forwarding the request to a Content Storage Server (CSS). That home 
 
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   proxy learns the maximum acceptable message size from registration 
   by UA2 (regardless of transport protocol) or through configuration. 
    
   UA2 home proxy, unaware of the limitations, can forward the large 
   message to UA2, who consequently, can reject the message with error 
   response �413 Request Entity Too Large�. UA2 can indicate its 
   maximum acceptable message size using extension in this document. 
    
   UA2 home proxy receiving the �413� response can either just forward 
   that response upstream or act as a B2BUA and resend the requested 
   with content-indirection to the CSS. 
    
   In the second scenario, UA1, using means outside the scope of this 
   document, posts the contents of the SIP request to the CSS and sends 
   the SIP request with a link to the contents as described in [3]. 
   Home proxies can then fetch the contents and amend the SIP request, 
   or it can be left up to UA2 to fetch the contents. The behaviour of 
   the home proxies receiving the SIP request is described in more 
   detail in the following chapters. 
    
    
4.0 UA Behaviour 
 
4.1 UAC Sending Requests 
 
   A UAC has a choice to either send requests with large contents 
   directly to a proxy, or it can post the contents to a Content 
   Storage Server (CSS) and then send the link to the posted content in 
   the SIP request. Note that posting to the CSS does not need the 
   precondition of large contents in a SIP message. 
    
   A UAC sending a SIP Request and wants to make sure that all proxies 
   on the path are congestion-safe proxies MUST insert a �Proxy-
   Require� header with the tag �congestion-safe� as described in [3]. 
    
    
4.2 UAC Behaviour when receiving Response 
    
   A SIP Response with large contents that exceed the limitations is 
   discarded. If the transport protocol is connection-oriented, the 
   connection MAY be closed to stop further sending of the response. 
    
4.2.1 �413� or �513� Error Responses 
    
   The max-size header MAY prompt the UAC of the request to send a 
   newly formulated request immediately with either smaller contents or 
   with content-indirection.  
    
   If the Retry-After header is also present, the UAC MUST either wait 
   for that time before sending the same request, or it immediately 
   sends the newly formulated request. It SHOULD NOT resend exactly the 
   same request within the retry-after period. The Retry-After value 
 
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   MAY be used as an indication of how long this Max-Size constraint is 
   valid for. 
    
   Max-size header indicates the SIP message size the UAS can handle 
   for the transport protocol used to send the request. This constraint 
   SHOULD NOT be applied to the same Request sent on different 
   transport protocol. 
    
   If an Accept-header is present without the value message/external-
   body, the UAC MUST NOT send the new message with content-type: 
   message/external-body. 
    
   The UAC MAY use this Max-size value to send new, unrelated Requests 
   to the same URI within the duration indicated by the retry-after 
   header. For example: UserA sends a SIP MESSAGE with large content to 
   userB. UserB rejects the request with �413 Request Entity Too Large� 
   with a max-size header of value 1300 bytes and a Retry-After value 
   of 80 seconds. UserA now needs to send an unrelated NOTIFY to userB, 
   due to an earlier SUBSCRIBE. This NOTIFY will be sent within the 80 
   seconds specified. UserA MAY use the 1300 byte constraint to send 
   the NOTIFY. This has to be done with care since the UAS receiving 
   the NOTIFY could be different that the one that received the 
   MESSAGE. The UAC has to be certain that UAS receiving the requests 
   are the same. 
    
    
4.3 UA Registering Desired Maximum SIP Message Size 
 
   Having learnt its capabilities and limitations (by user input or 
   other means), a SIP UA issuing a REGISTER request MAY indicate its 
   acceptable maximum size for a SIP message (including headers and 
   body).  
    
   This section introduces a new SIP-URI parameter �max-size�. 
    
   A SIP UA registering its address will include this SIP-URI parameter 
   in the contact-header supplied with the REGISTER request. 
    
   If the transport parameter is present in the SIP URI of the contact 
   header, then this max-size applies to that particular transport 
   protocol. If transport parameter is not present, then this max-size 
   applies to the default transport protocol used (UDP). 
    
   Example REGISTER Request: 
    
   REGISTER sip:registrar.nokia.com SIP/2.0 
   Via: SIP/2.0/UDP host1.nokia.com;branch=z9hG4bK346434r 
   From: sip:hisham.khartabil@nokia.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:hisham.khartabil@nokia.com 
   Contact: <sip:hisham.khartabil@host1.nokia.com;max-size=1300> 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 1 REGISTER 
   Max-Forwards:70 
 
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   Example REGISTER Request where the max-size applies to TCP: 
    
   REGISTER sip:registrar.nokia.com SIP/2.0 
   Via: SIP/2.0/UDP host1.Nokia.com;branch=z9hG4bK346434r 
   From: sip:hisham.khartabil@nokia.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:hisham.khartabil@nokia.com 
   Contact: <sip:hisham.khartabil@host1.nokia.com;transport=tcp;max-
   size=1300> 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 1 REGISTER 
   Max-Forwards:70 
    
    
4.4 UAS Receiving A SIP Message With Large Contents 
 
   A SIP UAS (terminal or application server like Presence Server) may 
   not have indicated its willingness or capabilities in receiving 
   large message content with the REGISTER request. This may be due to 
   lack of knowledge by the UA, at the time of registration, of its 
   memory or radio link constraints. 
    
   A SIP UAS receiving a SIP Request it is unable to handle due to 
   size, regardless of the underlying transport protocol, MAY reject 
   the request with �413 Request Entity Too Large�. The UAS MAY close 
   the connection, if the transport protocol was a reliable connection 
   oriented one, to prevent the sender from continuing the request. If 
   the condition is temporary, the UAS SHOULD include a Retry-After 
   header field to indicate that this condition is temporary and after 
   what time the sender can try to send the same request again. 
    
   The UAS MAY also indicate its acceptable message size capabilities. 
   Here we introduce a new SIP header �Max-Size�. The �413� response 
   sent back by the UAS MAY contain this header. This header contains 
   the maximum acceptable message size by this UAS. 
    
   The �413� response MAY also include an Accept-header with value 
   message/external-body indicating its capability to handle content-
   indirection. 
    
   Example response: 
    
   SIP/2.0 413 Request Entity Too Large 
   Via: SIP/2.0/UDP proxy.nokia.com;branch=z9hG4bK4kl5jr0iui0giu09df43 
   Via: SIP/2.0/UDP host1.somewhere.com;branch=z9hG4bK346434r 
   From: sip:markus.isomaki@nokia.com;tag=4jk123vxbc96 
   To: sip:hisham.khartabil@nokia.com;tag=fsad98754b6b64m 
   Call-Id: j4lk2j35879cvx7b 
   CSeq: 1 MESSAGE 
   Max-Size: 1300 
   Retry-After: 180 
   Max-Forwards:70 
   Accept: message/external-body 
 
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4.5 UAS Receiving a Request with Indirected Content 
    
   A UAS may receive a SIP request that had content-indirection 
   performed on it. In this case, the UAS MAY retrieve the contents 
   from the CSS using the URI supplied (See [3] for more details), 
   before it generates a response. It MAY retrieve the contents after 
   it generates the response. 
    
   If a request is an INVITE, the UAS MUST retrieve the contents before 
   generating the response. 
    
 
5.0 Proxy Behaviour 
 
5.1 Congestion Safe Proxy 
 
   A congestion safe proxy MUST be complaint with this specification as 
   well as [3] and [4]. This proxy MUST understand the option tag 
   �congestion-safe�. 
    
   A congestion-safe proxy MUST be able to handle content-indirection 
   if it is configured with a Content Storage Server (CSS) as described 
   in section 4.2. If not configured with CSS, the proxy behave as 
   described in [4] 
    
    
5.2 Proxy Receiving A SIP Message With Large Content 
 
   A proxy MAY be configured with a maximum allowable SIP message size 
   that is destined to a recipient on a certain interface. This value 
   is typically network configured with the lowest Maximum Transfer 
   Unit (MTU) en route to the recipient (UAS) on that interface. The 
   proxy server MUST only use the value when the recipient (UAS) is 
   using UDP as the transport protocol. How this configuration is 
   performed is out side the scope of this document. It is RECOMMENDED 
   that dynamic discovery and configuration is performed. A proxy 
   configured with this value is considered to be congestion-safe. 
    
   The proxy MAY also learn the maximum size of a particular UA by 
   means of registration as described in section 3.1 (Note in this 
   case, the value may possibly not be the MTU). This scenario is 
   possible when the registrar for a particular UA is co-located with 
   the proxy (this entity is typically referred to as the home proxy). 
    
   When both values are available to the home proxy (the one that 
   arrives in the contact-header of a registration request and the 
   locally configured value), and the transport protocol to be used is 
   connectionless oriented (UDP), the smaller of the two values is 
   used. If the registered URI uses connection-oriented protocol (such 
   as TCP), then the configured value MUST be ignored. 
    
 
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   When the proxy receives a request, it examines the message as 
   follows: 
    
   Note: This assumes that the proxy server has accepted to handle a 
   request with size larger than the configured MTU or larger than the 
   registered max-size for that recipient. If the proxy is not willing 
   to do so, it simply rejects the request with �513 Message Too 
   Large�. See section 4.3 for more details. 
    
   1. The proxy examines the SIP URI searching for host and transport 
   to send the request to according to procedures in [1]. 
    
   2. The proxy also searches for the max-size parameter (if a home 
   proxy) and for the configured maximum allowable message size for the 
   interface it will send the request on.  
    
   3. If the proxy is a home proxy and transport protocol is TCP, there 
   are 2 cases (since configured value is not used for TCP): 
    
     a. The proxy is configured with a max message size value and the 
        registered URI does not contain max-size parameter. In this 
        case, the message size is compared to the configured value. If 
        the message size is smaller, then processing proceeds as 
        normal. If the message size is greater, content-indirection MAY 
        be applied. See section 5.4 for more details. 
     b. The proxy is not configured with this value and the registered 
        URI does contain the max-size parameter. In this case, the 
        message size is compared to the registered URI max-size 
        parameter value. If the message size is smaller, then 
        processing proceeds as normal. If the message size is greater, 
        content-indirection MAY be applied. See section 5.4 for more 
        details. 
    
   4. If the proxy is a home proxy and the transport protocol is UDP, 
   there are 4 cases: 
    
     a. Proxy is not configured with this value and registered URI does 
        not contain max-size parameter. In this case, processing 
        proceeds as described in [3]. 
      
     b. Proxy is configured with this value and registered URI does not 
        contain max-size parameter. In this case, the message size is 
        compared to the configured value. If the message size is 
        smaller, then processing proceeds as normal. If the message 
        size is greater, content-indirection MAY be applied. See 
        section 5.4 for more details. 
     c. Proxy is not configured with this value and registered URI does 
        contain max-size parameter. In this case, the message size is 
        compared to the registered URI max-size parameter value. If the 
        message size is smaller, then processing proceeds as normal. If 
        the message size is greater, content-indirection MAY be 
        applied. See section 5.4 for more details. 
 
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     d. Proxy is configured with this value and registered URI does 
        contain max-size parameter. In this case, the message size is 
        compared to the smaller value of the two. If the message size 
        is smaller, then processing proceeds as normal. If the message 
        size is greater, content-indirection MAY be applied. See 
        section 5.4 for more details. 
           
   5. If the proxy is not responsible for that domain receives the 
   request and the transport protocol is a connection-oriented one, the 
   request is simply forwarded 
    
   6. If the proxy is not responsible for that domain receives the 
   request and the transport protocol is a connectionless one, there 
   are 2 cases: 
    
     a. Proxy is not configured with this value. In this case, 
       processing proceeds as described in [4]. 
     b. Proxy is configured with this value. In this case, the message 
       size is compared to the configured value. If the message size is 
       smaller, then processing proceeds as normal. If the message size 
       is greater, content-indirection MAY be applied. See section 5.4 
       for more details. 
    
   A SIP Response with large contents that exceeds the maximum 
   acceptable size is discarded. If the transport protocol is 
   connection-oriented, the connection MAY be closed to stop further 
   sending of the response. 
    
    
5.3 Proxy Refusing To Handle Large SIP Requests 
 
   There are circumstances where the proxy refuses to perform content-
   indirection on a received SIP Request destined to a recipient where 
   a maximum message size constraint has been imposed. These 
   circumstances may include the user not paying for this service or 
   simply that the proxy does not know how to perform content-
   indirection (see section 4.1 about a congestion safe proxy). The 
   definitions of these circumstances are outside the scope of this 
   document. 
    
   In any case, when the proxy refuses to handle this situation or it 
   is simply not a congestion safe proxy (does not understand the 
   option tag �congestion-safe�), it returns a �513 Message Too Large� 
   response. Reference [4] shows how a proxy SHOULD handle this 
   situation. 
    
   The �513� response MAY include an Accept-header with 
   message/external-body. 
 
    
5.4 Proxy Performing Content Indirection 
 
 
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   A proxy that is a congestion safe proxy MUST handle content 
   indirection. 
    
   Here we define a new functional SIP entity called �Content Storage 
   Server (CSS). A congestion safe proxy MUST be configured with the 
   CSS address. 
    
   Once the home proxy has accepted to perform content indirection on a 
   SIP Request, it forwards the message to the CSS. Section 16.6 of [1] 
   defines the steps to follow. Pay special attention to step 6 where 
   it discusses how a proxy may divert a SIP Request to a specific next 
   hop. Below is a rewrite of that section tailored to the terminology 
   used in this document: 
    
   A proxy MAY mandate that a request visit a CSS before being 
   delivered to the destination. This is to accommodate content 
   indirection. A proxy MUST ensure that CSS is a loose router. 
   Generally, this can only be known with certainty if the CSS is 
   within the same administrative domain. The CSS is represented by a 
   URI (which contains the lr parameter). This URI MUST be pushed into 
   the Route header field ahead of any existing values, if present. If 
   the Route header field is absent, it MUST be added, containing that 
   URI. 
    
    
5.5 Proxy Receiving �413� Response From Downstream 
    
   Open issue: should we just choose to forward the response upstream? 
    
   A UA or proxy refusing to handle a large SIP Request may reject the 
   request with �413� error response as describe in section 3.2. 
    
   When a proxy receives this response, it MAY choose to handle the 
   request itself. It does so by reissuing the request, but this time, 
   it sends it via the CSS using the route-header, as described in 
   section 5.4. The proxy MAY also send a �181 Call Is Being Forwarded� 
   provisional response back to the sender. 
    
   There are situations where the proxy refuses the handle the request 
   itself and alternatively just forward the �413� response to sender. 
   See section 5.3 for more details. 
    
    
5.6 Proxy Receiving a Request With Indirected Content 
    
   A proxy may receive a SIP request that had content-indirection 
   performed on it. In this case, the proxy MAY perform one of 2 
   things: 
    
   a. Retrieve the contents and amend the request to contain those 
      contents. It is RECOMMENDED for a proxy to do so if the CSS in 
      within the same administrative domain as the proxy. It is also 
      RECOMMENDED for a proxy to do so if the CSS is not within the 
 
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      administrative domain as the proxy, but there is a trust 
      relationship between the two domains. The proxy sends the new 
      request to the destination, maintaining any tags in the To and 
      From headers sent in the original request. This is to accommodate 
      for SIP requests within dialogs being delivered to the right 
      dialog. 
 
   b. Forward the request without retrieving the contents. It is NOT 
      RECOMMENDED for a to proxy forwards the request without first 
      retrieving the content and amending the request to contain it, if 
      the CSS is within the same administrative domain as the proxy. 
    
    
6.0 Content Storage Server (CSS) Behaviour 
    
   The CSS behaves as a SIP Back To Back User Agent (B2BUA). When it 
   receives a SIP Request performs the following steps: 
    
   a. Extracts the body of the Request and stores it in an HTTP server. 
 
   b. Responds to the sender with a �202 Accepted� response. INVITE is 
      a special case. The CSS MUST NOT respond with �202� in this case, 
      but instead wait for a response from the UAS. 
 
   c. Formulates a new request that includes the URL where the content 
      can be retrieved. This procedure is described in [3]. 
 
   d. Sends the new request to the destination, maintaining any tags in 
      the To and From headers sent in the original request. This is to 
      accommodate for SIP requests within dialogs being delivered to 
      the right dialog. 
 
   OPEN ISSUE: Should the behaviour of CSS be uniform across all 
   requests? I.e. Not generate a �202� response at all. 
    
    
7.0 Syntax for Extensions 
 
7.1 Max-Size Header 
    
   Max-Size = �Max-Size� HCOLON delta-seconds 
    
   Additions to SIP Table 3: 
    
   Header field          where   proxy ACK BYE CAN INV OPT REG 
   ----------------------------------------------------------- 
    
   Max-Size               413      a    -   -   -   -   -   - 
    
    
7.2 Max-size parameter 
 
   Max-size-param = �max-size=� 1*DIGIT 
 
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8.0 Security considerations 
    
   The presence of Max-size header in a SIP message has a significant 
   effect on the ways in which the request is handled at a server. As a 
   result, it is especially important that messages containing this 
   extension be authenticated. The same holds true for registrations 
   with contact parameters. 
    
   Processing of requests and looking up criteria for message size 
   requires set operations and searches, which can require some amount 
   of computation. This enables a DoS attack whereby a user can send 
   requests with substantial numbers messages with large contents, in 
   the hopes of overloading the server. To counter this, the server 
   SHOULD authenticate the sender. 
    
   REGISTER requests can reveal sensitive information about a UA�s 
   capabilities. If this information is sensitive, it SHOULD be 
   encrypted using SIP S/MIME capabilities. 
    
    
9.0 Examples 
 
9.1 UAC Posting Contents to CSS Before Sending SIP MESSAGE 
    
    
   UA2             UA2             UA1             UA1             UA1 
                   Home            Home            CSS 
                   Proxy           Proxy 
    |               |               |               |               | 
    |               |               |               |               | 
    |               |               |               | (F1) post     | 
    |               |               |               |<------------->| 
    |               |               |               |               | 
    |               |               |               |               | 
    |               |               |               | (F2) MESSAGE  | 
    |               |               |<------------------------------| 
    |               |               |               |               | 
    |               |               |               |               | 
    |               |               | (F3) get      |               | 
    |               |               |<------------->|               | 
    |               | (F4) MESSAGE  |               |               | 
    |               |<--------------|               |               | 
    | (F5) MESSAGE  |               |               |               | 
    |<--------------|               |               |               | 
    |               |               |               |               | 
    |               |               |               |               | 
    |(F6) 200 Ok    |               |               |               | 
    |-------------->| (F7) 2OO Ok   |               |               | 
    |               |-------------->| (F8) 200 Ok   |               | 
    |               |               |------------------------------>| 
    |               |               |               |               | 
    |               |               |               |               | 
 
Khartabil                                                    [Page 13] 

Internet Draft        Congestion Safety and C-I          October 2002 
 
 
    |               |               |               |               | 
    |               |               |               |               | 
    |               |               |               |               | 
    |               |               |               |               | 
    
   (F1) UA1 posts, using some means outside this document, the contents 
   to the CSS 
    
   (F2) UA1 sends the SIP MESSAGE with content-type: message/external-
   body 
    
   MESSAGE sip:ua2@nokia.com SIP/2.0 
   Via: SIP/2.0/UDP host1.somewhere.com;branch=z9hG4bK346434r 
   From: sip:ua1@somewhere.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:ua1@nokia.com 
   Contact: <sip:ua1@host1.nokia.com;max-size=1300> 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 1 MESSAGE 
   Max-Forwards: 70 
   Content-Type: message/external-body; access-type=URL; 
           
   url="http://131.228.13.2/im/9207807c03d4a5e0e6907b0dc89c9067"; 
   Content-Length: 32 
    
   Content-Type: text/html 
    
    
   (F3) UA1 home proxy fetches the contents and amends the SIP MESSAGE 
   with the new contents 
    
   (F4) Newly constructed MESSAGE send to the domain of the recipient 
    
   MESSAGE sip:ua2@nokia.com SIP/2.0 
   Via: SIP/2.0/TCP au1-homeproxy.nokia.com;branch=z9hG4bKewrlkjdfkjl 
   Via: SIP/2.0/UDP host1.somewhere.com;branch=z9hG4bK346434r 
   From: sip:ua1@somewhere.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:ua1@nokia.com 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 1 MESSAGE 
   Max-Forwards: 69 
   Content-type: text/html 
   Content-length: 20000 
    
   [Large Body] 
    
    
   (F5) UA2 home proxy decides that the content of this message is not 
   too large and therefore does not perform content redirection of the 
   message 
    
   (F6) � (F8) �200 Ok� for the request. 
 
 
 
Khartabil                                                    [Page 14] 

Internet Draft        Congestion Safety and C-I          October 2002 
 
 
9.2 Receiving A MESSAGE With Large Contents after a REGISTER, Home 
Proxy Performs Content-Indirection 
    
    UA2                 CSS                UA2                  UA1 
                                           Home 
                                           Proxy 
     |                   |                   |                   | 
     |                   |                   |                   | 
     |  (F1) REGISTER    |                   |                   | 
     |-------------------------------------->|                   | 
     |                   |                   |                   | 
     |                   | (F2) 200 Ok       |                   | 
     |<--------------------------------------|                   | 
     |                   |                   |                   | 
     |                   |                   |                   | 
     |                   |                   |                   | 
     |                   |                   | (F3) MESSAGE      | 
     |                   |                   |<------------------| 
     |                   | (F4) MESSAGE      |                   | 
     |                   |<------------------|                   | 
     |                   |                   |                   | 
     |                   |                   |                   | 
     |                   | (F5) 202 Accepted |                   | 
     |                   |------------------>|                   | 
     | (F7) MESSAGE      |                   | (F6) 202 Accepted | 
     |<------------------|                   |------------------>| 
     |                   |                   |                   | 
     |                   |                   |                   | 
     | (F8) 200 Ok       |                   |                   | 
     |------------------>|                   |                   | 
     |                   |                   |                   | 
     |                   |                   |                   | 
     |                   |                   |                   | 
    
    
   (F1) REGISTER sent from UA2 to Home Proxy (Registrar) 
    
   REGISTER sip:registrar.nokia.com SIP/2.0 
   Via: SIP/2.0/UDP host2.somewhere.com;branch=z9hG4bKue73jhhdue 
   From: sip:ua2@somewhere.com;tag=48er8fdjkcfnciue9843ifj 
   To: sip:ua2@somewhere.com 
   Call-Id: 3mkejdc9e9834judjd 
   CSeq: 1 REGISTER 
   Expires: 3600 
   Contact: <sip:ua2@host2.nokia.com;max-size=1300> 
   Max-Forwards: 70 
    
   (F2) 200 Ok from Home proxy to UA2 
    
   SIP/2.0 200 Ok 
   Via: SIP/2.0/UDP host2.somewhere.com;branch=z9hG4bKue73jhhdue 
   From: sip:ua2@somewhere.com;tag=48er8fdjkcfnciue9843ifj 
   To: sip:ua2@somewhere.com;tag=393k3lmn3n3934 
 
Khartabil                                                    [Page 15] 

Internet Draft        Congestion Safety and C-I          October 2002 
 
 
   Call-Id: 3mkejdc9e9834judjd 
   CSeq: 1 REGISTER 
   Contact: <sip:ua1@host2.nokia.com;max-size=1300>,expires=3600 
   Max-Forwards: 70 
    
   (F3) Message sent from UA1 to UA2 with very large content 
    
   MESSAGE sip:ua2@nokia.com SIP/2.0 
   Via: SIP/2.0/UDP host1.somewhere.com;branch=z9hG4bK346434r 
   From: sip:ua1@somewhere.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:ua1@nokia.com 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 1 MESSAGE 
   Max-Forwards: 70 
   Content-type: text/html 
   Content-length: 20000 
    
   [Large Body] 
    
   (F4) MESSAGE is directed by UA2's home proxy to CSS. Home proxy 
   accepts to content-indirect 
    
   MESSAGE sip:ua2@nokia.com SIP/2.0 
   Via: SIP/2.0/TCP homeproxy.nokia.com;branch=z9hG4bKewrlkjdfkjl 
   Via: SIP/2.0/UDP host1.somewhere.com;branch=z9hG4bK346434r 
   From: sip:ua1@somewhere.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:ua1@nokia.com 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 1 MESSAGE 
   Route: sip:css.nokia.com 
   Max-Forwards: 69 
   Content-type: text/html 
   Content-length: 20000 
    
   [Large Body] 
    
   (F5) CCS returns a 202 Accepted 
    
   SIP/2.0 202 Accepted 
   Via: SIP/2.0/TCP homeproxy.nokia.com;branch=z9hG4bKewrlkjdfkjl 
   Via: SIP/2.0/UDP host1.somewhere.com;branch=z9hG4bK346434r 
   From: sip:ua1@somewhere.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:ua1@nokia.com;tag=5nixucvzo3241bqewmn 
   Contact: <sip:ua1@host1.nokia.com;max-size=1300> 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 1 MESSAGE 
   Max-Forwards: 70 
    
   (F6) Home proxy forwards response to UA1 
    
   SIP/2.0 202 Accepted 
   Via: SIP/2.0/UDP host1.somewhere.com;branch=z9hG4bK346434r 
   From: sip:ua1@somewhere.com;tag=31jrlkejrq3kl3jkl879defje3 
 
Khartabil                                                    [Page 16] 

Internet Draft        Congestion Safety and C-I          October 2002 
 
 
   To: sip:ua1@nokia.com;tag=5nixucvzo3241bqewmn 
   Contact: <sip:ua1@host1.nokia.com;max-size=1300> 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 1 MESSAGE 
   Max-Forwards: 69 
    
   (F7) CCS sends the content-indirected MESSAGE to UA2 with the URL 
    
   MESSAGE sip:ua2@nokia.com SIP/2.0 
   Via: SIP/2.0/UDP css.nokia.com;branch=z9hG4bK342kj5klj43klndsm 
   From: sip:ua1@somewhere.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:ua1@nokia.com 
   Contact: <sip:ua1@host1.nokia.com;max-size=1300> 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 1 MESSAGE 
   Max-Forwards: 70 
   Content-Type: message/external-body; access-type=URL; 
           
   url="http://131.228.13.2/im/9207807c03d4a5e0e6907b0dc89c9067"; 
   Content-Length: 32 
    
   Content-Type: text/html 
    
   (F8) 200 Ok is returned by UA2 
    
   SIP/2.0 200 Ok 
   Via: SIP/2.0/UDP css.nokia.com;branch=z9hG4bK342kj5klj43klndsm 
   From: sip:ua1@somewhere.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:ua1@nokia.com;tag=dfjkla43kjl5ldskfj 
   Contact: <sip:ua1@host1.nokia.com;max-size=1300> 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 1 MESSAGE 
   Max-Forwards: 70 
    
    
9.3 Receiving A NOTIFY With Large Contents, UAS Refuses To Handle 
Request 
 
   This example assumes that a subscription has been established. It 
   also assumes that UA2 has not registered a max-size limit and that 
   the home proxy configured max-size is less than the size of the 
   NOTIFY. 
    
   UA2                 CSS                 UA2                  PS 
                                           Home 
                                           Proxy 
     |                   |                   |                   | 
     |                   |                   | (F1) NOTIFY       | 
     |                   |                   |<------------------| 
     |                   |                   |                   | 
     |  (F2) NOTIFY      |                   |                   | 
     |<--------------------------------------|                   | 
     |                   |                   |                   | 
 
Khartabil                                                    [Page 17] 

Internet Draft        Congestion Safety and C-I          October 2002 
 
 
     |                   | (F3) 413          |                   | 
     |-------------------------------------->|                   | 
     |                   |                   | (F4) 413          | 
     |                   |                   |------------------>| 
     |                   |                   |                   | 
    
   (F1) NOTIFY sent from PS to UA2, via home proxy with very large 
   content 
    
   NOTIFY sip:ua2@host2.nokia.com SIP/2.0 
   Via: SIP/2.0/TCP ps.nokia.com;branch=z9hG4bK346434r 
   From: sip:ps.nokia.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:ua2@nokia.com;tag=eqwriuo423nf 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 2 NOTIFY 
   Route: sip: homeproxy.nokia.com 
   Max-Forwards: 70 
   Content-Type: application/cpim-pidf+xml 
   Content-length: 20000 
    
   [Large Body] 
    
   (F2) NOTIFY forwarded from home proxy to UA2 with very large 
   content. Home proxy checked its max-size configuration and found 
   that this message passes.  
    
   NOTIFY sip:ua2@host2.nokia.com SIP/2.0 
   Via: SIP/2.0/TCP homeproxy.nokia.com;branch=z9hG4bKre78re89jfdkj 
   Via: SIP/2.0/TCP ps.nokia.com;branch=z9hG4bK346434r 
   From: sip:ps.nokia.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:ua2@nokia.com;tag=eqwriuo423nf 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 2 NOTIFY 
   Max-Forwards: 69 
   Content-Type: application/cpim-pidf+xml 
   Content-length: 20000 
    
   [Large Body] 
    
   (F3) UA2 refuses the request due to its large content and returns a 
   413 response 
    
   SIP/2.0 413 Message Too Large 
   Via: SIP/2.0/TCP homeproxy.nokia.com;branch=z9hG4bKre78re89jfdkj 
   Via: SIP/2.0/TCP ps.nokia.com;branch=z9hG4bK346434r 
   From: sip:ps.nokia.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:ua2@nokia.com;tag=eqwriuo423nf 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 2 NOTIFY 
   Max-size: 1300 
   Max-Forwards: 70 
    
 
Khartabil                                                    [Page 18] 

Internet Draft        Congestion Safety and C-I          October 2002 
 
 
   (F4) 413 passed by home proxy to PS. Home proxy did not content-
   indirect. 
   SIP/2.0 413 Message Too Large 
   Via: SIP/2.0/TCP ps.nokia.com;branch=z9hG4bK346434r 
   From: sip:ps.nokia.com;tag=31jrlkejrq3kl3jkl879defje3 
   To: sip:ua2@nokia.com;tag=eqwriuo423nf 
   Call-Id: vjn86732nv4fgiofd 
   CSeq: 2 NOTIFY 
   Max-size: 1300 
   Max-Forwards: 69 
    
    
10.0 IANA Considerations 
    
   Once the header and parameter names have been agreed on, they will 
   be registered with IANA. 
    
11.0 Acknowledgements 
    
   I would like to thank Jose Costa-Requena, Mikko Lonnfors, Pekka 
   Pessi and Nokia for their comments and support. 
    
    
12.0 References 
 
   [1] J. Rosenberg, et al. �SIP: Session Initiation Protocol�. RCF 
   3261, Internet Engineering Task Force, June 2002. 
    
   [2] B. Campbell et al. "SIP Extensions for Instant Messaging", 
   RFC3428, Internet Engineering Task Force, December 2002. 
     
   [3] S. Olson. "A Mechanism for Content Indirection is SIP Messages", 
   draft-ietf-sip-content-indirect-mech-01.txt. Internet Draft, 
   Internet Engineering Task Force, November 2002. Work in progress. 
    
   [4] D. Willis. "SIP Extension to Assure Congestion Safety", draft-
   ietf-sip-congestsafe-00.txt. Internet Draft, Internet Engineering 
   Task Force, August 2002. Work in progress. 
    
   [5] S. Bradner, "Key words for use in RFCs to indicate requirement 
      levels," RFC 2119, Internet Engineering Task Force, Mar. 1997. 
 
   [3] S. Olson. Et al. "SIMPLE Presence Publication Mechanism", draft-
   olson-simple-publish-01. Internet Draft, Internet Engineering Task 
   Force, October 2002. Work in progress. 
    
    
Authors' Addresses 
    
   Hisham Khartabil 
   Nokia 
       
   P.O. Box 321 
 
Khartabil                                                    [Page 19] 

Internet Draft        Congestion Safety and C-I          October 2002 
 
 
   FIN-00045 
   NOKIA GROUP 
   FINLAND 
    
   Email: Hisham.Khartabil@nokia.com 
   Tel: + 358 7180 76161 
    
Full Copyright Statement 
    
   Copyright (C) The Internet Society (2003).  All Rights Reserved. 
    
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   or assist in its implementation may be prepared, copied, published 
   and distributed, in whole or in part, without restriction of any 
   kind, provided that the above copyright notice and this paragraph 
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   The limited permissions granted above are perpetual and will not be 
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Acknowledgement 
    
   Funding for the RFC Editor function is currently provided by the 
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Khartabil                                                    [Page 20] 

--------------4CF01AB8B5C22B7DB1C66030--

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From: Sumit Garg <sgarg@aastra.com>
To: sip@ietf.org
Date: Tue, 11 Mar 2003 11:19:15 -0500
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Subject: [Sip] RFC 3398: REL cause
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According to the rfc 3398(SIP-ISUP mapping, section 7.1.3 and 7.2.2),for a
net to Phone call, when ACM timeout timer
 (T7- 20 - 30 s) expires, the ISUP leg  REL cause should be 102 (protocol
error, recovery on timer expiry) and corresponding SIP response should be
504 Server Timeout.
However, if one goes by Q.764 and table 1 of Q.850 the correct REL cause for
ISUP would be 31(normal unspecified)
and for the SIP leg it would then correspond to  480 (Temporarily
Unavailable). Apparently 102 is used mostly for
SUS-RES.

What should be the values used?


~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 
Sumit Garg 
Aastra Telecom US
8 Federal Street 
Billerica, MA 01821 
978-436-4244
email: <<mailto:sgarg@aastra.com>> 
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 


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Ok, here's the current working draft agenda for SIP WG. Please feedback asap
if you have issues/suggestions.


Posted at:

http://www.softarmor.com/sipwg/meets/ietf56/agenda.html


Monday, March 17, 1300-1500,  Continental 6


1300 Agenda Bash
1305 Status -- Chairs
1315 Non-Invite Transaction Issues, Robert Sparks
    draft-sparks-sip-noninvite-00.txt
1330 Referred-By Changes and Open Issues, Robert Sparks
     draft-ietf-sip-referredby-01.txt
1340 Resource Priority, Henning Schulzrinne
     draft-polk-sip-resource-02.txt
1355 Caller Preferences, Jonathan Rosenberg
     draft-ietf-sip-callerprefs-08.txt
1410 Congestion Safety, Dean Willis, Hisham Khartabil
     draft-ietf-sip-congestsafe-01.txt
      draft-khartabil-sip-congestionsafe-ci-02.txt
1430 General Discussion
1500 Break

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Subject: RE: [Sip] Draft agenda, SIP Working Group
Date: Wed, 12 Mar 2003 03:55:51 +0200
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Thread-Topic: [Sip] Draft agenda, SIP Working Group
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
Cc: "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>,
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Is there any intention to include the SIP MIB on the agenda? 

Thanks,

Dan


> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Wednesday, March 12, 2003 3:16 AM
> To: sip@ietf.org
> Cc: Gonzalo Camarillo; Jon Peterson 
> (jon.peterson@NeuStar.com); Rohan Mahy
> Subject: [Sip] Draft agenda, SIP Working Group
> 
> 
> 
> Ok, here's the current working draft agenda for SIP WG. 
> Please feedback asap
> if you have issues/suggestions.
> 
> 
> Posted at:
> 
> http://www.softarmor.com/sipwg/meets/ietf56/agenda.html
> 
> 
> Monday, March 17, 1300-1500,  Continental 6
> 
> 
> 1300 Agenda Bash
> 1305 Status -- Chairs
> 1315 Non-Invite Transaction Issues, Robert Sparks
>     draft-sparks-sip-noninvite-00.txt
> 1330 Referred-By Changes and Open Issues, Robert Sparks
>      draft-ietf-sip-referredby-01.txt
> 1340 Resource Priority, Henning Schulzrinne
>      draft-polk-sip-resource-02.txt
> 1355 Caller Preferences, Jonathan Rosenberg
>      draft-ietf-sip-callerprefs-08.txt
> 1410 Congestion Safety, Dean Willis, Hisham Khartabil
>      draft-ietf-sip-congestsafe-01.txt
>       draft-khartabil-sip-congestionsafe-ci-02.txt
> 1430 General Discussion
> 1500 Break
> 
> _______________________________________________
> 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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In RFC3265, it is mentioned that NOTIFY can certainly establish a dialog if it arrives before the 200 of a SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now arrives with record-route headers? Does the route-set get updated? (keeping in mind that the dialog is now in the established state).

Regards,
Hisham
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Romascanu, Dan \(Dan\)'" <dromasca@avaya.com>, <sip@ietf.org>
Cc: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>,
        "'Jon Peterson \(jon.peterson@NeuStar.com\)'" <jon.peterson@neustar.biz>,
        "'Rohan Mahy'" <rohan@cisco.com>
Subject: RE: [Sip] Draft agenda, SIP Working Group
Date: Wed, 12 Mar 2003 10:51:22 -0600
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Well, I don't recall that any of the MIB authors indicated that there was
any need for face time to discuss the MIB (of course, they might hve, and I
might have been asleep). That, of course, may or may not be what the rest of
the working group thinks.

Do we need to discuss the MIB? Any controversial and unresolved issues that
need to be talked through in the full group?

It would be "nice" to move the MIB along, now that 3261 and the other major
elements associated with it have basically stabilized.

--
Dean


> -----Original Message-----
> From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com] 
> Sent: Tuesday, March 11, 2003 7:56 PM
> To: Dean Willis; sip@ietf.org
> Cc: Gonzalo Camarillo; Jon Peterson 
> (jon.peterson@NeuStar.com); Rohan Mahy
> Subject: RE: [Sip] Draft agenda, SIP Working Group
> 
> 
> Is there any intention to include the SIP MIB on the agenda? 
> 
> Thanks,
> 
> Dan
> 

_______________________________________________
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: "Mark Watson" <mwatson@nortelnetworks.com>
To: sip@ietf.org, "'jon.peterson@neustar.biz'" <jon.peterson@neustar.biz>
Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-identity-01.txt
Date: Wed, 12 Mar 2003 17:41:12 -0000
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Jon,

Not sure if this was in the previous version, but I have a comment on the
following from the introduction:

  "Coincidentally, the address-of-record URI of a SIP user is also the
   URI with which a SIP UA populates the From header of requests from
   that user - in other words, the address-of-record is an identity.  So
   in this context users already have a means of providing their
   identity, which makes good sense: since the contents of a From header
   field are essentially a 'return address' for SIP requests, being able
   to prove that you are eligible to receive requests for that 'return
   address' should be identical to proving that you are authorized to
   assert this identity."

I agree that prove of eligability to receive requests for an address is a
sufficient condition for being authorised to assert an identity, but I do
not think it is necessary.

I can imagine scenarios in which I am authorised to assert an address as my
identity, but I am not eligable to receive requests for that address. Just
because I have an outgoing call service does not mean I have an incoming
calls service.

So, I suggest replacing "identical to proving" with "sufficient to prove" in
the above paragraph.

Regards...Mark

> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: 07 March 2003 20:39
> Cc: sip@ietf.org
> Subject: [Sip] I-D ACTION:draft-ietf-sip-identity-01.txt
> 
> 
> 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
> 	Filename	: draft-ietf-sip-identity-01.txt
> 	Pages		: 16
> 	Date		: 2003-3-7
> 	
> The existing mechanisms for expressing identity in the Session
> Initiation Protocol oftentimes do not permit an administrative domain
> to verify securely the identity of the originator of a request.  This
> document recommends practices and conventions for authenticating end
> users, and proposes a way to distribute cryptographically secure
> authenticated identities within SIP messages.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-identity-01.txt
> 
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<P><FONT SIZE=3D2>Jon,</FONT>
</P>

<P><FONT SIZE=3D2>Not sure if this was in the previous version, but I =
have a comment on the following from the introduction:</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp; &quot;Coincidentally, the address-of-record =
URI of a SIP user is also the</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; URI with which a SIP UA populates the =
From header of requests from</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; that user - in other words, the =
address-of-record is an identity.&nbsp; So</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; in this context users already have a =
means of providing their</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; identity, which makes good sense: since =
the contents of a From header</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; field are essentially a 'return =
address' for SIP requests, being able</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; to prove that you are eligible to =
receive requests for that 'return</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; address' should be identical to proving =
that you are authorized to</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; assert this identity.&quot;</FONT>
</P>

<P><FONT SIZE=3D2>I agree that prove of eligability to receive requests =
for an address is a sufficient condition for being authorised to assert =
an identity, but I do not think it is necessary.</FONT></P>

<P><FONT SIZE=3D2>I can imagine scenarios in which I am authorised to =
assert an address as my identity, but I am not eligable to receive =
requests for that address. Just because I have an outgoing call service =
does not mean I have an incoming calls service.</FONT></P>

<P><FONT SIZE=3D2>So, I suggest replacing &quot;identical to =
proving&quot; with &quot;sufficient to prove&quot; in the above =
paragraph.</FONT>
</P>

<P><FONT SIZE=3D2>Regards...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Internet-Drafts@ietf.org [<A =
HREF=3D"mailto:Internet-Drafts@ietf.org">mailto:Internet-Drafts@ietf.org=
</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 07 March 2003 20:39</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: [Sip] I-D =
ACTION:draft-ietf-sip-identity-01.txt</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; A New Internet-Draft is available from the =
on-line </FONT>
<BR><FONT SIZE=3D2>&gt; Internet-Drafts directories.</FONT>
<BR><FONT SIZE=3D2>&gt; This draft is a work item of the Session =
Initiation Protocol </FONT>
<BR><FONT SIZE=3D2>&gt; Working Group of the IETF.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Title&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
Enhancements for Authenticated </FONT>
<BR><FONT SIZE=3D2>&gt; Identity Management </FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp; in the Session =
Initiation Protocol (SIP)</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : J. Peterson</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-ietf-sip-identity-01.txt</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Pages&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
16</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Date&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
2003-3-7</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; The existing mechanisms for expressing identity =
in the Session</FONT>
<BR><FONT SIZE=3D2>&gt; Initiation Protocol oftentimes do not permit an =
administrative domain</FONT>
<BR><FONT SIZE=3D2>&gt; to verify securely the identity of the =
originator of a request.&nbsp; This</FONT>
<BR><FONT SIZE=3D2>&gt; document recommends practices and conventions =
for authenticating end</FONT>
<BR><FONT SIZE=3D2>&gt; users, and proposes a way to distribute =
cryptographically secure</FONT>
<BR><FONT SIZE=3D2>&gt; authenticated identities within SIP =
messages.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; A URL for this Internet-Draft is:</FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-identity-01.t=
xt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-sip-ide=
ntity-01.txt</A></FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; To remove yourself from the IETF Announcement =
list, send a message to </FONT>
<BR><FONT SIZE=3D2>&gt; ietf-announce-request with the word unsubscribe =
in the body </FONT>
<BR><FONT SIZE=3D2>&gt; of the message.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Internet-Drafts are also available by anonymous =
FTP. Login </FONT>
<BR><FONT SIZE=3D2>&gt; with the username</FONT>
<BR><FONT SIZE=3D2>&gt; &quot;anonymous&quot; and a password of your =
e-mail address. After logging in,</FONT>
<BR><FONT SIZE=3D2>&gt; type &quot;cd internet-drafts&quot; and =
then</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;get =
draft-ietf-sip-identity-01.txt&quot;.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; A list of Internet-Drafts directories can be =
found in</FONT>
<BR><FONT SIZE=3D2>&gt; <A HREF=3D"http://www.ietf.org/shadow.html" =
TARGET=3D"_blank">http://www.ietf.org/shadow.html</A> </FONT>
<BR><FONT SIZE=3D2>&gt; or <A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" =
TARGET=3D"_blank">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</A></FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Internet-Drafts can also be obtained by =
e-mail.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Send a message to:</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
mailserv@ietf.org.</FONT>
<BR><FONT SIZE=3D2>&gt; In the body type:</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;FILE =
/internet-drafts/draft-ietf-sip-identity-01.txt&quot;.</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; NOTE: The mail server at ietf.org can return =
the document in</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MIME-encoded =
form by using the &quot;mpack&quot; utility.&nbsp; To use this</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; feature, insert =
the command &quot;ENCODING mime&quot; before the =
&quot;FILE&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; command.&nbsp; =
To decode the response(s), you will need &quot;munpack&quot; or</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a MIME-compliant =
mail reader.&nbsp; Different MIME-compliant </FONT>
<BR><FONT SIZE=3D2>&gt; mail readers</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit =
different behavior, especially when dealing with</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&quot;multipart&quot; MIME messages (i.e. documents which have been =
split</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; up into multiple =
messages), so check your local documentation on</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; how to =
manipulate these messages.</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; Below is the data which will enable a MIME =
compliant mail reader</FONT>
<BR><FONT SIZE=3D2>&gt; implementation to automatically retrieve the =
ASCII version of the</FONT>
<BR><FONT SIZE=3D2>&gt; Internet-Draft.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

</BODY>
</HTML>
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From mailnull@www1.ietf.org  Wed Mar 12 13:15:15 2003
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Date: Wed, 12 Mar 2003 13:11:46 -0500
From: Kevin Lingle <klingle@cisco.com>
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To: Dean Willis <dean.willis@softarmor.com>
CC: "'Romascanu, Dan (Dan)'" <dromasca@avaya.com>, sip@ietf.org,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>,
        "'Jon Peterson (jon.peterson@NeuStar.com)'" <jon.peterson@neustar.biz>,
        "'Rohan Mahy'" <rohan@cisco.com>
Subject: Re: [Sip] Draft agenda, SIP Working Group
References: <001a01c2e8b7$9d37e360$ee036e3f@txdwillis>
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Dean Willis wrote:
> 
> Well, I don't recall that any of the MIB authors indicated that there was
> any need for face time to discuss the MIB (of course, they might hve, and I
> might have been asleep). That, of course, may or may not be what the rest of
> the working group thinks.

we (mib authors) did not ask for time on the agenda.

> 
> Do we need to discuss the MIB? Any controversial and unresolved issues that
> need to be talked through in the full group?
> 
> It would be "nice" to move the MIB along, now that 3261 and the other major
> elements associated with it have basically stabilized.

agreed.   we recently published -05 revision and expect to 
push out -06 because there were a few loose ends.  
after passing wg lc last year w/ rev -04,  we determined it
prudent to make the mib modules relevant to rfc3261.
our ability to get around do doing that work was hampered
by a number of factors.  the result of that work is rev -05.
we very much wish -06 to be the last revision so the mib can 
be moved along to iesg.

besides the last call review we had last year on rev -04,
we've not had what i would consider significant review of mib 
modules by the sip community over the life of the draft.      
mibs are not all that interesting to most folks it seems ;)
it does give me some concern that what we are producing may or 
may not have the "right stuff".

so, it would be great if more folks could take a look at 
http://ietf.org/internet-drafts/draft-ietf-sip-mib-05.txt
and provide feedback.  it's still not too late ;)

kevin

> 
> --
> Dean
> 
> > -----Original Message-----
> > From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
> > Sent: Tuesday, March 11, 2003 7:56 PM
> > To: Dean Willis; sip@ietf.org
> > Cc: Gonzalo Camarillo; Jon Peterson
> > (jon.peterson@NeuStar.com); Rohan Mahy
> > Subject: RE: [Sip] Draft agenda, SIP Working Group
> >
> >
> > Is there any intention to include the SIP MIB on the agenda?
> >
> > Thanks,
> >
> > Dan
> >
> 
> _______________________________________________
> 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

-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
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Reply-To: <vkenneally@cit.ie>
From: "Valerie Kenneally" <vkenneally@cit.ie>
To: <sip@ietf.org>
Date: Wed, 12 Mar 2003 18:23:19 -0000
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Subject: [Sip] From/To "tag" parameter
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Hi All,

  Is it possible for a From tag and/or To tag parameter to contain a
semi-colon?

Valerie


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Reply-To: <vkenneally@cit.ie>
From: "Valerie Kenneally" <vkenneally@cit.ie>
To: <sip@ietf.org>
Date: Wed, 12 Mar 2003 19:29:27 -0000
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Subject: [Sip] Via parameters
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Does anyone know where I might be able to find a complete list of all the
parameters that can be appended to the Via line contents e.g. received etc.

-Valerie


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Subject: RE: [Sip] From/To "tag" parameter
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> Is it possible for a From tag and/or To tag 
> parameter to contain a semi-colon?

It is not legal per rfc3261's "token" definition.
A non escaped semi-colon used within the 
tag would be interpreted as a header separator.

_______________________________________________
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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> Is it possible for a From tag and/or To tag 
> parameter to contain a semi-colon?

It is not legal per rfc3261's "token" definition.
A non escaped semi-colon used within the 
tag would be interpreted as a parameter separator
within the header.

_______________________________________________
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: Re: [Sip] Draft agenda, SIP Working Group
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Cc: Dean Willis <dean.willis@softarmor.com>,
        "'Romascanu, Dan (Dan)'" <dromasca@avaya.com>, sip@ietf.org,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>,
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Kevin,

Also, once -06 is out, it would be good to see if there is interest in 
a MIB test at SIPit.

thanks,
-rohan


On Wednesday, March 12, 2003, at 10:11 AM, Kevin Lingle wrote:

> Dean Willis wrote:
>>
>> Well, I don't recall that any of the MIB authors indicated that there 
>> was
>> any need for face time to discuss the MIB (of course, they might hve, 
>> and I
>> might have been asleep). That, of course, may or may not be what the 
>> rest of
>> the working group thinks.
>
> we (mib authors) did not ask for time on the agenda.
>
>>
>> Do we need to discuss the MIB? Any controversial and unresolved 
>> issues that
>> need to be talked through in the full group?
>>
>> It would be "nice" to move the MIB along, now that 3261 and the other 
>> major
>> elements associated with it have basically stabilized.
>
> agreed.   we recently published -05 revision and expect to
> push out -06 because there were a few loose ends.
> after passing wg lc last year w/ rev -04,  we determined it
> prudent to make the mib modules relevant to rfc3261.
> our ability to get around do doing that work was hampered
> by a number of factors.  the result of that work is rev -05.
> we very much wish -06 to be the last revision so the mib can
> be moved along to iesg.
>
> besides the last call review we had last year on rev -04,
> we've not had what i would consider significant review of mib
> modules by the sip community over the life of the draft.
> mibs are not all that interesting to most folks it seems ;)
> it does give me some concern that what we are producing may or
> may not have the "right stuff".
>
> so, it would be great if more folks could take a look at
> http://ietf.org/internet-drafts/draft-ietf-sip-mib-05.txt
> and provide feedback.  it's still not too late ;)
>
> kevin
>
>>
>> --
>> Dean
>>
>>> -----Original Message-----
>>> From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
>>> Sent: Tuesday, March 11, 2003 7:56 PM
>>> To: Dean Willis; sip@ietf.org
>>> Cc: Gonzalo Camarillo; Jon Peterson
>>> (jon.peterson@NeuStar.com); Rohan Mahy
>>> Subject: RE: [Sip] Draft agenda, SIP Working Group
>>>
>>>
>>> Is there any intention to include the SIP MIB on the agenda?
>>>
>>> Thanks,
>>>
>>> Dan
>>>
>>
>> _______________________________________________
>> 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
>
> -- 
> =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
>  Kevin R. Lingle       919.392.2029
> =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=

_______________________________________________
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: RE: [Sip] Draft agenda, SIP Working Group
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Kevin Lingle" <klingle@cisco.com>,
        "Dean Willis" <dean.willis@softarmor.com>
Cc: <sip@ietf.org>, "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>,
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In order to shorten the number of versions ahead, I suggest that the authors of the draft have a look (if they did not do it yet) at the MIB Review guidelines Internet-Draft http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-guidelines-01.txt. This is not final yet, but it is quite stable, and me or whoever will be assigned to review this MIB  will use it for the MIB quality review (from the SNMP / SMI point of view). 

Another issue is that because the MIB went through rather radical changes between versions 04 and 05, t will probably be appropriate to run this version or the next one through another WG last call. This may increase the chances to get closer to "the right stuff". Is this the plan?

Thanks,

Dan


> -----Original Message-----
> From: Kevin Lingle [mailto:klingle@cisco.com]
> Sent: Wednesday, March 12, 2003 8:12 PM
> To: Dean Willis
> Cc: Romascanu, Dan (Dan); sip@ietf.org; 'Gonzalo Camarillo'; 
> 'Jon Peterson (jon.peterson@NeuStar.com)'; 'Rohan Mahy'
> Subject: Re: [Sip] Draft agenda, SIP Working Group
> 
> 
> Dean Willis wrote:
> > 
> > Well, I don't recall that any of the MIB authors indicated 
> that there was
> > any need for face time to discuss the MIB (of course, they 
> might hve, and I
> > might have been asleep). That, of course, may or may not be 
> what the rest of
> > the working group thinks.
> 
> we (mib authors) did not ask for time on the agenda.
> 
> > 
> > Do we need to discuss the MIB? Any controversial and 
> unresolved issues that
> > need to be talked through in the full group?
> > 
> > It would be "nice" to move the MIB along, now that 3261 and 
> the other major
> > elements associated with it have basically stabilized.
> 
> agreed.   we recently published -05 revision and expect to 
> push out -06 because there were a few loose ends.  
> after passing wg lc last year w/ rev -04,  we determined it
> prudent to make the mib modules relevant to rfc3261.
> our ability to get around do doing that work was hampered
> by a number of factors.  the result of that work is rev -05.
> we very much wish -06 to be the last revision so the mib can 
> be moved along to iesg.
> 
> besides the last call review we had last year on rev -04,
> we've not had what i would consider significant review of mib 
> modules by the sip community over the life of the draft.      
> mibs are not all that interesting to most folks it seems ;)
> it does give me some concern that what we are producing may or 
> may not have the "right stuff".
> 
> so, it would be great if more folks could take a look at 
> http://ietf.org/internet-drafts/draft-ietf-sip-mib-05.txt
> and provide feedback.  it's still not too late ;)
> 
> kevin
> 
> > 
> > --
> > Dean
> > 
> > > -----Original Message-----
> > > From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
> > > Sent: Tuesday, March 11, 2003 7:56 PM
> > > To: Dean Willis; sip@ietf.org
> > > Cc: Gonzalo Camarillo; Jon Peterson
> > > (jon.peterson@NeuStar.com); Rohan Mahy
> > > Subject: RE: [Sip] Draft agenda, SIP Working Group
> > >
> > >
> > > Is there any intention to include the SIP MIB on the agenda?
> > >
> > > Thanks,
> > >
> > > Dan
> > >
> > 
> > _______________________________________________
> > 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
> 
> -- 
> =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
> =-=-=-=-=
>  Kevin R. Lingle       919.392.2029
> =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
> =-=-=-=-=
> 
_______________________________________________
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: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
CC: Dean Willis <dean.willis@softarmor.com>, sip@ietf.org,
        Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        "Jon Peterson (jon.peterson@NeuStar.com)" <jon.peterson@neustar.biz>,
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"Romascanu, Dan (Dan)" wrote:
> 
> In order to shorten the number of versions ahead, I suggest that the authors of the draft have a look (if they did not do it yet) at the MIB Review guidelines Internet-Draft http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-guidelines-01.txt. This is not final yet, but it is quite stable, and me or whoever will be assigned to review this MIB  will use it for the MIB quality review (from the SNMP / SMI point of view).

we did for rev -05.  jean-francios mule digested the guidelines and 
made sure our draft was aligned.

> 
> Another issue is that because the MIB went through rather radical changes between versions 04 and 05, t will probably be appropriate to run this version or the next one through another WG last call. This may increase the chances to get closer to "the right stuff". Is this the plan?

yes.  we anticipate needing another wg lc run.

kevin
> 
> Thanks,
> 
> Dan
> 
> > -----Original Message-----
> > From: Kevin Lingle [mailto:klingle@cisco.com]
> > Sent: Wednesday, March 12, 2003 8:12 PM
> > To: Dean Willis
> > Cc: Romascanu, Dan (Dan); sip@ietf.org; 'Gonzalo Camarillo';
> > 'Jon Peterson (jon.peterson@NeuStar.com)'; 'Rohan Mahy'
> > Subject: Re: [Sip] Draft agenda, SIP Working Group
> >
> >
> > Dean Willis wrote:
> > >
> > > Well, I don't recall that any of the MIB authors indicated
> > that there was
> > > any need for face time to discuss the MIB (of course, they
> > might hve, and I
> > > might have been asleep). That, of course, may or may not be
> > what the rest of
> > > the working group thinks.
> >
> > we (mib authors) did not ask for time on the agenda.
> >
> > >
> > > Do we need to discuss the MIB? Any controversial and
> > unresolved issues that
> > > need to be talked through in the full group?
> > >
> > > It would be "nice" to move the MIB along, now that 3261 and
> > the other major
> > > elements associated with it have basically stabilized.
> >
> > agreed.   we recently published -05 revision and expect to
> > push out -06 because there were a few loose ends.
> > after passing wg lc last year w/ rev -04,  we determined it
> > prudent to make the mib modules relevant to rfc3261.
> > our ability to get around do doing that work was hampered
> > by a number of factors.  the result of that work is rev -05.
> > we very much wish -06 to be the last revision so the mib can
> > be moved along to iesg.
> >
> > besides the last call review we had last year on rev -04,
> > we've not had what i would consider significant review of mib
> > modules by the sip community over the life of the draft.
> > mibs are not all that interesting to most folks it seems ;)
> > it does give me some concern that what we are producing may or
> > may not have the "right stuff".
> >
> > so, it would be great if more folks could take a look at
> > http://ietf.org/internet-drafts/draft-ietf-sip-mib-05.txt
> > and provide feedback.  it's still not too late ;)
> >
> > kevin
> >
> > >
> > > --
> > > Dean
> > >
> > > > -----Original Message-----
> > > > From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
> > > > Sent: Tuesday, March 11, 2003 7:56 PM
> > > > To: Dean Willis; sip@ietf.org
> > > > Cc: Gonzalo Camarillo; Jon Peterson
> > > > (jon.peterson@NeuStar.com); Rohan Mahy
> > > > Subject: RE: [Sip] Draft agenda, SIP Working Group
> > > >
> > > >
> > > > Is there any intention to include the SIP MIB on the agenda?
> > > >
> > > > Thanks,
> > > >
> > > > Dan
> > > >
> > >
> > > _______________________________________________
> > > 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
> >
> > --
> > =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
> > =-=-=-=-=
> >  Kevin R. Lingle       919.392.2029
> > =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
> > =-=-=-=-=
> >

-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
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Subject: RE: [Sip] Draft agenda, SIP Working Group
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From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>,
        "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
Cc: "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>,
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Dan wrote:
> Is there any intention to include the SIP MIB on the agenda? 
> 
> Thanks,
> 
> Dan

We have not requested a timeslot since draft05 is only a partial update
to get the mib realign with rfc3261. I'll be at the meeting for the
entire week though and we can certainly get together and do an indepth
review with Kevin.

Jean-Francois.


 
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From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
To: "Kevin Lingle" <klingle@cisco.com>,
        "Romascanu, Dan (Dan)" <dromasca@avaya.com>
Cc: "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>,
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I concur with Kevin.

> "Romascanu, Dan (Dan)" wrote:
> > 
> > In order to shorten the number of versions ahead, I suggest 
> that the 
> > authors of the draft have a look (if they did not do it yet) at the 
> > MIB Review guidelines Internet-Draft 
> > 
> http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-guidelin
> > es-01.txt. This is not final yet, but it is quite stable, and me or 
> > whoever will be assigned to review this MIB  will use it 
> for the MIB 
> > quality review (from the SNMP / SMI point of view).

Kevin wrote: 
> we did for rev -05.  jean-francios mule digested the guidelines and 
> made sure our draft was aligned.
To be precise, we did address *some* of the ops-mib-review guidelines
(Bert has been showing us the way in the ipcdn wg so we took it into
account as part of this draft update). What we did revise per the mib
guidelines are:
 - boilerplate
 - module identity copyright
 - normative vs. informative references
 - counters: we looked at the way we use counters with Kevin
 - I think we're clean on inetaddress'es & rfc3291 but haven't looked
back at the details

That is all I believe and we may have missed a couple.
Jean-Francois.
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From mailnull@www1.ietf.org  Thu Mar 13 03:42:42 2003
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Dear all,
RFC 3325 (Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks) is stating that: 

The P-Asserted-Identity header field is used among trusted SIP
entities (typically intermediaries) to carry the identity of the user
sending a SIP message as it was verified by authentication.

From my point of view this statement includes that Requests and Responses can be verified within the SIP domain. But is this possible and described within SIP that a Response will be also verified by authenticated if needed?

Background of this question is, if it is possible to interwork a P-Asserted-ID in a Response (if this could be to a connected number parameter within PSTN. The Connected number within PSTN must be a trusted number.


Thank You 

Best Regards

Roland Jesske




Deutsche Telekom AG
T Com Zentrale
Roland Jesske, T38-12
Section T38; Signalling, Gateways and Switching Systems 
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940 
Fax:    +49 6151 83-4577 
 <mailto:r.jesske@telekom.de>



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Hi....

  For a SIP proxy, on what conditions it should generate 5xx series =
responses and what conditions it should forward 5xx responses?

Any call flow scenarios are available?

Regards,
Jayasree.




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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
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<DIV><FONT face=3DArial size=3D2>Hi....</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; For a SIP proxy, on what =
conditions it=20
should generate 5xx series responses and what conditions it should =
forward 5xx=20
responses?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Any call flow scenarios are =
available?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Regards,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Jayasree.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

<br>--------------------------------------------------------------<br>Dexcel Electronics Designs (P) Ltd., Bangalore, India<br>
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There seems to be a problem with implementing SIP digest authentication as 
described in RFC 2617 (HTTP authentication) while using LDAP servers at 
the back-end. The authenticating server will receive the hashed password 
from the client, but cannot compare this with the password in the 
directory, because LDAP servers do not support a uniform way for querying 
passwords.

For example, some LDAP servers return the user's password hashed (using 
MD5,
SHA or something else) but this hashed value can not be compared with the 
hashed password from the client. Even if the same hashing algorithm is 
used in the client and in the LDAP server it will not be possible to 
compare the hashed passwords since LDAP hashes only the password, while 
the client-supplied hash includes more info in the digest. The only way 
that it can be done is by getting the password in cleartext from LDAP, and 
do our own digest, but most LDAP servers do not allow that, not even for 
the directory admin.

Another approach is to forward the client-supplied credentials (hashed 
password,
etc) to the LDAP server as a SASL request (RFC 2222). According to RFC 
2829
(Authentication methods for LDAP) section 4 "Implementations providing
password-based authenticated access MUST support authentication using the
DIGEST-MD5 SASL mechanism". Unfortunately, most LDAP servers do not follow 
that either.

It would have been nice if SIP supported cleartext password 
authentication,
like HTTP does, at least for TLS-encrypted connections. As I understand, 
it does not.

The note above may be more appropriate for the SIP implementors list but 
it seems that there is an inherent problem that we need to solve in SIP 
authentication itself or assume that all LDAP servers will be modified 
which might be a too optimistic assumption.

Avshalom

 
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This list is for NEW development of the core SIP Protocol
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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Jesske, R'" <R.Jesske@telekom.de>, sip@ietf.org
Cc: "Alexeitsev, D" <D.Alexeitsev@telekom.de>
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Roland,

The P-Asserted-Identity can be used in responses.

The extract you quote was intended to mean that the 'identity of the user'
was verified by authentication, rather than the 'message' itself being
verified by authentication (I guess the 'it' in that extract could be
ambiguous). It doesn't say how or when the authentication was carried out.

For example, if a proxy has authenticated a user by some secure means and
then later if the same proxy can securely determine that a particular
response came from the same user, THEN that proxy can use the result of the
initial authentication to insert a P-Asserted-Identity in the later
response.

This could be achieved, for example, by use of some kind of integrity
protection on the UA-Proxy link.

As to whether you could interwork this P-Asserted-Identity to the Connected
Number within the PSTN, this depends on whether the rules of your particular
Trust Domain for Asserted Identity (in terms of what it means for a number
to be 'trusted') match the rules within the PSTN (potentially the particular
national PSTN) for Connected Number.

Certainly it is possible and conceivable for the rules to match - and this
would motivate including a protocol mapping within any interworking
specification you might happen to be writing - but it should be clear that
(as with CLI), whether this mapping takes place in a given situation depends
on the policies of the Trust Domains.

...Mark


> -----Original Message-----
> From: Jesske, R [mailto:R.Jesske@telekom.de]
> Sent: 13 March 2003 08:39
> To: sip@ietf.org
> Cc: Watson, Mark [MOP:EP10:EXCH]; Alexeitsev, D
> Subject: Authintification of Responses
> 
> 
> Dear all,
> RFC 3325 (Private Extensions to the Session Initiation 
> Protocol (SIP) for Asserted Identity within Trusted Networks) 
> is stating that: 
> 
> The P-Asserted-Identity header field is used among trusted SIP
> entities (typically intermediaries) to carry the identity of the user
> sending a SIP message as it was verified by authentication.
> 
> From my point of view this statement includes that Requests 
> and Responses can be verified within the SIP domain. But is 
> this possible and described within SIP that a Response will 
> be also verified by authenticated if needed?
> 
> Background of this question is, if it is possible to 
> interwork a P-Asserted-ID in a Response (if this could be to 
> a connected number parameter within PSTN. The Connected 
> number within PSTN must be a trusted number.
> 
> 
> Thank You 
> 
> Best Regards
> 
> Roland Jesske
> 
> 
> 
> 
> Deutsche Telekom AG
> T Com Zentrale
> Roland Jesske, T38-12
> Section T38; Signalling, Gateways and Switching Systems 
> Am Kavalleriesand 3, 64295 Darmstadt, Germany
> Phone:  +49 6151 83-5940 
> Fax:    +49 6151 83-4577 
>  <mailto:r.jesske@telekom.de>
> 
> 
> 
> 

------_=_NextPart_001_01C2E978.FB1CC34E
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>RE: Authintification of Responses</TITLE>
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<BODY>

<P><FONT SIZE=3D2>Roland,</FONT>
</P>

<P><FONT SIZE=3D2>The P-Asserted-Identity can be used in =
responses.</FONT>
</P>

<P><FONT SIZE=3D2>The extract you quote was intended to mean that the =
'identity of the user' was verified by authentication, rather than the =
'message' itself being verified by authentication (I guess the 'it' in =
that extract could be ambiguous). It doesn't say how or when the =
authentication was carried out.</FONT></P>

<P><FONT SIZE=3D2>For example, if a proxy has authenticated a user by =
some secure means and then later if the same proxy can securely =
determine that a particular response came from the same user, THEN that =
proxy can use the result of the initial authentication to insert a =
P-Asserted-Identity in the later response.</FONT></P>

<P><FONT SIZE=3D2>This could be achieved, for example, by use of some =
kind of integrity protection on the UA-Proxy link.</FONT>
</P>

<P><FONT SIZE=3D2>As to whether you could interwork this =
P-Asserted-Identity to the Connected Number within the PSTN, this =
depends on whether the rules of your particular Trust Domain for =
Asserted Identity (in terms of what it means for a number to be =
'trusted') match the rules within the PSTN (potentially the particular =
national PSTN) for Connected Number.</FONT></P>

<P><FONT SIZE=3D2>Certainly it is possible and conceivable for the =
rules to match - and this would motivate including a protocol mapping =
within any interworking specification you might happen to be writing - =
but it should be clear that (as with CLI), whether this mapping takes =
place in a given situation depends on the policies of the Trust =
Domains.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Jesske, R [<A =
HREF=3D"mailto:R.Jesske@telekom.de">mailto:R.Jesske@telekom.de</A>]</FON=
T>
<BR><FONT SIZE=3D2>&gt; Sent: 13 March 2003 08:39</FONT>
<BR><FONT SIZE=3D2>&gt; To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: Watson, Mark [MOP:EP10:EXCH]; Alexeitsev, =
D</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Authintification of Responses</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Dear all,</FONT>
<BR><FONT SIZE=3D2>&gt; RFC 3325 (Private Extensions to the Session =
Initiation </FONT>
<BR><FONT SIZE=3D2>&gt; Protocol (SIP) for Asserted Identity within =
Trusted Networks) </FONT>
<BR><FONT SIZE=3D2>&gt; is stating that: </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The P-Asserted-Identity header field is used =
among trusted SIP</FONT>
<BR><FONT SIZE=3D2>&gt; entities (typically intermediaries) to carry =
the identity of the user</FONT>
<BR><FONT SIZE=3D2>&gt; sending a SIP message as it was verified by =
authentication.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; From my point of view this statement includes =
that Requests </FONT>
<BR><FONT SIZE=3D2>&gt; and Responses can be verified within the SIP =
domain. But is </FONT>
<BR><FONT SIZE=3D2>&gt; this possible and described within SIP that a =
Response will </FONT>
<BR><FONT SIZE=3D2>&gt; be also verified by authenticated if =
needed?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Background of this question is, if it is =
possible to </FONT>
<BR><FONT SIZE=3D2>&gt; interwork a P-Asserted-ID in a Response (if =
this could be to </FONT>
<BR><FONT SIZE=3D2>&gt; a connected number parameter within PSTN. The =
Connected </FONT>
<BR><FONT SIZE=3D2>&gt; number within PSTN must be a trusted =
number.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Thank You </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Best Regards</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Roland Jesske</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Deutsche Telekom AG</FONT>
<BR><FONT SIZE=3D2>&gt; T Com Zentrale</FONT>
<BR><FONT SIZE=3D2>&gt; Roland Jesske, T38-12</FONT>
<BR><FONT SIZE=3D2>&gt; Section T38; Signalling, Gateways and Switching =
Systems </FONT>
<BR><FONT SIZE=3D2>&gt; Am Kavalleriesand 3, 64295 Darmstadt, =
Germany</FONT>
<BR><FONT SIZE=3D2>&gt; Phone:&nbsp; +49 6151 83-5940 </FONT>
<BR><FONT SIZE=3D2>&gt; Fax:&nbsp;&nbsp;&nbsp; +49 6151 83-4577 </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &lt;<A =
HREF=3D"mailto:r.jesske@telekom.de">mailto:r.jesske@telekom.de</A>&gt;</=
FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

</BODY>
</HTML>
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_______________________________________________
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From mailnull@www1.ietf.org  Thu Mar 13 11:02:21 2003
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Jesske, R'" <R.Jesske@telekom.de>, sip@ietf.org
Cc: mwatson@nortelnetworks.com, "Alexeitsev, D" <D.Alexeitsev@telekom.de>
Subject: RE: [Sip] Authintification of Responses
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The P-Asserted-Identity is generated by a trusted entity. In order to perform this action, the trusted entity must know who the user is, hence the concept of authentication. However that authentication does not need to be performed over each individual message. For example, in 3GPP, that authentication is performed at the time of sending a REGISTER from that user, and subsequent requests and responses are sent using the agreed security mechanism (see RFC 3329), and therefore the P-Asserted-Identity of the destination user can be added by the trusted proxy associated with that destination user to any response (or request) generated by that destination user.

For the PSTN interworking case, we effectively have two trust domains. One in the ISUP network and the other in the SIP network. Assuming that equivalent levels of trust exist between those two trust domains, which entirely within a public network environment can be assumed, then a Connected number on the BICC / ISUP side can result in a P-Asserted-Identity in a SIP response.

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Jesske, R [mailto:R.Jesske@telekom.de]
> Sent: 13 March 2003 08:39
> To: sip@ietf.org
> Cc: mwatson@nortelnetworks.com; Alexeitsev, D
> Subject: [Sip] Authintification of Responses
> 
> 
> Dear all,
> RFC 3325 (Private Extensions to the Session Initiation 
> Protocol (SIP) for Asserted Identity within Trusted Networks) 
> is stating that: 
> 
> The P-Asserted-Identity header field is used among trusted SIP
> entities (typically intermediaries) to carry the identity of the user
> sending a SIP message as it was verified by authentication.
> 
> From my point of view this statement includes that Requests 
> and Responses can be verified within the SIP domain. But is 
> this possible and described within SIP that a Response will 
> be also verified by authenticated if needed?
> 
> Background of this question is, if it is possible to 
> interwork a P-Asserted-ID in a Response (if this could be to 
> a connected number parameter within PSTN. The Connected 
> number within PSTN must be a trusted number.
> 
> 
> Thank You 
> 
> Best Regards
> 
> Roland Jesske
> 
> 
> 
> 
> Deutsche Telekom AG
> T Com Zentrale
> Roland Jesske, T38-12
> Section T38; Signalling, Gateways and Switching Systems 
> Am Kavalleriesand 3, 64295 Darmstadt, Germany
> Phone:  +49 6151 83-5940 
> Fax:    +49 6151 83-4577 
>  <mailto:r.jesske@telekom.de>
> 
> 
> 
> _______________________________________________
> 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 list is for NEW development of the core SIP Protocol
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Subject: [Sip] changes in caller prefs
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Folks,

As you may have seen, I have submitted an update to the caller prefs 
spec. A lot of work went into this draft since the last rev. There are a 
lot of changes, which are summarized below. Generally speaking, the 
reasons for many of these changes were to make caller prefs functional 
for many of the use cases that we are considering. The reason that the 
draft took so long is that it still cannot cover all of them, and I 
believe significantly more complexity would be needed to support all of 
them. However, I have concluded that we should simply declare those out 
of scope for now, primarily because we dont really know what people are 
going to use caller prefs for. We need to gain more experience with it.

The specific cases which arent well covered at this point are those 
where you wish to preferentially route to a UA that has declared a 
capability that is the best explicit match for the callers preference. 
For example, if the caller asks for audio, video, messaging, the call 
would be routed first to a UA that said it did audio and video (but said 
nothing about messaging), and second to a UA that said it did audio (but 
said nothign about messaging or video).

If we agree that this feature is out of scope, I believe caller prefs as 
written is done, and there are no open issues. A lot of good fixes went 
into this version, and its time to declare victory. I'll be talking more 
about this during the ietf meeting.


Changes:

* added Paul K as co-author to acknowledge his contributions to the
work.

* we always had this problem that there was no way to determine which
contact parameters were caller prefs. Enough folks have complained
that I put a solution in place. The parameters defined in this spec
itself can appear as they are. However, any other parameters
(including ones registered already) have to appera with a leading
+. So, if you invent a new media tag foo, you would use it in the
contact params thusly:

Contact: <sip:user@domain>;audio="TRUE";+foo="FALSE"

* eliminated the Require-Contact header field. Replaced it with the
new require parameter of the Accept-Contact header field.

* added a weighting to the proxy computations, so that contacts that
are a better match to the accept-contact header result in a higher
q-value.

* aligned the "mime" media type feature tags around the SDP media types
instead. Thus, the values defined include audio, video, application,
data and control.

* added text to deal with the possible future definition of new SDP
media types:

If a new SDP media type were to be defined, such as ``message'', a new
feature tag registration SHOULD be created for it. The name of the
feature tag MUST equal that of the media type, unless there is an
unlikely naming collision between the new media type and an existing
feature tag registration. As a result of this, implementations can
safely contruct caller preferences and callee capabilities for the new
media type before it is registered, as long as there is no naming
conflict.

* added the new "explicit" parameter to the Accept-Contact header
field. It is used to signal that a contact must match explicitly.

* changed the "voicemail" tag to "msgserver" since the voicemail term
is voice specific.

* the q-value parameter can now appear anywhere in the header field
value; it need not occur after the feature tags, as before.

* allow the feature tag to appaer with no value. i.e.,:

Accept-Contact: *;video;audio

when present in this form, it means ="TRUE".

* allow for the value to be a comma separate lists of numerics and 
booleans,
not just tokens, as previously. However, you still can only have a
single string value.

* you can negate booleans and numerics now (used to be only token
equalities could be negated)

* fixed the syntax bug which would result in quoted strings being
double-quoted.

* changed the range syntax to use a colon, and removed the useless "R"

* numbers were formerly represented in fractional form (i.e., 3/4)
instead of decimal (0.75). THis was just because thats how rfc2533
represents them. Several folks complained, so I changed it to a
decimal form. The section on mapping to rfc2533 describes the
conversion between the two. Of course, in practice, it is probably
never needed to convert.

* Changed the syntax of the feature tag itself, adding support for
escaping and the + sign

* Allowed for generic-param in Accept and Reject contact, to allow for
future extensions.

* Mandated that, when numeric parameters are present in a feature
param, they be representable as an ANSI C double.

* Changed the title to reflect sip-guidlines guidance

* changed the header table column for proxies from r to ar. THis means
a proxy can add or modify a value. Discussion of this is added in the
proxy processing section. Proxies can't remove or modify header field
values because of s/mime, and this is pointed out.

* removed implicit preferences for media, languages and priority per
list discussion.

* added a note saying that you are now allowed to express explicit
preferences for methods, event packages or schemes not defined in ietf
standards track protocols.

* added UA processing of OPTIONS. The contact header in the 200 OK to
options can contain feature parameters.

* added isfocus parameter. This is needed for the conferencing
work. It seemed easier to add it in here than to issue a separate RFC
just to register it.

* if application of caller prefs results in the elimination of all
registered contacts, the algorithm is re-run, but this time, any
implict preferences are not at the require strength. This is as per
list discussion.

* added an extensive example to the proxy matching section

* note that if a UA
registers against two separate addresses-of-record, and the contacts
registered for each have different capabilities, a UA MUST use
different URIs in each registration. This is so that the UA can
uniquely determine the feature set that is associated with the request
URI of an incoming request.

* updated table 2/3 to also include MESSAGE.

* added a nice figure to explain the proxy matching operation

* added an example service that is enabeld by called prefs to help
motivate interest.

* clients registering caller prefs params SHOULD use sips for privacy
services, and callers SHOULD use sips for integrity protection (iesg
wants to see baseline mechanisms for addressign all threats described
in security considerations)

* Previously, the uri-domain and uri-user feature tags had very
special processing. Only the UA would compute them (they were not real
feature tags) for registered
contacts, and apply them to any Accept or Reject-Contact. This was to
solve the problem previously solved by the "only=true" flag. Namely, a
proxy couldn't reject a contact because of a mismatch in the URI,
since a downstream forwarding operation might actually route to that
contact. As it turns out, this problem is not specific to the URI
parameters. Generally, it is only safe to compute caller preferences
when they refer to a URI that is a UA, as opposed to an AOR. So,
several changes were made. First, uri-domain and uri-user would made
full feature tags. Like all the others, they are explicitly registered
by a UA. A UA can now opt to register NO feature parameters for a
contact. Such contacts are "immune" from caller preferences
processing. Thus, when a UA registers an AOR, it includes no feature
parameters. When it registers a contact for a UA, it includes feature
parameters. It can include the uri-user and uri-domain
parameters. They must be explicit now. These receive no special
processing at the proxy. The only "exception" is that in Accept or
Reject-Contact, they can appear as parameters, or appear as the URI
itself. To some degree, that makes the URI redundant, but I retained
it for backwards compatibility.

* special treatment of the schemes parameter by the proxy has been
removed. Formerly, if it wasnt explicitly indicated, the proxy
computed an implicit schemes parameter. No longer, for the same
problem as above.

* implicit preferences are only computed by the proxy if no explicit
preferences at all were provided. This eliminates the need for the
proxy to figure out how to combine the explicit and implicit
preferences. Text was added the the UA section RECOMMENDing inclusion
of method and event preferences, and to do so by adding a term to all
existing A-C predicates.

* UA registration procedures explicitly call out the handling of
contacts that are not UA instances. There is a good discussion on the
meaning of registering an AOR, and how to associate feature parameters
with it.

* the spec was unclear about exactly WHEN in the UAS processing steps
the caller preferences checks are done. It actually matters. It needs
to be done after authentication/authorization, but before the rest of
8.2.1 of rfc3261. Also, the spec didnt state what to do if there was
or wasnt a match. If there is no match, send 480. If there was a
match, continue with request processing.

* the spec had evolved over time to an inconsistent usage of the
schemes feature parameter. There are two interpretations. One is "this
is the scheme of the URI associated with my UA", making it a peer of
the uri-domain and uri-user parameters. Another interptation is "I
know how to handle redirects to URIs of these schemes, or how to
process them if they were entered on my UI". The desired
interpretation is the latter. That has been clarified, and usage made
consistent with this view throughout the spec.

* clarify that quoted strings in rfc2533 are matched using CS
matching, and tokens using CI.

* provided a warning that the "application" feature tag is not that
useful in genreal.

* clarified that the priority feature tag indicates that a UA can
support calls of a certain priority and higher.

* added sip-extensions feature tag

* Accept-Contact and Reject-Contact used to have a URI or a * as the
value, along with the feature params. With the explicit uri-user and
uri-domain params, continued use of the URI introduced redundancy in
the header, and also made the text more confusing. So, it has been
removed. The two headers ALWAYS have a value of *, followed by the
feature params, which can include the uri-user and uri-domain if the
caller wants to express preferences on those. The * itself can't be
removed, since the sip
guidelines spec mandates that a token be present in the value.

* added some guidance on proper redirection using caller prefs. If you
redirect and include q-values that were the result of caller prefs
modification, the redirect can't include the caller prefs feature
parameters. If they did, the upstream proxy would apply the same
caller prefs operation again!

* substantial rework of the algorithm used to combine the various
q-values involved. Now, a specific algorithm is provided, engineered
to have particular properties.


-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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From mailnull@www1.ietf.org  Fri Mar 14 10:24:59 2003
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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Date: Fri, 14 Mar 2003 15:21:06 -0000
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Subject: [Sip] S/MIME signatures
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All,

RFC3261, 23.3 says:

	"multipart/signed" MUST be used only with CMS detached signatures.

But according to RFC2633 (S/MIME) "multipart/signed" is only used with
detached signatures anyway. From RFC2633, Section 3.4.3:

   "The multipart/signed MIME type has two parts. The first part contains
the
   MIME entity that is signed; the second part contains the "detached
   signature" CMS SignedData object in which the encapContentInfo
   eContent field is absent."

So, what is the intention of the statement from RFC3261 ?

Is it meant instead that the case where the content is encapsulated in the
CMS SignedData is NOT allowed in SIP ? This would use
"application/pkcs7-mime", not "multipart/signed".

Regards...Mark


------_=_NextPart_001_01C2EA3C.EEBFFE94
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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<HEAD>
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charset=3Diso-8859-1">
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5.5.2656.31">
<TITLE>S/MIME signatures</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>All,</FONT>
</P>

<P><FONT SIZE=3D2>RFC3261, 23.3 says:</FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&quot;multipart/signed&quot; MUST be used only with CMS =
detached signatures.</FONT>
</P>

<P><FONT SIZE=3D2>But according to RFC2633 (S/MIME) =
&quot;multipart/signed&quot; is only used with detached signatures =
anyway. From RFC2633, Section 3.4.3:</FONT></P>

<P><FONT SIZE=3D2>&nbsp;&nbsp; &quot;The multipart/signed MIME type has =
two parts. The first part contains the</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; MIME entity that is signed; the second =
part contains the &quot;detached</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; signature&quot; CMS SignedData object =
in which the encapContentInfo</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; eContent field is absent.&quot;</FONT>
</P>

<P><FONT SIZE=3D2>So, what is the intention of the statement from =
RFC3261 ?</FONT>
</P>

<P><FONT SIZE=3D2>Is it meant instead that the case where the content =
is encapsulated in the CMS SignedData is NOT allowed in SIP ? This =
would use &quot;application/pkcs7-mime&quot;, not =
&quot;multipart/signed&quot;.</FONT></P>

<P><FONT SIZE=3D2>Regards...Mark</FONT>
</P>

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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'sip@ietf.org'" <sip@ietf.org>
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Subject: [Sip] S/MIME & Selective Anonymity
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All,

Firstly, apologies if this has all been discussed before - I couldn't find a
searchable version of the archive to check.

RFC3261 describes the use of S/MIME to provide selective anonymity in
Section 23.4.3. This is the case where I wish to establish a session and
remain anonymous to all but the intended recipient of the session (and any
proxies I authenticate with, of course).

The UAC puts 'anonymous' in the From: field and includes an encrypted
message/sip MIME body containing the real From value.

The text then seems to recommend that the result is signed by the UAC:

   "In order to provide end-to-end integrity, encrypted "message/sip"
   MIME bodies SHOULD be signed by the sender.  This creates a
   "multipart/signed" MIME body that contains an encrypted body and a
   signature, both of type "application/pkcs7-mime". 

[BTW, this last bit seems to be in error since the signature part should be
"application/pkcs7-signature", unless I have missed something.]

But now it appears to be possible to identify the sender based on the
SignerIdentifier in the signerInfo of the CMS SignedData message or the
subjectAltName of the certificate, which according to 23.2 MUST be included.

Have I missed something here ?

Wouldn't it be better for the UAC to sign the message/sip MIME body first,
and then encrypt ?  I understand that an S/MIME compliant receiver should
accept this, but should this be described in the SIP context ?

Even if it were hard to identify a user based just on the SignerIdentifier,
it would at least be possible to recognise multiple requests from the same
user as indeed being from the same user, which is also a privacy breach.

Regards...Mark


 

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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>S/MIME &amp; Selective Anonymity</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>All,</FONT>
</P>

<P><FONT SIZE=3D2>Firstly, apologies if this has all been discussed =
before - I couldn't find a searchable version of the archive to =
check.</FONT></P>

<P><FONT SIZE=3D2>RFC3261 describes the use of S/MIME to provide =
selective anonymity in Section 23.4.3. This is the case where I wish to =
establish a session and remain anonymous to all but the intended =
recipient of the session (and any proxies I authenticate with, of =
course).</FONT></P>

<P><FONT SIZE=3D2>The UAC puts 'anonymous' in the From: field and =
includes an encrypted message/sip MIME body containing the real From =
value.</FONT></P>

<P><FONT SIZE=3D2>The text then seems to recommend that the result is =
signed by the UAC:</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp; &quot;In order to provide end-to-end =
integrity, encrypted &quot;message/sip&quot;</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; MIME bodies SHOULD be signed by the =
sender.&nbsp; This creates a</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; &quot;multipart/signed&quot; MIME body =
that contains an encrypted body and a</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; signature, both of type =
&quot;application/pkcs7-mime&quot;. </FONT>
</P>

<P><FONT SIZE=3D2>[BTW, this last bit seems to be in error since the =
signature part should be &quot;application/pkcs7-signature&quot;, =
unless I have missed something.]</FONT></P>

<P><FONT SIZE=3D2>But now it appears to be possible to identify the =
sender based on the SignerIdentifier in the signerInfo of the CMS =
SignedData message or the subjectAltName of the certificate, which =
according to 23.2 MUST be included.</FONT></P>

<P><FONT SIZE=3D2>Have I missed something here ?</FONT>
</P>

<P><FONT SIZE=3D2>Wouldn't it be better for the UAC to sign the =
message/sip MIME body first, and then encrypt ?&nbsp; I understand that =
an S/MIME compliant receiver should accept this, but should this be =
described in the SIP context ?</FONT></P>

<P><FONT SIZE=3D2>Even if it were hard to identify a user based just on =
the SignerIdentifier, it would at least be possible to recognise =
multiple requests from the same user as indeed being from the same =
user, which is also a privacy breach.</FONT></P>

<P><FONT SIZE=3D2>Regards...Mark</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&nbsp;</FONT>
</P>

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Cc: "'jari.arkko@ericsson.com'" <jari.arkko@ericsson.com>,
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        "'Gonzalo.Camarillo@ericsson.com'" <Gonzalo.Camarillo@ericsson.com>,
        "'aki.niemi@nokia.com'" <aki.niemi@nokia.com>,
        "'tao.haukka@nokia.com'"
	 <tao.haukka@nokia.com>
Date: Fri, 14 Mar 2003 15:46:29 -0000
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Subject: [Sip] Some comments on RFC 3329 (Security Mechanism Agreement for the S
 ession Initiation Protocol)
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In looking at this RFC we detect some issues that ought to be resolved even though this is a published RFC. There is also an issue that is raised with regard to RFC 3261.

1)	Section 2.6, table 1. The Security-Verify header is not included in the PRACK according to the table. It is our belief that the header should be applicable to all methods except ACK and CANCEL, where we agree that it makes no sense. 

2)	Section 2.3.1, 9th paragraph. In relation to the above comment, this text states:

   All the subsequent SIP requests sent by the client to that server
   SHOULD make use of the security mechanism initiated in the previous
   step.  These requests MUST contain a Security-Verify header field
   that mirrors the server's list received previously in the Security-
   Server header field.  These requests MUST also have both a Require
   and Proxy-Require header fields with the value "sec-agree".

This text should explicitly exclude the ACK and CANCEL requests. Something along the lines of:

   All the subsequent SIP requests except the ACK request and the CANCEL 
   request sent by the client to that server...

3)	Section 6.4 defines the new status code 494 (By the way what is the distinction between the term "status code" and "response code" - RFC 3261 seems to use both interchangeably). Unlike other codes in RFC 3261, this RFC does not define any semantic meaning to the status code.

While obviously we should not define user action on receiving status codes, one idea we would like to convey is that any recovery by the user caused by a 494 being sent in response to a Security-Verify header check, specified in 2.3.1 as follows:

   The server can proceed processing a particular request if, and only
   if, the list was not modified.  If modification of the list is
   detected, the server MUST respond to the client with a 494 (Security
   Agreement Required) response.  This response MUST include the
   server's unmodified list of supported security mechanisms.  If the
   list was not modified, and the server is a proxy, it MUST remove the
   "sec-agree" value from both the Require and Proxy-Require header
   fields, and then remove the header fields if no values remain.

should be appropriate to the security mechanism that is being agreed, rather than a matter of sticking the appropriate headers in the next method to be sent. Thus if the security architecture used specifies doing the security agreement over the REGISTER method, then that is the method that must be used for the recovery.

Some extra words in 6.4 to this effect would be useful.

I would suggest something along the lines of:

   "The server received a request which either requires a security agreement
   to be made, or the security agreement indicated did not match that already
   made. In order to continue, the UAC will need to make a new security 
   agreement using requests appropriate to the security mechanism for which
   agreement is made. Any existing dialogs can have been terminated as a 
   result of the loss of the security agreement."

4)	General. The document is inconsistent in its use of the Require and Proxy-Require headers compared to RFC 3261. As a result I believe changes are probably required to both documents.

The current situation in RFC 3261 is that proxies use the Proxy-Require header, and that UAs user the Require header. Neither entity is allowed to remove items from either header.

RFC 3329 currently seems to allow proxies to read and react to contents of the Require header, and both Proxy-Require and Require headers have the sec-agree option-tags removed by a proxy.

What I believe should happen is that RFC 3261 should be modified to specify "d" for proxies in the table that Jonathan dislikes - I mean table 2 and 3 of RFC 3261 for both these headers. Text concerning these headers should be modifed such that where a proxy is reacting and providing capabilities in accordance with the extension/application described in the Proxy-Require header, then the application/extension can define procedures for the deletion of option-tags relating to that extension/application in both the Require and Proxy-Require headers. Unless this is defined in the extension/application, then the proxy must not remove option-tags.

What needs to change in RFC 3329 is that proxy procedures should react only to the Proxy-Require header contents, and UAs should react only to Require header contents, strictly in accordance with RFC 3261. The procedures currently define the modification of Proxy-Require and Require headers for this extension.

regards

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 
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The working group's bugzilla instance has been
temporarily shifted to http://bugs.sipit.net/

I'll send a note when it returns to sipwg.org.

Sorry for the inconvenience,

RjS

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From mailnull@www1.ietf.org  Fri Mar 14 15:05:11 2003
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Date: Fri, 14 Mar 2003 15:02:57 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-mib-05.txt
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I have begun to look at the mib a bit, and I am a bit concerned that 
there is far too much information exposed, in terms of statistics, at 
least. I have to apologize for raising this so late in the game, but it 
is only recently that my day job has brought me into OAMP areas.

One item that struck me as particularly demanding on an implementation 
is the sipTransactionTable. There is a huge amount of work in 
maintaining this table. I really doubt people will want to implement it. 
My suspicion is that this kind of information will normally be logged by 
implementations, and when you need it, its not in real time, but rather 
through some kind of offline analysis of the logs.

Generally, it is my perception that MIB status variables are primarily 
aimed at fault and performance analysis. I dont see how a full list of 
each transaction, with the call-id, cseq and all, supports either aim.

Thanks,
Jonathan R.

Internet-Drafts@ietf.org wrote:
> 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		: Management Information Base for Session Initiation 
>                           Protocol
> 	Author(s)	: K. Lingle, J. Maeng, J. Mule, D. Walker
> 	Filename	: draft-ietf-sip-mib-05.txt
> 	Pages		: 97
> 	Date		: 2003-3-6
> 	
> This memo defines a portion of the Management Information Base (MIB) 
> for use with network management protocols in the Internet community.  
> In particular, it describes a set of managed objects that are used 
> to manage Session Initiation Protocol (SIP) entities, which include 
> User Agents, Proxy servers, Redirect servers and Registrars.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-05.txt
> 
> To remove yourself from the IETF Announcement list, send a message to 
> ietf-announce-request with the word unsubscribe in the body of the message.
> 
> Internet-Drafts are also available by anonymous FTP. Login with the username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
> 	"get draft-ietf-sip-mib-05.txt".
> 
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html 
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
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> In the body type:
> 	"FILE /internet-drafts/draft-ietf-sip-mib-05.txt".
> 	
> NOTE:	The mail server at ietf.org can return the document in
> 	MIME-encoded form by using the "mpack" utility.  To use this
> 	feature, insert the command "ENCODING mime" before the "FILE"
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> 	a MIME-compliant mail reader.  Different MIME-compliant mail readers
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> 		
> 		
> Below is the data which will enable a MIME compliant mail reader
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> Internet-Draft.
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From mailnull@www1.ietf.org  Sun Mar 16 11:18:47 2003
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From: "Shane McCormack" <smccormack@tssg.org>
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Date: Thu, 13 Mar 2003 10:58:24 -0000
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Subject: [Sip] SUBSCRIBE for Geographical Coordinates
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Hi,

Im just wondering how to return geographical coordinates for a sip user
agent. The Scenario would be SIP Location enabled handsets(GPS?).
Would you use an event notification package and if so would the requirements
be simiilar to subscribing to presence events only returning a geographical
coordinate on a specific time interval. Any feedback would be helpful.

Regards
ShaneShane McCormack

Research Assistant,
TSSG, Confederation House,
Waterford Business Park, Cork Road,
Waterford, Co. Waterford.

Phone: +353 51 302927     Fax: +353 51 302901
Mobile: +353 87 9955505
http://www.tssg.org <http://www.tssg.org/>
MSN: mccormackshane@hotmail.com <mailto:mccormackshane@hotmail.com>




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From mailnull@www1.ietf.org  Mon Mar 17 11:22:09 2003
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From: "Bob Penfield" <bpenfield@acmepacket.com>
To: <sip@ietf.org>
Cc: "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>, <jmpolk@cisco.com>
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Subject: [Sip] use of 420 in draft-polk-sip-resource-02.txt
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I have a comment on the use of the 420 response in the resource priority
draft.

In section 4.2, it says the 420 (Extension Required) response is used if the
SIP element does not understand the "Resource-Priority" tag or it does not
understand the namespace or priority value.

I believe this is in conflict with the use of the 417 response described in
section 4.1 and the proper use of 420 as described in RFC 3261.

The 420 response should only be used if it does not understand the feature
tag. The 417 response (as described in section 4.1) should be used when it
does not understand the namespace or priority value. If 420 was only used as
described in 3261, the Accept-Resource-Priority header would not appear in a
420 response.

cheers,
(-:bob

Robert F. Penfield
Chief Software Architect
Acme Packet, Inc.
130 New Boston Street
Woburn, MA 01801
bpenfield@acmepacket.com




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From: murali <muraliv@bharatmail.com>
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Date: Mon Mar 17 18:28:07  2003
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Subject: [Sip] sip grammar
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Hi,

	When i was going through the SIP Document: draft-ietf-sip-call-flows-05.txt
i came across a From Header which is like this

INVITE sip:+1-650-555-2222@ss1.wcom.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ift.here.com:5060
From: sip:+1-303-555-1111@ift.here.com;user=phone
To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>
Call-ID: 1717@ift.here.com
CSeq: 17 INVITE
Contact: <sip:+1-303-555-1111@ift.here.com;user=phone>
Content-Type: application/sdp
Content-Length: 146

v=0
o=IFAXTERMINAL01 2890844527 2890844527 IN IP4 ift.here.com
s=Session SDP
c=IN IP4 iftmg.here.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
(Ref: 3.1.7 Successful SIP to SIP with re-INVITE) 

but Section 20.20 of RFC 3261 says that
"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"

so, i'd like to know if the above usage is correct?
 In case it is correct how do i differentiate between the from parameters and URI-parameters as both are delimited by ;.
I mean to say that i can very well take that(user=phone) as generic parameter in from parameters instead of taking it as uri parameter

Thanks in advance
Murali



-------
Murali Voleti,
#503,Maheswari complex,
Masab Tank,
Hyderabad
Tele:(040) 6502272 ext:211

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Hi,

Not sure if I am sending my mail to the proper mailing list. 

Just a wild thought. Can I think of network of SIP proxies connected in some 
way dynamically exchanging messages in order to assist in forwarding of say, 
INVITES to an appropriate proxy ? Just like OSPF for link state information, 
proxy network for say proxy configuration information exchange ?

Please let me know if I am talking sense.

Anand
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From: "Vishal Verma" <vishal@sonim-india.com>
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Subject: [Sip] query regarding SDP exchange
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Hello friends ,
  I have query regarding SDP exchange during SIP call establishment

let us say user A  calls user B .User A supports audio codec 1 while user B
supports audio codec 2. In this situation SDPs will not match.

My questions are

1. What is the response code to indicate  this failure to originator ?

2. Are there any mandatory audio codecs that must be supported by SIP
devices?


regards, vishal

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jonathan,

thanks for your comments.  please send more if you have them.
my comments inline below...

Jonathan Rosenberg wrote:
> 
> I have begun to look at the mib a bit, and I am a bit concerned that
> there is far too much information exposed, in terms of statistics, at
> least. I have to apologize for raising this so late in the game, but it
> is only recently that my day job has brought me into OAMP areas.

not sure if the above reference to statistics is directly related
to your comment on sipTransactionTable below or not.  if not, are there
other statistics that you feel fall into the category or
exposing far too much information?

> 
> One item that struck me as particularly demanding on an implementation
> is the sipTransactionTable. There is a huge amount of work in
> maintaining this table. I really doubt people will want to implement it.

i had similar feedback from others when support of the sipTransactionTable
was being considered by a project.    the effort involved in supporting
the table was a problem.

we can focus some discussion on the viability of the table during wg last call.  
i'd rather do that than to yank the table w/out further discussion.  
at the very least, we can definitely make the table optional.  
right now it appears to be in a mandatory group - which seem wrong to me anyway.

kevin
> My suspicion is that this kind of information will normally be logged by
> implementations, and when you need it, its not in real time, but rather
> through some kind of offline analysis of the logs.
> 
> Generally, it is my perception that MIB status variables are primarily
> aimed at fault and performance analysis. I dont see how a full list of
> each transaction, with the call-id, cseq and all, supports either aim.
> 
> Thanks,
> Jonathan R.
> 
> Internet-Drafts@ietf.org wrote:
> > 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           : Management Information Base for Session Initiation
> >                           Protocol
> >       Author(s)       : K. Lingle, J. Maeng, J. Mule, D. Walker
> >       Filename        : draft-ietf-sip-mib-05.txt
> >       Pages           : 97
> >       Date            : 2003-3-6
> >
> > This memo defines a portion of the Management Information Base (MIB)
> > for use with network management protocols in the Internet community.
> > In particular, it describes a set of managed objects that are used
> > to manage Session Initiation Protocol (SIP) entities, which include
> > User Agents, Proxy servers, Redirect servers and Registrars.
> >
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-05.txt
> >
> > To remove yourself from the IETF Announcement list, send a message to
> > ietf-announce-request with the word unsubscribe in the body of the message.
> >
> > Internet-Drafts are also available by anonymous FTP. Login with the username
> > "anonymous" and a password of your e-mail address. After logging in,
> > type "cd internet-drafts" and then
> >       "get draft-ietf-sip-mib-05.txt".
> >
> > A list of Internet-Drafts directories can be found in
> > http://www.ietf.org/shadow.html
> > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> >
> > Internet-Drafts can also be obtained by e-mail.
> >
> > Send a message to:
> >       mailserv@ietf.org.
> > In the body type:
> >       "FILE /internet-drafts/draft-ietf-sip-mib-05.txt".
> >
> > NOTE: The mail server at ietf.org can return the document in
> >       MIME-encoded form by using the "mpack" utility.  To use this
> >       feature, insert the command "ENCODING mime" before the "FILE"
> >       command.  To decode the response(s), you will need "munpack" or
> >       a MIME-compliant mail reader.  Different MIME-compliant mail readers
> >       exhibit different behavior, especially when dealing with
> >       "multipart" MIME messages (i.e. documents which have been split
> >       up into multiple messages), so check your local documentation on
> >       how to manipulate these messages.
> >
> >
> > Below is the data which will enable a MIME compliant mail reader
> > implementation to automatically retrieve the ASCII version of the
> > Internet-Draft.
> >
> 
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> 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

-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
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From mailnull@www1.ietf.org  Tue Mar 18 19:58:10 2003
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Subject: [Sip] Re: use of 420 in draft-polk-sip-resource-02.txt
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> In section 4.2, it says the 420 (Extension Required) response is used if the
> SIP element does not understand the "Resource-Priority" tag or it does not
> understand the namespace or priority value.

And if the Require mechanism is in use. (This is not always the case; 
indeed, it is not usually desired.)

> 
> I believe this is in conflict with the use of the 417 response described in
> section 4.1 and the proper use of 420 as described in RFC 3261.

Why? 417 is used if the request fails due to lack of Rewsource-Priority. 
  This is subtly different - it indicates that the priority may not be 
high enough, say.

> 
> The 420 response should only be used if it does not understand the feature
> tag. The 417 response (as described in section 4.1) should be used when it
> does not understand the namespace or priority value. If 420 was only used as
> described in 3261, the Accept-Resource-Priority header would not appear in a
> 420 response.

Why couldn't an Accept-Resource-Priority header appear in a 420? There 
is no restriction on what header fields can appear in a response.

I agree that there may be a need to refactor the error cases, as the 
current description is less than clear. There seem to be the following 
cases:

(0) UAS does *not* know R-P and no Require --> normal SIP behavior, as 
if there was no R-P indication at all (likely, 503)

(1) UAS does *not* know R-P and it is Required --> clearly, 420

(2) UAS does know R-P:

(2s) User insists on 'strict' handling (via Require)

(2s1) UAS does not support namespace or priority --> 417, with Allow-R-P

(2s2) lack of resources --> 417

(2s3) not authorized for this level --> 403

(2L) User allows 'loose' handling (no Require)

(2L1) UAS does not support namespace or priority --> tries to queue as 
if no R-P, might fail as 2L2

(2L2) lack of resources and no queueing for non-priority calls --> 417 
or maybe 503

(2L3) not authorized for this level --> ignores R-P, might fall back to 
2L2 case

Does that cover the cases?

> 
> cheers,
> (-:bob
> 
> Robert F. Penfield
> Chief Software Architect
> Acme Packet, Inc.
> 130 New Boston Street
> Woburn, MA 01801
> bpenfield@acmepacket.com
> 
> 
> 

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From: "Bob Penfield" <bpenfield@acmepacket.com>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>
Cc: <sip@ietf.org>, <jmpolk@cisco.com>
References: <000401c2eca0$e60f2570$ec458182@BPenfield> <3E77BF4B.10203@cs.columbia.edu>
Subject: Re: [Sip] Re: use of 420 in draft-polk-sip-resource-02.txt
Date: Tue, 18 Mar 2003 21:24:22 -0500
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You seem to be supporting my point of view. But just to clarify, I think the
text in section 4.2 which now reads:
                                         Following standard behavior
   (Section 8.2.2.3 of [2]), a UAS MUST then reject the request with
   response code 420 (Bad Extension) if it does not understand the
   mechanism, the namespace or the priority value. If the UAS is capable
   of the resource priority indication in general, but does not
   understand the namespace or priority value, it MUST also include a
   Accept-Resource-Priority header field indicating the namespace-
   priority combinations it can accept.

Should be changed to:
                                         Following standard behavior
   (Section 8.2.2.3 of [2]), a UAS MUST then reject the request with
   response code 420 (Bad Extension) if it does not understand the
   mechanism. If the UAS is capable of the resource priority indication
   in general, but does not understand the namespace or priority
   value, it MUST reject the request with response code 417 and
   also include a Accept-Resource-Priority header field indicating the
   namespace-priority combinations it can accept.

It should not use 420 if the UAS *does* understand the mechanism.

cheers,
(-:bob

Robert F. Penfield
Chief Software Architect
Acme Packet, Inc.
130 New Boston Street
Woburn, MA 01801
bpenfield@acmepacket.com

----- Original Message -----
From: "Henning Schulzrinne" <hgs@cs.columbia.edu>
To: "Bob Penfield" <bpenfield@acmepacket.com>
Cc: <sip@ietf.org>; <jmpolk@cisco.com>
Sent: Tuesday, March 18, 2003 7:52 PM
Subject: [Sip] Re: use of 420 in draft-polk-sip-resource-02.txt


> > In section 4.2, it says the 420 (Extension Required) response is used if
the
> > SIP element does not understand the "Resource-Priority" tag or it does
not
> > understand the namespace or priority value.
>
> And if the Require mechanism is in use. (This is not always the case;
> indeed, it is not usually desired.)
>
> >
> > I believe this is in conflict with the use of the 417 response described
in
> > section 4.1 and the proper use of 420 as described in RFC 3261.
>
> Why? 417 is used if the request fails due to lack of Rewsource-Priority.
>   This is subtly different - it indicates that the priority may not be
> high enough, say.
>
> >
> > The 420 response should only be used if it does not understand the
feature
> > tag. The 417 response (as described in section 4.1) should be used when
it
> > does not understand the namespace or priority value. If 420 was only
used as
> > described in 3261, the Accept-Resource-Priority header would not appear
in a
> > 420 response.
>
> Why couldn't an Accept-Resource-Priority header appear in a 420? There
> is no restriction on what header fields can appear in a response.
>
> I agree that there may be a need to refactor the error cases, as the
> current description is less than clear. There seem to be the following
> cases:
>
> (0) UAS does *not* know R-P and no Require --> normal SIP behavior, as
> if there was no R-P indication at all (likely, 503)
>
> (1) UAS does *not* know R-P and it is Required --> clearly, 420
>
> (2) UAS does know R-P:
>
> (2s) User insists on 'strict' handling (via Require)
>
> (2s1) UAS does not support namespace or priority --> 417, with Allow-R-P
>
> (2s2) lack of resources --> 417
>
> (2s3) not authorized for this level --> 403
>
> (2L) User allows 'loose' handling (no Require)
>
> (2L1) UAS does not support namespace or priority --> tries to queue as
> if no R-P, might fail as 2L2
>
> (2L2) lack of resources and no queueing for non-priority calls --> 417
> or maybe 503
>
> (2L3) not authorized for this level --> ignores R-P, might fall back to
> 2L2 case
>
> Does that cover the cases?
>
> >
> > cheers,
> > (-:bob
> >
> > Robert F. Penfield
> > Chief Software Architect
> > Acme Packet, Inc.
> > 130 New Boston Street
> > Woburn, MA 01801
> > bpenfield@acmepacket.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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From mailnull@www1.ietf.org  Wed Mar 19 08:22:39 2003
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Right now, there is no way to do it, principally because there is no 
standard way to represent geographic information. However, the geopriv 
working group is working on a means for describing geographic location 
information. There was a draft recently on using presence (e.g., 
draft-ietf-simple-presence) to subscribe to such information:

http://www.ietf.org/internet-drafts/draft-peterson-geopriv-pres-00.txt

-Jonathan R.

Shane McCormack wrote:
> Hi,
> 
> Im just wondering how to return geographical coordinates for a sip user
> agent. The Scenario would be SIP Location enabled handsets(GPS?).
> Would you use an event notification package and if so would the requirements
> be simiilar to subscribing to presence events only returning a geographical
> coordinate on a specific time interval. Any feedback would be helpful.
> 
> Regards
> ShaneShane McCormack
> 
> Research Assistant,
> TSSG, Confederation House,
> Waterford Business Park, Cork Road,
> Waterford, Co. Waterford.
> 
> Phone: +353 51 302927     Fax: +353 51 302901
> Mobile: +353 87 9955505
> http://www.tssg.org <http://www.tssg.org/>
> MSN: mccormackshane@hotmail.com <mailto:mccormackshane@hotmail.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
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Kevin Lingle wrote:

>>I have begun to look at the mib a bit, and I am a bit concerned that
>>there is far too much information exposed, in terms of statistics, at
>>least. I have to apologize for raising this so late in the game, but it
>>is only recently that my day job has brought me into OAMP areas.
> 
> 
> not sure if the above reference to statistics is directly related
> to your comment on sipTransactionTable below or not. 

Yes.

> if not, are there
> other statistics that you feel fall into the category or
> exposing far too much information?

I want to look some more and provide some additional comments.

> 
> 
>>One item that struck me as particularly demanding on an implementation
>>is the sipTransactionTable. There is a huge amount of work in
>>maintaining this table. I really doubt people will want to implement it.
> 
> 
> i had similar feedback from others when support of the sipTransactionTable
> was being considered by a project.    the effort involved in supporting
> the table was a problem.
> 
> we can focus some discussion on the viability of the table during wg last call.  

Well, discussion can happen any time. Its happening now, in this email 
thread. I dont know what else you are expecting. I don't think the table 
is viable. You had similar feedback. Seems like a clear cut resolution 
to me.

> i'd rather do that than to yank the table w/out further discussion.  

*This* is the discussion.

> at the very least, we can definitely make the table optional.  
> right now it appears to be in a mandatory group - which seem wrong to me anyway.

Mandatory is definitely wrong. I am not even sure optional is useful.

So - here is the question. What is the bar for putting something in the 
MIB? At one extreme, every piece of state managed by a server could be 
in the mib. At the other extreme, there is nothing, or a single summary 
statistic. I think a reasonable bar, is that there should be clear value 
in monitoring that information, from the perspective of fault and 
performance management in particular. If you cannot demonstrate such a 
use, it should be out.

I cannot think of a reason that I would want, from an operations center, 
to monitor the current set of transactions held in the system. Thus, I 
think it should be out.

Is not our aim to take things out of protocols until we can't take out 
any more, rather than the other way around?

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Hi,

When sending a request (ie REGISTER) to a server I can compare the request 
URI to the common name (or the alt dns name) in the certificate. If the 
names match, I can conclude that the certificate is OK.
(I'm using OpenSSL, and they recommend this post connection assertion).

I have two questions thou:

1 - What name should I use for comparison when accepting a connection?
Usually only the UAC will demand certificate, I am concerned with te case of 
two proxies trying to connect using TLS and the UAS proxy asking for client 
certificates. (what uri will the UAS proxy has, there is no message yet).

2 - how should broken connection be handled? lets say UAC1 sent a request 
over TLS to UAS1. the handshake went well and the request sent. than for 
some reason, the connection was broken and UAS1 now needs to reestablish the 
connection. What should UAS1 do? use TLS w/out certificates?


Regards,
Eron Stein

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Jonathan Rosenberg wrote:
> 
> Kevin Lingle wrote:
> 
> >>I have begun to look at the mib a bit, and I am a bit concerned that
> >>there is far too much information exposed, in terms of statistics, at
> >>least. I have to apologize for raising this so late in the game, but it
> >>is only recently that my day job has brought me into OAMP areas.
> >
> >
> > not sure if the above reference to statistics is directly related
> > to your comment on sipTransactionTable below or not.
> 
> Yes.
> 
> > if not, are there
> > other statistics that you feel fall into the category or
> > exposing far too much information?
> 
> I want to look some more and provide some additional comments.

that would be great.

> 
> >
> >
> >>One item that struck me as particularly demanding on an implementation
> >>is the sipTransactionTable. There is a huge amount of work in
> >>maintaining this table. I really doubt people will want to implement it.
> >
> >
> > i had similar feedback from others when support of the sipTransactionTable
> > was being considered by a project.    the effort involved in supporting
> > the table was a problem.
> >
> > we can focus some discussion on the viability of the table during wg last call.
> 
> Well, discussion can happen any time. Its happening now, in this email
> thread. I dont know what else you are expecting. 

sorry.  didn't mean to imply what came across.  we can definitely discuss
it now.  i'm just not used to actually having active discussion of the mib.
i'm used to getting one or two sporatic comments and only having focused
discussion during last call - when someone is forced to review the mib ;)

> I don't think the table
> is viable. You had similar feedback. Seems like a clear cut resolution
> to me.

not that i don't take your point of view as very significant, i do.  
i was hoping that others might concur before we decide to remove the table.
the table has been in every revision of the mib, so it has been reviewed
multiple times by multiple people... no one - until now has recommended it
come out.   i guess i had taken that as acceptance of it's potential usefulness
to someone.  my case of one project finding it questionable does not necessarily
make a case for it's removal.   we determined that support was feasible but
not all that useful for our platform.  

implementation experience with a mib usually decides such things and we don't 
have much (any?) feedback in that area (not likely to until the mibs are in an
rfc).   still, if we can resolve the question of usefulness now it would be
better than 
to have to revise the mib (ie, rfc) and deprecate objects found - through 
implementation experience - to be useless.

> 
> > i'd rather do that than to yank the table w/out further discussion.
> 
> *This* is the discussion.

agreed.

> 
> > at the very least, we can definitely make the table optional.
> > right now it appears to be in a mandatory group - which seem wrong to me anyway.
> 
> Mandatory is definitely wrong. I am not even sure optional is useful.
> 
> So - here is the question. What is the bar for putting something in the
> MIB? At one extreme, every piece of state managed by a server could be
> in the mib. At the other extreme, there is nothing, or a single summary
> statistic. I think a reasonable bar, is that there should be clear value
> in monitoring that information, from the perspective of fault and
> performance management in particular. If you cannot demonstrate such a
> use, it should be out.
> 
> I cannot think of a reason that I would want, from an operations center,
> to monitor the current set of transactions held in the system. Thus, I
> think it should be out.

it's hard for me to justify or defend the existence of this table as i
was not the original author of it.  but for the sake of discussion...

is there no value from a troubleshooting perspective?   you mentioned that
this information is likely logged for offline analysis - not real-time.
are transactions generally occuring too fast for real-time analysis to
be of worth?   it would seem to me that unless you know specifically which
transactions are of interest to you, it would be hard to use this table
for anything.  the table is defintely not a logging mechanism as transactions
will not persist for any historically significant time.

from a performance mgmt point of view...
could the increase in entries in this table be used to gain insight into
congestion in this or other parts of the sip network?   even if that were
the case, the simple sipCurrentTransactions gauge object would provide
similar insight (to congestion at least).   perhaps the to/from details 
in the table would lend some more insight to problems elsewhere in the
network?   i'm not sure.   

dave walker was the original author of this table, so maybe
he can offer some insight.  dave? 
i did find one old comment related to this table squirreled away in some 
old emails....

Kurt Robohm <kurt@mci.net> wrote:
> 
> sipCurrentTransacations         Does this reflect number of
>                                 calls in progress?
>                                 Intent of objects in table to
> sipTransactionTable             provide real time info on
>                                 number of calls being handled
>                                 real time?

Dave Walker <drwalker@ss8networks.com> responded:
> Transactions aren't calls.  The intent, as mentioned in Adelaide,
> is to evolve the transaction table to a call leg table (probably 
> for UAs only).  This hasn't been resolved yet, so no real changes 
> were made.


i can't find any further discussion by dave (or others) in subsequent 
revisions of the draft to work out any further evolution of this table.  
only minor comments related to the table - none questioning its existence.
dave's involvement in the last couple of revisions of the draft have been nil.
so perhaps the table is an articfact of an idea that was not fully
conceived or fleshed out.  that would lend itself to removal imo.

kevin
> 
> Is not our aim to take things out of protocols until we can't take out
> any more, rather than the other way around?
> 
> -Jonathan R.
> 
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> 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

-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
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Subject: Re: [Sip] Of-hook and On-hook @ sip?
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Well, we have the dialog event package which allows you to find out the 
state of dialogs at a UA:

http://www.ietf.org/internet-drafts/draft-ietf-sipping-dialog-package-01.txt

-Jonathan R.

Vijay K. Gurbani wrote:
> Dean Willis wrote:
> [...]
> 
>> Option 3: We use an event package, and the upstream server subscribes 
>> to the
>> session state of my adapter. When my adapter starts a session, it sends a
>> NOTIFY to the server saying "I have a session, for which I have an
>> identifier.". When my adapter stops a session, it sends anotehr NOIFY 
>> saying
>> "This session (using identifier) has ended".
> 
> 
> One such event package is specified by the SPIRITS protocol.  For
> implementation experience, see:
> http://www.ietf.org/internet-drafts/draft-gurbani-spirits-implementation-00.txt 
> 
> 
> Cheers,
> 
> - vijay

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From: Pekka Pessi <Pekka.Pessi@nokia.com>
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Subject: [Sip] Referred-By
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	Hello all,

	I *think* I volunteered to review referredby, so here I go.

	My first interpretation was that the AIB body should contain
	copies of the (header) fields from REFER request. However, there
	is language in a note within section 4 that suggested otherwise. 
	The suggestion here is that the Referred-By token would be an
	AIB usable by referree.

	I think most of the open issues and confusing stuff in the spec
	is because of this. The draft tries to define how a referrer
	could construct a AIB that a referree could send and target
	could check almost like a AIB, but not quite. Obviously, this is
	pretty hard task.

	Does Referred-By token (RBT) warrant a disposition-type of its
	own?

	I feel that it does - otherwise it is bit hard to authenticate
	origin of REFER request with AIB. Also, referree may include its
	own AIB in the referred request. Validating ordinary AIB or RBT
	are different tasks, so I think it will only create confusion if
	they both use same disposition-type.

	Sooo. Here is my proposal:

	- define a disposition-type for RBT ("referred-by+aib")

	- determine which headers put in the token

	  Referred-By, Refer-To, Date, [Expires], [To]

	  because token is not an ordinary AIB, no Call-ID or CSeq is
	  needed.

	- determine how target validates token

	  - To in incoming request should match Refer-To in RBT
	  - From in incoming request should match To in RBT
	  - Referred-By URL in incoming request should match Referred-By
	    in RBT

	But back to open issue #1. How to determine that the real
	referree is sending the request to target? Include the identity
	in Referred-By header, so that the target could use it in From
	header and authenticate itself using an ordinary AIB? Include
	the From header used by referree in NOTIFY/429 body?

	BR,
					Pekka
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All -

I apologize now for violating the no-crosspost policy, but
this spans all the groups. Please, please, reply _ONLY_ to
sip@ietf.org (which is what should happen if you hit your
reply button if I've built this message correctly).

If you have any opinions, positive or negative, about our
use of bugzilla for issue tracking and whether other groups
should follow the model, now would be a good time to comment.

Thanks!

RjS

-----Forwarded Message-----

> From: john.loughney@nokia.com
> To: rsparks@dynamicsoft.com, wgchairs@ietf.org
> Subject: RE: SIP WG bugzilla instance
> Date: 19 Mar 2003 22:46:52 +0200
> 
> Robert,
> 
> Any comments on how the WG & WG Chairs like it?
> 
> John
> 
> > At the session on improving working group chair training
> > (and chair effectiveness), tools came up. The SIP related
> > working groups have been effectively using a bugzilla variant
> > to track document issues and help drive open issues to
> > conclusion.
> > 
> > It's available at http://www.sipwg.org (or http://bugs.sipit.net).
> > 
> > One point to bring forth at the beginning - we use this as
> > a tracking tool, not an alternative venue for list discussion.
> > Document/issue owners are able to make changes to a bug -
> > people ask the owner to add things to the tracking tool on
> > the appropriate list. The chairs make sure that the owners
> > accurately capture threads on the tracking tool.
> > 
> > RjS
> > 
> > 
> > 

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I have a few related questions/comments:

- A 180 response to a request (like INVITE) establishes a dialog. This 180 response carries record-route headers and therefore sets the route-set. The 200 response to the request might also carry record-route headers. The dialog transitions to confirmed state, but I assume the route-set does not get updated using the record-route headers in the 200 response since the dialog is established already. Is this a correct assumption? should the UAS send record-route headers at all in the 200 if it sent a 1xx already?

Should there be a mandate that 1xx and 2xx response carry the same record-route headers? You would assume that this happens naturally at the UAS, but proxies may play around with the record-route headers.

- In the NOTIFY problem below, the NOTIFY establishes the dialog, but does not carry record-route headers. If the same assumption is made as above, the record-route headers appearing in the 200 response to the SUBSCRIBE are ignored since the dialog is in established state. Isn't this a problem? the route-set is lost.

I appreciate some clarifications.

Regards,
Hisham


> -----Original Message-----
> From: ext hisham.khartabil@nokia.com 
> [mailto:hisham.khartabil@nokia.com]
> Sent: Wednesday, March 12, 2003 5:37 PM
> To: sip@ietf.org
> Subject: [Sip] NOTIFY establishes a dialog
> 
> 
> In RFC3265, it is mentioned that NOTIFY can certainly 
> establish a dialog if it arrives before the 200 of a 
> SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now 
> arrives with record-route headers? Does the route-set get 
> updated? (keeping in mind that the dialog is now in the 
> established state).
> 
> Regards,
> Hisham
> _______________________________________________
> 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
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Robert Sparks wrote:
> All -
> 
> I apologize now for violating the no-crosspost policy, but
> this spans all the groups. Please, please, reply _ONLY_ to
> sip@ietf.org (which is what should happen if you hit your
> reply button if I've built this message correctly).
> 
> If you have any opinions, positive or negative, about our
> use of bugzilla for issue tracking and whether other groups
> should follow the model, now would be a good time to comment.

I think it has helped us immensely in the SIP and SIPPING WGs.
It is nice to be able to go to the bug site and view a list
of open bugs, and furthermore, track them to completion.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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It would be even more helpful if this existed for all active working 
group documents, automatically, not just the 326* documents. We could 
then make sure all open issues are automatically included in the next 
IETF discussion slot.

Vijay K. Gurbani wrote:
> Robert Sparks wrote:
> 
>> All -
>>
>> I apologize now for violating the no-crosspost policy, but
>> this spans all the groups. Please, please, reply _ONLY_ to
>> sip@ietf.org (which is what should happen if you hit your
>> reply button if I've built this message correctly).
>>
>> If you have any opinions, positive or negative, about our
>> use of bugzilla for issue tracking and whether other groups
>> should follow the model, now would be a good time to comment.
> 
> 
> I think it has helped us immensely in the SIP and SIPPING WGs.
> It is nice to be able to go to the bug site and view a list
> of open bugs, and furthermore, track them to completion.
> 
> - vijay

_______________________________________________
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: Re: [Sip] Re: [Simple] [Fwd: RE: SIP WG bugzilla instance]
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: sip@ietf.org
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Hmm - I'll look at that - probably can't _completely_ automate it, but
I might be able to automate prodding the work that it takes to set a 
document up.

In the meantime, I'll remind any WG document owner that all it takes to
get the doc into the database is to send me a peice of mail.

RjS

On Wed, 2003-03-19 at 16:56, Henning Schulzrinne wrote:
> It would be even more helpful if this existed for all active working 
> group documents, automatically, not just the 326* documents. We could 
> then make sure all open issues are automatically included in the next 
> IETF discussion slot.
> 
> Vijay K. Gurbani wrote:
> > Robert Sparks wrote:
> > 
> >> All -
> >>
> >> I apologize now for violating the no-crosspost policy, but
> >> this spans all the groups. Please, please, reply _ONLY_ to
> >> sip@ietf.org (which is what should happen if you hit your
> >> reply button if I've built this message correctly).
> >>
> >> If you have any opinions, positive or negative, about our
> >> use of bugzilla for issue tracking and whether other groups
> >> should follow the model, now would be a good time to comment.
> > 
> > 
> > I think it has helped us immensely in the SIP and SIPPING WGs.
> > It is nice to be able to go to the bug site and view a list
> > of open bugs, and furthermore, track them to completion.
> > 
> > - vijay
> 
> _______________________________________________
> 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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From: "Arunachalam Venkatraman" <arunvenk@cisco.com>
To: <hisham.khartabil@nokia.com>, <sip@ietf.org>
Subject: RE: [Sip] NOTIFY establishes a dialog
Date: Wed, 19 Mar 2003 19:47:09 -0600
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Hisham
The reason the Record_Route headers must be sent on the 200 is that the 1xx
is not guaranteed to be delivered, unless they are sent reliably.

I think in the context of the Notify crossing the 200, the arrival of the
200 will establish the Route set for future requests. Each transaction
completes independently.

I am copying in a old message on this subject from the sip list below that
you may find useful - (I think it would have been useful to state this in
the RFC3261).

------------------ Start Of Old Message --------------------------

>  -----Original Message-----
> From: 	Arunachalam Venkatraman [mailto:arunvenk@cisco.com]
> Sent:	Friday, March 23, 2001 3:39 PM
> To:	sip-implementors@cs.columbia.edu
> Cc:	Jonathan Rosenberg
> Subject:	Replace Contact in Route set ?
>
> As per the protocol, the Contact header is added to the end of the Route
header.
>
> Suppose a Record-Route was first received in a message with a Contact C1.
> The UA has built a route set with the Contact C1 at the bottom of the
Route.
>
> If the next request to the UA has a new Contact C2, should C1 in the Route
Set be replaced
> with C2?
>

Yes. And if there was no Contact, don't change anything.

Note that if the next request has new record-routes, those do NOT update the
existing route sets.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

---------------------------------End of Old Message ----------------------

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
hisham.khartabil@nokia.com
Sent: Wednesday, March 19, 2003 4:48 PM
To: sip@ietf.org
Subject: RE: [Sip] NOTIFY establishes a dialog


I have a few related questions/comments:

- A 180 response to a request (like INVITE) establishes a dialog. This 180
response carries record-route headers and therefore sets the route-set. The
200 response to the request might also carry record-route headers. The
dialog transitions to confirmed state, but I assume the route-set does not
get updated using the record-route headers in the 200 response since the
dialog is established already. Is this a correct assumption? should the UAS
send record-route headers at all in the 200 if it sent a 1xx already?

Should there be a mandate that 1xx and 2xx response carry the same
record-route headers? You would assume that this happens naturally at the
UAS, but proxies may play around with the record-route headers.

- In the NOTIFY problem below, the NOTIFY establishes the dialog, but does
not carry record-route headers. If the same assumption is made as above, the
record-route headers appearing in the 200 response to the SUBSCRIBE are
ignored since the dialog is in established state. Isn't this a problem? the
route-set is lost.

I appreciate some clarifications.

Regards,
Hisham


> -----Original Message-----
> From: ext hisham.khartabil@nokia.com
> [mailto:hisham.khartabil@nokia.com]
> Sent: Wednesday, March 12, 2003 5:37 PM
> To: sip@ietf.org
> Subject: [Sip] NOTIFY establishes a dialog
>
>
> In RFC3265, it is mentioned that NOTIFY can certainly
> establish a dialog if it arrives before the 200 of a
> SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now
> arrives with record-route headers? Does the route-set get
> updated? (keeping in mind that the dialog is now in the
> established state).
>
> Regards,
> Hisham
> _______________________________________________
> 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

_______________________________________________
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: RE: [Sip] NOTIFY establishes a dialog
Date: Thu, 20 Mar 2003 05:59:05 +0200
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Thread-Topic: [Sip] NOTIFY establishes a dialog
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I don't think you have accurately my questions.

The spec says once a dialog is established, you cannot change the route-set. A 180 establishes a dialog, so my assumption is that the 200 ok record-route headers are ignored.

The same goes for NOTIFY (by the way, 200 for SUBSCRIBE in going in the same direction as NOTIFY, not crossing on the wire). NOTIFY establishes a dialog and therefore the route-set delivered in the NOTIFY is lost. Your conclusions below break the rule where a route-set cannot be changed once a dialog has been established.

/Hisham

> -----Original Message-----
> From: ext Arunachalam Venkatraman [mailto:arunvenk@cisco.com]
> Sent: Thursday, March 20, 2003 3:47 AM
> To: Khartabil Hisham (NMP/Helsinki); sip@ietf.org
> Subject: RE: [Sip] NOTIFY establishes a dialog
> 
> 
> Hisham
> The reason the Record_Route headers must be sent on the 200 
> is that the 1xx
> is not guaranteed to be delivered, unless they are sent reliably.
> 
> I think in the context of the Notify crossing the 200, the 
> arrival of the
> 200 will establish the Route set for future requests. Each transaction
> completes independently.
> 
> I am copying in a old message on this subject from the sip 
> list below that
> you may find useful - (I think it would have been useful to 
> state this in
> the RFC3261).
> 
> ------------------ Start Of Old Message --------------------------
> 
> >  -----Original Message-----
> > From: 	Arunachalam Venkatraman [mailto:arunvenk@cisco.com]
> > Sent:	Friday, March 23, 2001 3:39 PM
> > To:	sip-implementors@cs.columbia.edu
> > Cc:	Jonathan Rosenberg
> > Subject:	Replace Contact in Route set ?
> >
> > As per the protocol, the Contact header is added to the end 
> of the Route
> header.
> >
> > Suppose a Record-Route was first received in a message with 
> a Contact C1.
> > The UA has built a route set with the Contact C1 at the 
> bottom of the
> Route.
> >
> > If the next request to the UA has a new Contact C2, should 
> C1 in the Route
> Set be replaced
> > with C2?
> >
> 
> Yes. And if there was no Contact, don't change anything.
> 
> Note that if the next request has new record-routes, those do 
> NOT update the
> existing route sets.
> 
> -Jonathan R.
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> ---------------------------------End of Old Message 
> ----------------------
> 
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> hisham.khartabil@nokia.com
> Sent: Wednesday, March 19, 2003 4:48 PM
> To: sip@ietf.org
> Subject: RE: [Sip] NOTIFY establishes a dialog
> 
> 
> I have a few related questions/comments:
> 
> - A 180 response to a request (like INVITE) establishes a 
> dialog. This 180
> response carries record-route headers and therefore sets the 
> route-set. The
> 200 response to the request might also carry record-route headers. The
> dialog transitions to confirmed state, but I assume the 
> route-set does not
> get updated using the record-route headers in the 200 
> response since the
> dialog is established already. Is this a correct assumption? 
> should the UAS
> send record-route headers at all in the 200 if it sent a 1xx already?
> 
> Should there be a mandate that 1xx and 2xx response carry the same
> record-route headers? You would assume that this happens 
> naturally at the
> UAS, but proxies may play around with the record-route headers.
> 
> - In the NOTIFY problem below, the NOTIFY establishes the 
> dialog, but does
> not carry record-route headers. If the same assumption is 
> made as above, the
> record-route headers appearing in the 200 response to the 
> SUBSCRIBE are
> ignored since the dialog is in established state. Isn't this 
> a problem? the
> route-set is lost.
> 
> I appreciate some clarifications.
> 
> Regards,
> Hisham
> 
> 
> > -----Original Message-----
> > From: ext hisham.khartabil@nokia.com
> > [mailto:hisham.khartabil@nokia.com]
> > Sent: Wednesday, March 12, 2003 5:37 PM
> > To: sip@ietf.org
> > Subject: [Sip] NOTIFY establishes a dialog
> >
> >
> > In RFC3265, it is mentioned that NOTIFY can certainly
> > establish a dialog if it arrives before the 200 of a
> > SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now
> > arrives with record-route headers? Does the route-set get
> > updated? (keeping in mind that the dialog is now in the
> > established state).
> >
> > Regards,
> > Hisham
> > _______________________________________________
> > 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 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: Re: Ri: [Sip] SIP proxy network
To: loretosa@vodafone.it
Date: Wed, 19 Mar 2003 16:03:44 +0530 (GMT+05:30)
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In-Reply-To: <hbxxxp$ICbMEHMSvkVHNKMHR1rJ9CO31qCRd_cm9@vodafone.it> from "loretosa@vodafone.it" at Mar 18, 2003 11:46:37 AM
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The kind of information the proxies exchange should enable the UA or local proxy
to dynamically locate and choose where to forward say,INVITE to. Since the 
information contained within proxies is at the application level I am thinking if
an application level and perhaps text based routing protocol is necessary.


> 
> I'm not sure about your question,
> are you asking about the possibility for UA or an
> inBound Proxy to choose the best way where send
> a message (like INVITE)?
> 
> 
> > Hi,
> > 
> > Not sure if I am sending my mail to the proper mailing list. 
> > 
> > Just a wild thought. Can I think of network of SIP proxies connected in some 
> > way dynamically exchanging messages in order to assist in forwarding of say, 
> > INVITES to an appropriate proxy ? Just like OSPF for link state information, 
> > proxy network for say proxy configuration information exchange ?
> > 
> > Please let me know if I am talking sense.
> > 
> > Anand
> > _______________________________________________
> > 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
> > 
> 
>  Per registrarti gratuitamente a Vodafone Mail vai su www.190.it 
> 
> 

_______________________________________________
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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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Subject: RE: [Sip] NOTIFY establishes a dialog
From: Ganesh Sivaraman <ganesh.sivaraman@wipro.com>
To: ganesh.sivaraman@wipro.com
Cc: sip@ietf.org
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Hi Hisham,
 I'd posted the same question about setting dialog-state (route-set
etc.) and I got the following response:
<http://lists.cs.columbia.edu/pipermail/sip-implementors/2003-February/004569.html>

It says:
"RFC 3265 augments 3261, in that the dialog in this case can be created
either by a response to SUBSCRIBE or by NOTIFY".

So that would mean that the route-set could be constructed with
Route-headers (present in NOTIFY) also. Is this a correct assumption?

A word from the authors of the RFC on this would help bring more
clarity.

Regards
Ganesh

On Thu, 2003-03-20 at 04:17, hisham.khartabil@nokia.com wrote:
> I have a few related questions/comments:
> 
> - A 180 response to a request (like INVITE) establishes a dialog. This 180 response carries record-route headers and therefore sets the route-set. The 200 response to the request might also carry record-route headers. The dialog transitions to confirmed state, but I assume the route-set does not get updated using the record-route headers in the 200 response since the dialog is established already. Is this a correct assumption? should the UAS send record-route headers at all in the 200 if it sent a 1xx already?
> 
> Should there be a mandate that 1xx and 2xx response carry the same record-route headers? You would assume that this happens naturally at the UAS, but proxies may play around with the record-route headers.
> 
> - In the NOTIFY problem below, the NOTIFY establishes the dialog, but does not carry record-route headers. If the same assumption is made as above, the record-route headers appearing in the 200 response to the SUBSCRIBE are ignored since the dialog is in established state. Isn't this a problem? the route-set is lost.
> 
> I appreciate some clarifications.
> 
> Regards,
> Hisham
> 
> 
> > -----Original Message-----
> > From: ext hisham.khartabil@nokia.com 
> > [mailto:hisham.khartabil@nokia.com]
> > Sent: Wednesday, March 12, 2003 5:37 PM
> > To: sip@ietf.org
> > Subject: [Sip] NOTIFY establishes a dialog
> > 
> > 
> > In RFC3265, it is mentioned that NOTIFY can certainly 
> > establish a dialog if it arrives before the 200 of a 
> > SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now 
> > arrives with record-route headers? Does the route-set get 
> > updated? (keeping in mind that the dialog is now in the 
> > established state).
> > 
> > Regards,
> > Hisham
> > _______________________________________________
> > 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

_______________________________________________
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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We too have sip-mib implementation in our product.
Following is my view with respect to sipTransactionTable:

1. Maintaining this table info in the system is not
   'simple and straight forward'.
   Reason: Data involved are very volatile.
   Moreover it requires indexing the transactions which
   themselves are volatile (destroyed any time).

2. This information is usually obtained (if et-all obtained!)
   will be via GET-BULK operation. To support GET-BULK to
   retrieve information specific to all transactions
   has momentory performance impact on the Server.
   There are issues of thread-safety between GET-BULK
   and the transaction handling operation.

3. Same information can be obtained either through
   protocol trace logs or CDR (call detail records).

I feel sip-mib should not have tables involving volatile data.

I also feel sipCurrentTransTable is not useful.
May be TRAP objects based on sipCurrentTransTable, will be
more useful to SNMP admin.

I think sip-mib has captured very elaborately on following things.
1. Server Administration/Configuration
2. User data management
3. Statistics
I really appreciate the above factor.

I have some comments on following MIB fields, please
clarify them as my understanding may differ from the
auther's of sip-mib.

1. "sipMaxSessions" of SipCommonCfgEntry:
   This is a READ-ONLY property supplied by the
   SIP-ENTITY. As per the definition the entity
   has to determine what is the maximum load it
   can handle.

   Again, the above factor depends on the
   hardware/software configuration of the deployment
   platform. Do we expect for instance a SIP-Proxy
   to run some algorithm/load itself to find out
   this value?

   I feel this field may not be useful.

2. "sipTransportSnd" of SipPortEntry: I donot
   understand how this field is utilized in SIP,
   specific to a port number.

   "sipTransportRcv" can be understood as for
   associated port number the SIP entity will
   receive message for TRANSPORTS that are
   enabled in this field.

3. "sipPgpVersion" of SipServerCfgEntry,
   "sipPgpPrivateKey" of SipProxyCfgEntry:
   PGP is deprecated in SIP. Can these be removed?
   There may be more things specific to PGP.

4. "sipServerRespectUAAction" of SipServerCfgEntry:
   This mentions action parameter specific operation
   in the proxy which is deprecated in SIP.
   Can we deprecate here also?

5. "sipRegMaxUsers" of "SipRegCfgEntry":
   This is a READ-ONLY field which is supposed
   to be given out by REGISTRAR as what is the
   number of users does it support.

   Again, what is the criteria with which registrar
   arrives at this data. This data again depends on
   deployment platform such as capacity of the
   database etc. This data too may not be useful
   to administrator. This value is either known
   to the admin or enforced by admin on REGISTRAR.
   Therefore keeping it as READ-ONLY has no purpose.

6. "sipContactAction" of SipContactEntry:
   Again deprecated by SIP. Can we deprecate here too?

7. "sipContactRetryAfter" of SipContactEntry:
   Again deprecated by SIP. Can we deprecate here too?

8. Transaction timer specific configuration are based
   on pre-bis05 drafts. They have to be updated
   according to latest RFC. I think action-item is already
   initiated for post mib-05 drafts.

Thanks and Regards,
Shrinivas Shetti





Kevin Lingle <klingle@cisco.com> on 03/19/2003 09:10:17 PM

To:   Jonathan Rosenberg <jdrosen@dynamicsoft.com>
cc:   sip@ietf.org, Dave Walker <drwalker@ss8networks.com> (bcc: Shrinivas
      Shetti/HSSBLR)

Subject:  Re: [Sip] I-D ACTION:draft-ietf-sip-mib-05.txt




Jonathan Rosenberg wrote:
>
> Kevin Lingle wrote:
>
> >>I have begun to look at the mib a bit, and I am a bit concerned that
> >>there is far too much information exposed, in terms of statistics, at
> >>least. I have to apologize for raising this so late in the game, but it
> >>is only recently that my day job has brought me into OAMP areas.
> >
> >
> > not sure if the above reference to statistics is directly related
> > to your comment on sipTransactionTable below or not.
>
> Yes.
>
> > if not, are there
> > other statistics that you feel fall into the category or
> > exposing far too much information?
>
> I want to look some more and provide some additional comments.

that would be great.

>
> >
> >
> >>One item that struck me as particularly demanding on an implementation
> >>is the sipTransactionTable. There is a huge amount of work in
> >>maintaining this table. I really doubt people will want to implement
it.
> >
> >
> > i had similar feedback from others when support of the
sipTransactionTable
> > was being considered by a project.    the effort involved in supporting
> > the table was a problem.
> >
> > we can focus some discussion on the viability of the table during wg
last call.
>
> Well, discussion can happen any time. Its happening now, in this email
> thread. I dont know what else you are expecting.

sorry.  didn't mean to imply what came across.  we can definitely discuss
it now.  i'm just not used to actually having active discussion of the mib.
i'm used to getting one or two sporatic comments and only having focused
discussion during last call - when someone is forced to review the mib ;)

> I don't think the table
> is viable. You had similar feedback. Seems like a clear cut resolution
> to me.

not that i don't take your point of view as very significant, i do.
i was hoping that others might concur before we decide to remove the table.
the table has been in every revision of the mib, so it has been reviewed
multiple times by multiple people... no one - until now has recommended it
come out.   i guess i had taken that as acceptance of it's potential
usefulness
to someone.  my case of one project finding it questionable does not
necessarily
make a case for it's removal.   we determined that support was feasible but
not all that useful for our platform.

implementation experience with a mib usually decides such things and we
don't
have much (any?) feedback in that area (not likely to until the mibs are in
an
rfc).   still, if we can resolve the question of usefulness now it would be
better than
to have to revise the mib (ie, rfc) and deprecate objects found - through
implementation experience - to be useless.

>
> > i'd rather do that than to yank the table w/out further discussion.
>
> *This* is the discussion.

agreed.

>
> > at the very least, we can definitely make the table optional.
> > right now it appears to be in a mandatory group - which seem wrong to
me anyway.
>
> Mandatory is definitely wrong. I am not even sure optional is useful.
>
> So - here is the question. What is the bar for putting something in the
> MIB? At one extreme, every piece of state managed by a server could be
> in the mib. At the other extreme, there is nothing, or a single summary
> statistic. I think a reasonable bar, is that there should be clear value
> in monitoring that information, from the perspective of fault and
> performance management in particular. If you cannot demonstrate such a
> use, it should be out.
>
> I cannot think of a reason that I would want, from an operations center,
> to monitor the current set of transactions held in the system. Thus, I
> think it should be out.

it's hard for me to justify or defend the existence of this table as i
was not the original author of it.  but for the sake of discussion...

is there no value from a troubleshooting perspective?   you mentioned that
this information is likely logged for offline analysis - not real-time.
are transactions generally occuring too fast for real-time analysis to
be of worth?   it would seem to me that unless you know specifically which
transactions are of interest to you, it would be hard to use this table
for anything.  the table is defintely not a logging mechanism as
transactions
will not persist for any historically significant time.

from a performance mgmt point of view...
could the increase in entries in this table be used to gain insight into
congestion in this or other parts of the sip network?   even if that were
the case, the simple sipCurrentTransactions gauge object would provide
similar insight (to congestion at least).   perhaps the to/from details
in the table would lend some more insight to problems elsewhere in the
network?   i'm not sure.

dave walker was the original author of this table, so maybe
he can offer some insight.  dave?
i did find one old comment related to this table squirreled away in some
old emails....

Kurt Robohm <kurt@mci.net> wrote:
>
> sipCurrentTransacations         Does this reflect number of
>                                 calls in progress?
>                                 Intent of objects in table to
> sipTransactionTable             provide real time info on
>                                 number of calls being handled
>                                 real time?

Dave Walker <drwalker@ss8networks.com> responded:
> Transactions aren't calls.  The intent, as mentioned in Adelaide,
> is to evolve the transaction table to a call leg table (probably
> for UAs only).  This hasn't been resolved yet, so no real changes
> were made.


i can't find any further discussion by dave (or others) in subsequent
revisions of the draft to work out any further evolution of this table.
only minor comments related to the table - none questioning its existence.
dave's involvement in the last couple of revisions of the draft have been
nil.
so perhaps the table is an articfact of an idea that was not fully
conceived or fleshed out.  that would lend itself to removal imo.

kevin
>
> Is not our aim to take things out of protocols until we can't take out
> any more, rather than the other way around?
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> 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

--
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
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This list is for NEW development of the core SIP Protocol
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Subject: RE: [Sip] NOTIFY establishes a dialog
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Ganesh Sivaraman <ganesh.sivaraman@wipro.com>, hisham.khartabil@nokia.com
Cc: sip@ietf.org
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response to one of the questions inline:

On Wed, 2003-03-19 at 20:55, Ganesh Sivaraman wrote:
> Hi Hisham,
>  I'd posted the same question about setting dialog-state (route-set
> etc.) and I got the following response:
> <http://lists.cs.columbia.edu/pipermail/sip-implementors/2003-February/004569.html>
> 
> It says:
> "RFC 3265 augments 3261, in that the dialog in this case can be created
> either by a response to SUBSCRIBE or by NOTIFY".
> 
> So that would mean that the route-set could be constructed with
> Route-headers (present in NOTIFY) also. Is this a correct assumption?
> 
> A word from the authors of the RFC on this would help bring more
> clarity.
> 
> Regards
> Ganesh
> 
> On Thu, 2003-03-20 at 04:17, hisham.khartabil@nokia.com wrote:
> > I have a few related questions/comments:
> > 
> > - A 180 response to a request (like INVITE) establishes a dialog. This 180 response carries record-route headers and therefore sets the route-set. The 200 response to the request might also carry record-route headers. The dialog transitions to confirmed state, but I assume the route-set does not get updated using the record-route headers in the 200 response since the dialog is established already. Is this a correct assumption? should the UAS send record-route headers at all in the 200 if it sent a 1xx already?

>From rfc3261 page 82:

   If the dialog identifier in the 2xx response matches the dialog
   identifier of an existing dialog, the dialog MUST be transitioned to
   the "confirmed" state, and the route set for the dialog MUST be
   recomputed based on the 2xx response using the procedures of Section
   12.2.1.2.  Otherwise, a new dialog in the "confirmed" state MUST be
   constructed using the procedures of Section 12.1.2.

In context, you'll see the "existing dialog" above is an "early dialog".

> > 
> > Should there be a mandate that 1xx and 2xx response carry the same record-route headers? You would assume that this happens naturally at the UAS, but proxies may play around with the record-route headers.
> > 
> > - In the NOTIFY problem below, the NOTIFY establishes the dialog, but does not carry record-route headers. If the same assumption is made as above, the record-route headers appearing in the 200 response to the SUBSCRIBE are ignored since the dialog is in established state. Isn't this a problem? the route-set is lost.
> > 
> > I appreciate some clarifications.
> > 
> > Regards,
> > Hisham
> > 
> > 
> > > -----Original Message-----
> > > From: ext hisham.khartabil@nokia.com 
> > > [mailto:hisham.khartabil@nokia.com]
> > > Sent: Wednesday, March 12, 2003 5:37 PM
> > > To: sip@ietf.org
> > > Subject: [Sip] NOTIFY establishes a dialog
> > > 
> > > 
> > > In RFC3265, it is mentioned that NOTIFY can certainly 
> > > establish a dialog if it arrives before the 200 of a 
> > > SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now 
> > > arrives with record-route headers? Does the route-set get 
> > > updated? (keeping in mind that the dialog is now in the 
> > > established state).
> > > 
> > > Regards,
> > > Hisham
> > > _______________________________________________
> > > 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
> 
> _______________________________________________
> 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: "'hisham.khartabil@nokia.com'" <hisham.khartabil@nokia.com>, sip@ietf.org
Subject: RE: [Sip] NOTIFY establishes a dialog
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> From: hisham.khartabil@nokia.com [mailto:hisham.khartabil@nokia.com]
> 
> In RFC3265, it is mentioned that NOTIFY can certainly 
> establish a dialog if it arrives before the 200 of a 
> SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now 
> arrives with record-route headers? Does the route-set get 
> updated? (keeping in mind that the dialog is now in the 
> established state).

Not if it's part of the same dialog as the aforementioned NOTIFY.

In the case that the SUBSCRIBE 200 doesn't match any dialog that's
already been established (e.g. by a NOTIFY), then it *does* establish
a dialog. If the event package doesn't allow forking, such a dialog
conceptually terminates after the the subscriber 481s the corresponding
NOTIFY.

> the NOTIFY establishes the dialog, but does not carry record-route
> headers. If the same assumption is made as above, the record-route
> headers appearing in the 200 response to the SUBSCRIBE are ignored
> since the dialog is in established state. Isn't this a problem? the
> route-set is lost.

Section 3.1.5 of 3265 clearly states:

   If a proxy wishes to see all of the SUBSCRIBE
   and NOTIFY requests for a given dialog, it MUST record-route the
   initial SUBSCRIBE and any dialog-establishing NOTIFY requests. Such
   proxies SHOULD also record-route all other SUBSCRIBE and NOTIFY
   requests.

A consequence of this clause is that a proxy which Record-Routes the
200 response will also Record-Route the NOTIFY in the same dialog.
A further consequence is that the route set in a SUBSCRIBE 200 and
that of a NOTIFY in the same dialog will always be identical.

/a
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Subject: RE: [Sip] NOTIFY establishes a dialog
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This sounds like a solution to the first problem. I guess I couldn't find the text because it was in the wrong section. May I suggest a bug report specifying moving the text to section 12.

The other problem with NOTIFY can be solved by specifying that a NOTIFY establishes an early dialog. Is that ok? bug report?

Regards,
Hisham

> -----Original Message-----
> From: ext Robert Sparks [mailto:rsparks@dynamicsoft.com]
> Sent: Thursday, March 20, 2003 7:55 AM
> To: Ganesh Sivaraman; Khartabil Hisham (NMP/Helsinki)
> Cc: sip@ietf.org
> Subject: RE: [Sip] NOTIFY establishes a dialog
> 
> 
> response to one of the questions inline:
> 
> On Wed, 2003-03-19 at 20:55, Ganesh Sivaraman wrote:
> > Hi Hisham,
> >  I'd posted the same question about setting dialog-state (route-set
> > etc.) and I got the following response:
> > 
> <http://lists.cs.columbia.edu/pipermail/sip-implementors/2003-
> February/004569.html>
> > 
> > It says:
> > "RFC 3265 augments 3261, in that the dialog in this case 
> can be created
> > either by a response to SUBSCRIBE or by NOTIFY".
> > 
> > So that would mean that the route-set could be constructed with
> > Route-headers (present in NOTIFY) also. Is this a correct 
> assumption?
> > 
> > A word from the authors of the RFC on this would help bring more
> > clarity.
> > 
> > Regards
> > Ganesh
> > 
> > On Thu, 2003-03-20 at 04:17, hisham.khartabil@nokia.com wrote:
> > > I have a few related questions/comments:
> > > 
> > > - A 180 response to a request (like INVITE) establishes a 
> dialog. This 180 response carries record-route headers and 
> therefore sets the route-set. The 200 response to the request 
> might also carry record-route headers. The dialog transitions 
> to confirmed state, but I assume the route-set does not get 
> updated using the record-route headers in the 200 response 
> since the dialog is established already. Is this a correct 
> assumption? should the UAS send record-route headers at all 
> in the 200 if it sent a 1xx already?
> 
> >From rfc3261 page 82:
> 
>    If the dialog identifier in the 2xx response matches the dialog
>    identifier of an existing dialog, the dialog MUST be 
> transitioned to
>    the "confirmed" state, and the route set for the dialog MUST be
>    recomputed based on the 2xx response using the procedures 
> of Section
>    12.2.1.2.  Otherwise, a new dialog in the "confirmed" state MUST be
>    constructed using the procedures of Section 12.1.2.
> 
> In context, you'll see the "existing dialog" above is an 
> "early dialog".
> 
> > > 
> > > Should there be a mandate that 1xx and 2xx response carry 
> the same record-route headers? You would assume that this 
> happens naturally at the UAS, but proxies may play around 
> with the record-route headers.
> > > 
> > > - In the NOTIFY problem below, the NOTIFY establishes the 
> dialog, but does not carry record-route headers. If the same 
> assumption is made as above, the record-route headers 
> appearing in the 200 response to the SUBSCRIBE are ignored 
> since the dialog is in established state. Isn't this a 
> problem? the route-set is lost.
> > > 
> > > I appreciate some clarifications.
> > > 
> > > Regards,
> > > Hisham
> > > 
> > > 
> > > > -----Original Message-----
> > > > From: ext hisham.khartabil@nokia.com 
> > > > [mailto:hisham.khartabil@nokia.com]
> > > > Sent: Wednesday, March 12, 2003 5:37 PM
> > > > To: sip@ietf.org
> > > > Subject: [Sip] NOTIFY establishes a dialog
> > > > 
> > > > 
> > > > In RFC3265, it is mentioned that NOTIFY can certainly 
> > > > establish a dialog if it arrives before the 200 of a 
> > > > SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now 
> > > > arrives with record-route headers? Does the route-set get 
> > > > updated? (keeping in mind that the dialog is now in the 
> > > > established state).
> > > > 
> > > > Regards,
> > > > Hisham
> > > > _______________________________________________
> > > > 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
> > 
> > _______________________________________________
> > 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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Subject: RE: [Sip] NOTIFY establishes a dialog
From: Robert Sparks <rsparks@dynamicsoft.com>
To: hisham.khartabil@nokia.com
Cc: ganesh.sivaraman@wipro.com, sip@ietf.org
In-Reply-To: <2038BCC78B1AD641891A0D1AE133DBB7FE73F0@esebe019.ntc.nokia.com>
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A NOTIFY that beats a 200 OK establishes a full-blown established
dialog, not an early one.

If the 200 dialog identifiers match the NOTIFY, the route set will not
be updated. If the 200 identifies some other dialog, then the dialog
established by the NOTIFY continues to exist - state unchanged.

This is covered in 3.3.4 of 3265 - the first of a NOTIFY or a 200
with a fully qualified dialog identifier establishes a dialog (and
the corresponding route set). Please read that section again - if
you still think it's not clear, propose text.

RjS

On Wed, 2003-03-19 at 22:19, hisham.khartabil@nokia.com wrote:
> This sounds like a solution to the first problem. I guess I couldn't find the text because it was in the wrong section. May I suggest a bug report specifying moving the text to section 12.
> 
> The other problem with NOTIFY can be solved by specifying that a NOTIFY establishes an early dialog. Is that ok? bug report?
> 
> Regards,
> Hisham
> 
> > -----Original Message-----
> > From: ext Robert Sparks [mailto:rsparks@dynamicsoft.com]
> > Sent: Thursday, March 20, 2003 7:55 AM
> > To: Ganesh Sivaraman; Khartabil Hisham (NMP/Helsinki)
> > Cc: sip@ietf.org
> > Subject: RE: [Sip] NOTIFY establishes a dialog
> > 
> > 
> > response to one of the questions inline:
> > 
> > On Wed, 2003-03-19 at 20:55, Ganesh Sivaraman wrote:
> > > Hi Hisham,
> > >  I'd posted the same question about setting dialog-state (route-set
> > > etc.) and I got the following response:
> > > 
> > <http://lists.cs.columbia.edu/pipermail/sip-implementors/2003-
> > February/004569.html>
> > > 
> > > It says:
> > > "RFC 3265 augments 3261, in that the dialog in this case 
> > can be created
> > > either by a response to SUBSCRIBE or by NOTIFY".
> > > 
> > > So that would mean that the route-set could be constructed with
> > > Route-headers (present in NOTIFY) also. Is this a correct 
> > assumption?
> > > 
> > > A word from the authors of the RFC on this would help bring more
> > > clarity.
> > > 
> > > Regards
> > > Ganesh
> > > 
> > > On Thu, 2003-03-20 at 04:17, hisham.khartabil@nokia.com wrote:
> > > > I have a few related questions/comments:
> > > > 
> > > > - A 180 response to a request (like INVITE) establishes a 
> > dialog. This 180 response carries record-route headers and 
> > therefore sets the route-set. The 200 response to the request 
> > might also carry record-route headers. The dialog transitions 
> > to confirmed state, but I assume the route-set does not get 
> > updated using the record-route headers in the 200 response 
> > since the dialog is established already. Is this a correct 
> > assumption? should the UAS send record-route headers at all 
> > in the 200 if it sent a 1xx already?
> > 
> > >From rfc3261 page 82:
> > 
> >    If the dialog identifier in the 2xx response matches the dialog
> >    identifier of an existing dialog, the dialog MUST be 
> > transitioned to
> >    the "confirmed" state, and the route set for the dialog MUST be
> >    recomputed based on the 2xx response using the procedures 
> > of Section
> >    12.2.1.2.  Otherwise, a new dialog in the "confirmed" state MUST be
> >    constructed using the procedures of Section 12.1.2.
> > 
> > In context, you'll see the "existing dialog" above is an 
> > "early dialog".
> > 
> > > > 
> > > > Should there be a mandate that 1xx and 2xx response carry 
> > the same record-route headers? You would assume that this 
> > happens naturally at the UAS, but proxies may play around 
> > with the record-route headers.
> > > > 
> > > > - In the NOTIFY problem below, the NOTIFY establishes the 
> > dialog, but does not carry record-route headers. If the same 
> > assumption is made as above, the record-route headers 
> > appearing in the 200 response to the SUBSCRIBE are ignored 
> > since the dialog is in established state. Isn't this a 
> > problem? the route-set is lost.
> > > > 
> > > > I appreciate some clarifications.
> > > > 
> > > > Regards,
> > > > Hisham
> > > > 
> > > > 
> > > > > -----Original Message-----
> > > > > From: ext hisham.khartabil@nokia.com 
> > > > > [mailto:hisham.khartabil@nokia.com]
> > > > > Sent: Wednesday, March 12, 2003 5:37 PM
> > > > > To: sip@ietf.org
> > > > > Subject: [Sip] NOTIFY establishes a dialog
> > > > > 
> > > > > 
> > > > > In RFC3265, it is mentioned that NOTIFY can certainly 
> > > > > establish a dialog if it arrives before the 200 of a 
> > > > > SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now 
> > > > > arrives with record-route headers? Does the route-set get 
> > > > > updated? (keeping in mind that the dialog is now in the 
> > > > > established state).
> > > > > 
> > > > > Regards,
> > > > > Hisham
> > > > > _______________________________________________
> > > > > 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
> > > 
> > > _______________________________________________
> > > 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

_______________________________________________
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: RE: [Sip] NOTIFY establishes a dialog
From: Robert Sparks <rsparks@dynamicsoft.com>
To: hisham.khartabil@nokia.com
Cc: ganesh.sivaraman@wipro.com, sip@ietf.org
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On Wed, 2003-03-19 at 22:19, hisham.khartabil@nokia.com wrote:
> This sounds like a solution to the first problem. I guess I couldn't find the text because it was in the wrong section. May I suggest a bug report specifying moving the text to section 12.

Added as bug 703.

> The other problem with NOTIFY can be solved by specifying that a NOTIFY establishes an early dialog. Is that ok? bug report?
> 
> Regards,
> Hisham
> 
> > -----Original Message-----
> > From: ext Robert Sparks [mailto:rsparks@dynamicsoft.com]
> > Sent: Thursday, March 20, 2003 7:55 AM
> > To: Ganesh Sivaraman; Khartabil Hisham (NMP/Helsinki)
> > Cc: sip@ietf.org
> > Subject: RE: [Sip] NOTIFY establishes a dialog
> > 
> > 
> > response to one of the questions inline:
> > 
> > On Wed, 2003-03-19 at 20:55, Ganesh Sivaraman wrote:
> > > Hi Hisham,
> > >  I'd posted the same question about setting dialog-state (route-set
> > > etc.) and I got the following response:
> > > 
> > <http://lists.cs.columbia.edu/pipermail/sip-implementors/2003-
> > February/004569.html>
> > > 
> > > It says:
> > > "RFC 3265 augments 3261, in that the dialog in this case 
> > can be created
> > > either by a response to SUBSCRIBE or by NOTIFY".
> > > 
> > > So that would mean that the route-set could be constructed with
> > > Route-headers (present in NOTIFY) also. Is this a correct 
> > assumption?
> > > 
> > > A word from the authors of the RFC on this would help bring more
> > > clarity.
> > > 
> > > Regards
> > > Ganesh
> > > 
> > > On Thu, 2003-03-20 at 04:17, hisham.khartabil@nokia.com wrote:
> > > > I have a few related questions/comments:
> > > > 
> > > > - A 180 response to a request (like INVITE) establishes a 
> > dialog. This 180 response carries record-route headers and 
> > therefore sets the route-set. The 200 response to the request 
> > might also carry record-route headers. The dialog transitions 
> > to confirmed state, but I assume the route-set does not get 
> > updated using the record-route headers in the 200 response 
> > since the dialog is established already. Is this a correct 
> > assumption? should the UAS send record-route headers at all 
> > in the 200 if it sent a 1xx already?
> > 
> > >From rfc3261 page 82:
> > 
> >    If the dialog identifier in the 2xx response matches the dialog
> >    identifier of an existing dialog, the dialog MUST be 
> > transitioned to
> >    the "confirmed" state, and the route set for the dialog MUST be
> >    recomputed based on the 2xx response using the procedures 
> > of Section
> >    12.2.1.2.  Otherwise, a new dialog in the "confirmed" state MUST be
> >    constructed using the procedures of Section 12.1.2.
> > 
> > In context, you'll see the "existing dialog" above is an 
> > "early dialog".
> > 
> > > > 
> > > > Should there be a mandate that 1xx and 2xx response carry 
> > the same record-route headers? You would assume that this 
> > happens naturally at the UAS, but proxies may play around 
> > with the record-route headers.
> > > > 
> > > > - In the NOTIFY problem below, the NOTIFY establishes the 
> > dialog, but does not carry record-route headers. If the same 
> > assumption is made as above, the record-route headers 
> > appearing in the 200 response to the SUBSCRIBE are ignored 
> > since the dialog is in established state. Isn't this a 
> > problem? the route-set is lost.
> > > > 
> > > > I appreciate some clarifications.
> > > > 
> > > > Regards,
> > > > Hisham
> > > > 
> > > > 
> > > > > -----Original Message-----
> > > > > From: ext hisham.khartabil@nokia.com 
> > > > > [mailto:hisham.khartabil@nokia.com]
> > > > > Sent: Wednesday, March 12, 2003 5:37 PM
> > > > > To: sip@ietf.org
> > > > > Subject: [Sip] NOTIFY establishes a dialog
> > > > > 
> > > > > 
> > > > > In RFC3265, it is mentioned that NOTIFY can certainly 
> > > > > establish a dialog if it arrives before the 200 of a 
> > > > > SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now 
> > > > > arrives with record-route headers? Does the route-set get 
> > > > > updated? (keeping in mind that the dialog is now in the 
> > > > > established state).
> > > > > 
> > > > > Regards,
> > > > > Hisham
> > > > > _______________________________________________
> > > > > 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
> > > 
> > > _______________________________________________
> > > 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

_______________________________________________
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: "Arunachalam Venkatraman" <arunvenk@cisco.com>
To: "Robert Sparks" <rsparks@dynamicsoft.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] NOTIFY establishes a dialog
Date: Thu, 20 Mar 2003 09:54:28 -0600
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Robert
Can you explain the reasoning behind the section from RFC3261 that you have
quoted below?
I would think the Record-Route headers on the 1xx and 200 will be identical
since the responses would traverse (backwards) the same path that the INVITE
took, using the same VIA set.
Why should one recompute the Route Set? The only thing that may change is
the Contact, isn't it?
Venkat

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Robert
Sparks
Sent: Wednesday, March 19, 2003 11:55 PM
To: Ganesh Sivaraman; hisham.khartabil@nokia.com
Cc: sip@ietf.org
Subject: RE: [Sip] NOTIFY establishes a dialog


response to one of the questions inline:

On Wed, 2003-03-19 at 20:55, Ganesh Sivaraman wrote:
> Hi Hisham,
>  I'd posted the same question about setting dialog-state (route-set
> etc.) and I got the following response:
>
<http://lists.cs.columbia.edu/pipermail/sip-implementors/2003-February/00456
9.html>
>
> It says:
> "RFC 3265 augments 3261, in that the dialog in this case can be created
> either by a response to SUBSCRIBE or by NOTIFY".
>
> So that would mean that the route-set could be constructed with
> Route-headers (present in NOTIFY) also. Is this a correct assumption?
>
> A word from the authors of the RFC on this would help bring more
> clarity.
>
> Regards
> Ganesh
>
> On Thu, 2003-03-20 at 04:17, hisham.khartabil@nokia.com wrote:
> > I have a few related questions/comments:
> >
> > - A 180 response to a request (like INVITE) establishes a dialog. This
180 response carries record-route headers and therefore sets the route-set.
The 200 response to the request might also carry record-route headers. The
dialog transitions to confirmed state, but I assume the route-set does not
get updated using the record-route headers in the 200 response since the
dialog is established already. Is this a correct assumption? should the UAS
send record-route headers at all in the 200 if it sent a 1xx already?

>From rfc3261 page 82:

   If the dialog identifier in the 2xx response matches the dialog
   identifier of an existing dialog, the dialog MUST be transitioned to
   the "confirmed" state, and the route set for the dialog MUST be
   recomputed based on the 2xx response using the procedures of Section
   12.2.1.2.  Otherwise, a new dialog in the "confirmed" state MUST be
   constructed using the procedures of Section 12.1.2.

In context, you'll see the "existing dialog" above is an "early dialog".

> >
> > Should there be a mandate that 1xx and 2xx response carry the same
record-route headers? You would assume that this happens naturally at the
UAS, but proxies may play around with the record-route headers.
> >
> > - In the NOTIFY problem below, the NOTIFY establishes the dialog, but
does not carry record-route headers. If the same assumption is made as
above, the record-route headers appearing in the 200 response to the
SUBSCRIBE are ignored since the dialog is in established state. Isn't this a
problem? the route-set is lost.
> >
> > I appreciate some clarifications.
> >
> > Regards,
> > Hisham
> >
> >
> > > -----Original Message-----
> > > From: ext hisham.khartabil@nokia.com
> > > [mailto:hisham.khartabil@nokia.com]
> > > Sent: Wednesday, March 12, 2003 5:37 PM
> > > To: sip@ietf.org
> > > Subject: [Sip] NOTIFY establishes a dialog
> > >
> > >
> > > In RFC3265, it is mentioned that NOTIFY can certainly
> > > establish a dialog if it arrives before the 200 of a
> > > SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now
> > > arrives with record-route headers? Does the route-set get
> > > updated? (keeping in mind that the dialog is now in the
> > > established state).
> > >
> > > Regards,
> > > Hisham
> > > _______________________________________________
> > > 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
>
> _______________________________________________
> 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

_______________________________________________
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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	Thu, 20 Mar 2003 10:42:51 -0600
Subject: RE: [Sip] NOTIFY establishes a dialog
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Arunachalam Venkatraman <arunvenk@cisco.com>
Cc: sip@ietf.org
In-Reply-To: <GFEJJOMGHDNDPFCCCIGNCEKKCOAA.arunvenk@cisco.com>
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The only think I can think of this morning (IETF weeks are always
brutal) is that as the text is currently written, the proxies can
rewrite the record-route header field values as the responses
propogate back (to deal with things like multiple interfaces or
transport protocol changes that can't be dealt with via SRV).

Its possible for a proxy to provide a URI with
different parameters on the provisional and final response (to
indicate to itself whether its dealing with an early or confirmed
dialog perhaps). 

Now, subsequent conversations indicate that we probably want to
discourage this rewriting in favor of the double-record-route
solution so that the RR header field in a response can be signed
by the responding endpoint. The quoted text below would become
irrelevant if we made it illegal for proxies to rewrite RR header
field values.

RjS

On Thu, 2003-03-20 at 07:54, Arunachalam Venkatraman wrote:
> Robert
> Can you explain the reasoning behind the section from RFC3261 that you have
> quoted below?
> I would think the Record-Route headers on the 1xx and 200 will be identical
> since the responses would traverse (backwards) the same path that the INVITE
> took, using the same VIA set.
> Why should one recompute the Route Set? The only thing that may change is
> the Contact, isn't it?
> Venkat
> 
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Robert
> Sparks
> Sent: Wednesday, March 19, 2003 11:55 PM
> To: Ganesh Sivaraman; hisham.khartabil@nokia.com
> Cc: sip@ietf.org
> Subject: RE: [Sip] NOTIFY establishes a dialog
> 
> 
> response to one of the questions inline:
> 
> On Wed, 2003-03-19 at 20:55, Ganesh Sivaraman wrote:
> > Hi Hisham,
> >  I'd posted the same question about setting dialog-state (route-set
> > etc.) and I got the following response:
> >
> <http://lists.cs.columbia.edu/pipermail/sip-implementors/2003-February/00456
> 9.html>
> >
> > It says:
> > "RFC 3265 augments 3261, in that the dialog in this case can be created
> > either by a response to SUBSCRIBE or by NOTIFY".
> >
> > So that would mean that the route-set could be constructed with
> > Route-headers (present in NOTIFY) also. Is this a correct assumption?
> >
> > A word from the authors of the RFC on this would help bring more
> > clarity.
> >
> > Regards
> > Ganesh
> >
> > On Thu, 2003-03-20 at 04:17, hisham.khartabil@nokia.com wrote:
> > > I have a few related questions/comments:
> > >
> > > - A 180 response to a request (like INVITE) establishes a dialog. This
> 180 response carries record-route headers and therefore sets the route-set.
> The 200 response to the request might also carry record-route headers. The
> dialog transitions to confirmed state, but I assume the route-set does not
> get updated using the record-route headers in the 200 response since the
> dialog is established already. Is this a correct assumption? should the UAS
> send record-route headers at all in the 200 if it sent a 1xx already?
> 
> >From rfc3261 page 82:
> 
>    If the dialog identifier in the 2xx response matches the dialog
>    identifier of an existing dialog, the dialog MUST be transitioned to
>    the "confirmed" state, and the route set for the dialog MUST be
>    recomputed based on the 2xx response using the procedures of Section
>    12.2.1.2.  Otherwise, a new dialog in the "confirmed" state MUST be
>    constructed using the procedures of Section 12.1.2.
> 
> In context, you'll see the "existing dialog" above is an "early dialog".
> 
> > >
> > > Should there be a mandate that 1xx and 2xx response carry the same
> record-route headers? You would assume that this happens naturally at the
> UAS, but proxies may play around with the record-route headers.
> > >
> > > - In the NOTIFY problem below, the NOTIFY establishes the dialog, but
> does not carry record-route headers. If the same assumption is made as
> above, the record-route headers appearing in the 200 response to the
> SUBSCRIBE are ignored since the dialog is in established state. Isn't this a
> problem? the route-set is lost.
> > >
> > > I appreciate some clarifications.
> > >
> > > Regards,
> > > Hisham
> > >
> > >
> > > > -----Original Message-----
> > > > From: ext hisham.khartabil@nokia.com
> > > > [mailto:hisham.khartabil@nokia.com]
> > > > Sent: Wednesday, March 12, 2003 5:37 PM
> > > > To: sip@ietf.org
> > > > Subject: [Sip] NOTIFY establishes a dialog
> > > >
> > > >
> > > > In RFC3265, it is mentioned that NOTIFY can certainly
> > > > establish a dialog if it arrives before the 200 of a
> > > > SUBSCRIBE. What happens when the 200 for the SUBSCRIBE now
> > > > arrives with record-route headers? Does the route-set get
> > > > updated? (keeping in mind that the dialog is now in the
> > > > established state).
> > > >
> > > > Regards,
> > > > Hisham
> > > > _______________________________________________
> > > > 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
> >
> > _______________________________________________
> > 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

_______________________________________________
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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Date: Thu, 20 Mar 2003 19:24:31 +0200
To: Robert Sparks <rsparks@dynamicsoft.com>
Cc: Arunachalam Venkatraman <arunvenk@cisco.com>, sip@ietf.org
Subject: RE: [Sip] NOTIFY establishes a dialog
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Robert Sparks writes:

 > Now, subsequent conversations indicate that we probably want to
 > discourage this rewriting in favor of the double-record-route
 > solution so that the RR header field in a response can be signed
 > by the responding endpoint. The quoted text below would become
 > irrelevant if we made it illegal for proxies to rewrite RR header
 > field values.

at least proxys should be allowed to add parameters to record
route/route headers.

by the way, i think it is a big mistake that the current sip rfc doesn't
allow a re-invite to include a new record route.  consider the case
where a call originates in a mobile network and then roams to wlan, for
example.  it would be really stupid if the subsequent messages of the
dialog would need to go via the proxies of the mobile network.

-- juha

_______________________________________________
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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Can a Cancel have a message body, and if so does a stateful proxy have
the same responsibility to maintain that message body in a generated
Cancel request as with other messages?  Also, is the proxy required to
maintain ALL headers, such as a "Reason" header in the original Cancel
request?  Thanks for the help.
=20
Giles

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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Can a Cancel have a message body, and if so does a =
stateful
proxy have the same responsibility to maintain that message body in a =
generated
Cancel request as with other messages?<span =
style=3D'mso-spacerun:yes'>&nbsp;
</span>Also, is the proxy required to maintain ALL headers, such as a =
&#8220;Reason&#8221;
header in the original Cancel request?<span =
style=3D'mso-spacerun:yes'>&nbsp;
</span>Thanks for the help.<o:p></o:p></span></font></p>

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

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

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Subject: Re: [Sip] NOTIFY establishes a dialog
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Hi Robert,

> >From rfc3261 page 82:
>
>    If the dialog identifier in the 2xx response matches the dialog
>    identifier of an existing dialog, the dialog MUST be transitioned to
>    the "confirmed" state, and the route set for the dialog MUST be
>    recomputed based on the 2xx response using the procedures of Section
>    12.2.1.2.  Otherwise, a new dialog in the "confirmed" state MUST be
>    constructed using the procedures of Section 12.1.2.
>
> In context, you'll see the "existing dialog" above is an "early dialog".

Chapter 4 in RFC3262 says:

"The UAC MAY acknowledge reliable provisional responses received after
the final response or MAY discard them."

Now, IF the UAC chooses to acknowledge reliable provisional responses after the final response is received, which route set should it use for the PRACK? The route set it used for the other PRACKs in the "early state", or the possible new route set is has for the "confirmed state"?

Regards,

Christer Holmberg
Ericsson Finland


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Subject: Re: [Sip] NOTIFY establishes a dialog
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
Cc: sip@ietf.org
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Nice question.

>From a constructionist view, the PRACK is clearly associated with the
early dialog and should use the early dialog route set. From a practical
point of view, using either the early or confirmed route set gets the
PRACK to the place that needs it, so what does it matter? (Can you
construct a case where a proxy would care?)

I think, though, that the idea that the early and established route
sets can be different is just an artifact of the way we defined
record-route rewriting instead of a feature we really intended to
engineer. (If someone knows differently, please speak up). The
double-record-route solution for those cases where rewriting
is necessary would allow the RR field in a response to be signed.
That would _prevent_ the early and confirmed route sets from being
different. I suspect that's a more desired behavior.

So to answer your question more succinctly:
 - I don't think it matters which you use
 - The potential difference should go away anyhow

RjS

On Thu, 2003-03-20 at 21:49, Christer Holmberg wrote:
> Hi Robert,
> 
> > >From rfc3261 page 82:
> >
> >    If the dialog identifier in the 2xx response matches the dialog
> >    identifier of an existing dialog, the dialog MUST be transitioned to
> >    the "confirmed" state, and the route set for the dialog MUST be
> >    recomputed based on the 2xx response using the procedures of Section
> >    12.2.1.2.  Otherwise, a new dialog in the "confirmed" state MUST be
> >    constructed using the procedures of Section 12.1.2.
> >
> > In context, you'll see the "existing dialog" above is an "early dialog".
> 
> Chapter 4 in RFC3262 says:
> 
> "The UAC MAY acknowledge reliable provisional responses received after
> the final response or MAY discard them."
> 
> Now, IF the UAC chooses to acknowledge reliable provisional responses after the final response is received, which route set should it use for the PRACK? The route set it used for the other PRACKs in the "early state", or the possible new route set is has for the "confirmed state"?
> 
> Regards,
> 
> Christer Holmberg
> Ericsson Finland

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hai
   I am new to ser and sip. I would like  to know how to connect ser and UA. I have Red Hat Linux V. 8 and Windows NT and For UA I have Xphone from xten
Thankz 
Keerthi

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From: Milinski Alexander <alexander.milinski@siemens.com>
To: "'Kevin Lingle'" <klingle@cisco.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
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Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-05.txt
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Hi Kevin and Janathan,
the concern raised by Jonathan seems very valid. Like Jonathan I do not think that it is benefitial to make transient information (like details of each ongoing dialog in the SipTransactionEntry) available via the SIP MIB.
Kind regards,
Alexander

Alexander Milinski
Siemens mobile

-----Original Message-----
From: Kevin Lingle [mailto:klingle@cisco.com]
Sent: Wednesday, March 19, 2003 4:40 PM
To: Jonathan Rosenberg
Cc: sip@ietf.org; Dave Walker
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-mib-05.txt


Jonathan Rosenberg wrote:
> 
> Kevin Lingle wrote:
> 
> >>I have begun to look at the mib a bit, and I am a bit concerned that
> >>there is far too much information exposed, in terms of statistics, at
> >>least. I have to apologize for raising this so late in the game, but it
> >>is only recently that my day job has brought me into OAMP areas.
> >
> >
> > not sure if the above reference to statistics is directly related
> > to your comment on sipTransactionTable below or not.
> 
> Yes.
> 
> > if not, are there
> > other statistics that you feel fall into the category or
> > exposing far too much information?
> 
> I want to look some more and provide some additional comments.

that would be great.

> 
> >
> >
> >>One item that struck me as particularly demanding on an implementation
> >>is the sipTransactionTable. There is a huge amount of work in
> >>maintaining this table. I really doubt people will want to implement it.
> >
> >
> > i had similar feedback from others when support of the sipTransactionTable
> > was being considered by a project.    the effort involved in supporting
> > the table was a problem.
> >
> > we can focus some discussion on the viability of the table during wg last call.
> 
> Well, discussion can happen any time. Its happening now, in this email
> thread. I dont know what else you are expecting. 

sorry.  didn't mean to imply what came across.  we can definitely discuss
it now.  i'm just not used to actually having active discussion of the mib.
i'm used to getting one or two sporatic comments and only having focused
discussion during last call - when someone is forced to review the mib ;)

> I don't think the table
> is viable. You had similar feedback. Seems like a clear cut resolution
> to me.

not that i don't take your point of view as very significant, i do.  
i was hoping that others might concur before we decide to remove the table.
the table has been in every revision of the mib, so it has been reviewed
multiple times by multiple people... no one - until now has recommended it
come out.   i guess i had taken that as acceptance of it's potential usefulness
to someone.  my case of one project finding it questionable does not necessarily
make a case for it's removal.   we determined that support was feasible but
not all that useful for our platform.  

implementation experience with a mib usually decides such things and we don't 
have much (any?) feedback in that area (not likely to until the mibs are in an
rfc).   still, if we can resolve the question of usefulness now it would be
better than 
to have to revise the mib (ie, rfc) and deprecate objects found - through 
implementation experience - to be useless.

> 
> > i'd rather do that than to yank the table w/out further discussion.
> 
> *This* is the discussion.

agreed.

> 
> > at the very least, we can definitely make the table optional.
> > right now it appears to be in a mandatory group - which seem wrong to me anyway.
> 
> Mandatory is definitely wrong. I am not even sure optional is useful.
> 
> So - here is the question. What is the bar for putting something in the
> MIB? At one extreme, every piece of state managed by a server could be
> in the mib. At the other extreme, there is nothing, or a single summary
> statistic. I think a reasonable bar, is that there should be clear value
> in monitoring that information, from the perspective of fault and
> performance management in particular. If you cannot demonstrate such a
> use, it should be out.
> 
> I cannot think of a reason that I would want, from an operations center,
> to monitor the current set of transactions held in the system. Thus, I
> think it should be out.

it's hard for me to justify or defend the existence of this table as i
was not the original author of it.  but for the sake of discussion...

is there no value from a troubleshooting perspective?   you mentioned that
this information is likely logged for offline analysis - not real-time.
are transactions generally occuring too fast for real-time analysis to
be of worth?   it would seem to me that unless you know specifically which
transactions are of interest to you, it would be hard to use this table
for anything.  the table is defintely not a logging mechanism as transactions
will not persist for any historically significant time.

from a performance mgmt point of view...
could the increase in entries in this table be used to gain insight into
congestion in this or other parts of the sip network?   even if that were
the case, the simple sipCurrentTransactions gauge object would provide
similar insight (to congestion at least).   perhaps the to/from details 
in the table would lend some more insight to problems elsewhere in the
network?   i'm not sure.   

dave walker was the original author of this table, so maybe
he can offer some insight.  dave? 
i did find one old comment related to this table squirreled away in some 
old emails....

Kurt Robohm <kurt@mci.net> wrote:
> 
> sipCurrentTransacations         Does this reflect number of
>                                 calls in progress?
>                                 Intent of objects in table to
> sipTransactionTable             provide real time info on
>                                 number of calls being handled
>                                 real time?

Dave Walker <drwalker@ss8networks.com> responded:
> Transactions aren't calls.  The intent, as mentioned in Adelaide,
> is to evolve the transaction table to a call leg table (probably 
> for UAs only).  This hasn't been resolved yet, so no real changes 
> were made.


i can't find any further discussion by dave (or others) in subsequent 
revisions of the draft to work out any further evolution of this table.  
only minor comments related to the table - none questioning its existence.
dave's involvement in the last couple of revisions of the draft have been nil.
so perhaps the table is an articfact of an idea that was not fully
conceived or fleshed out.  that would lend itself to removal imo.

kevin
> 
> Is not our aim to take things out of protocols until we can't take out
> any more, rather than the other way around?
> 
> -Jonathan R.
> 
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> 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

-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
_______________________________________________
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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thanks for all of the input wrt sipTransactionTable.
consider the table removed from the mib.    i plan on 
keeping the sipCurrentTransactions gauge object but will
make it part of an optional group.

shrinivas,
thanks for you other comments. i'll go through them soon.

kevin

sshetti@hss.hns.com wrote:
> 
> We too have sip-mib implementation in our product.
> Following is my view with respect to sipTransactionTable:
> 
> 1. Maintaining this table info in the system is not
>    'simple and straight forward'.
>    Reason: Data involved are very volatile.
>    Moreover it requires indexing the transactions which
>    themselves are volatile (destroyed any time).
> 
> 2. This information is usually obtained (if et-all obtained!)
>    will be via GET-BULK operation. To support GET-BULK to
>    retrieve information specific to all transactions
>    has momentory performance impact on the Server.
>    There are issues of thread-safety between GET-BULK
>    and the transaction handling operation.
> 
> 3. Same information can be obtained either through
>    protocol trace logs or CDR (call detail records).
> 
> I feel sip-mib should not have tables involving volatile data.
> 
> I also feel sipCurrentTransTable is not useful.
> May be TRAP objects based on sipCurrentTransTable, will be
> more useful to SNMP admin.
> 
> I think sip-mib has captured very elaborately on following things.
> 1. Server Administration/Configuration
> 2. User data management
> 3. Statistics
> I really appreciate the above factor.
> 
> I have some comments on following MIB fields, please
> clarify them as my understanding may differ from the
> auther's of sip-mib.
> 
> 1. "sipMaxSessions" of SipCommonCfgEntry:
>    This is a READ-ONLY property supplied by the
>    SIP-ENTITY. As per the definition the entity
>    has to determine what is the maximum load it
>    can handle.
> 
>    Again, the above factor depends on the
>    hardware/software configuration of the deployment
>    platform. Do we expect for instance a SIP-Proxy
>    to run some algorithm/load itself to find out
>    this value?
> 
>    I feel this field may not be useful.
> 
> 2. "sipTransportSnd" of SipPortEntry: I donot
>    understand how this field is utilized in SIP,
>    specific to a port number.
> 
>    "sipTransportRcv" can be understood as for
>    associated port number the SIP entity will
>    receive message for TRANSPORTS that are
>    enabled in this field.
> 
> 3. "sipPgpVersion" of SipServerCfgEntry,
>    "sipPgpPrivateKey" of SipProxyCfgEntry:
>    PGP is deprecated in SIP. Can these be removed?
>    There may be more things specific to PGP.
> 
> 4. "sipServerRespectUAAction" of SipServerCfgEntry:
>    This mentions action parameter specific operation
>    in the proxy which is deprecated in SIP.
>    Can we deprecate here also?
> 
> 5. "sipRegMaxUsers" of "SipRegCfgEntry":
>    This is a READ-ONLY field which is supposed
>    to be given out by REGISTRAR as what is the
>    number of users does it support.
> 
>    Again, what is the criteria with which registrar
>    arrives at this data. This data again depends on
>    deployment platform such as capacity of the
>    database etc. This data too may not be useful
>    to administrator. This value is either known
>    to the admin or enforced by admin on REGISTRAR.
>    Therefore keeping it as READ-ONLY has no purpose.
> 
> 6. "sipContactAction" of SipContactEntry:
>    Again deprecated by SIP. Can we deprecate here too?
> 
> 7. "sipContactRetryAfter" of SipContactEntry:
>    Again deprecated by SIP. Can we deprecate here too?
> 
> 8. Transaction timer specific configuration are based
>    on pre-bis05 drafts. They have to be updated
>    according to latest RFC. I think action-item is already
>    initiated for post mib-05 drafts.
> 
> Thanks and Regards,
> Shrinivas Shetti
> 
> Kevin Lingle <klingle@cisco.com> on 03/19/2003 09:10:17 PM
> 
> To:   Jonathan Rosenberg <jdrosen@dynamicsoft.com>
> cc:   sip@ietf.org, Dave Walker <drwalker@ss8networks.com> (bcc: Shrinivas
>       Shetti/HSSBLR)
> 
> Subject:  Re: [Sip] I-D ACTION:draft-ietf-sip-mib-05.txt
> 
> Jonathan Rosenberg wrote:
> >
> > Kevin Lingle wrote:
> >
> > >>I have begun to look at the mib a bit, and I am a bit concerned that
> > >>there is far too much information exposed, in terms of statistics, at
> > >>least. I have to apologize for raising this so late in the game, but it
> > >>is only recently that my day job has brought me into OAMP areas.
> > >
> > >
> > > not sure if the above reference to statistics is directly related
> > > to your comment on sipTransactionTable below or not.
> >
> > Yes.
> >
> > > if not, are there
> > > other statistics that you feel fall into the category or
> > > exposing far too much information?
> >
> > I want to look some more and provide some additional comments.
> 
> that would be great.
> 
> >
> > >
> > >
> > >>One item that struck me as particularly demanding on an implementation
> > >>is the sipTransactionTable. There is a huge amount of work in
> > >>maintaining this table. I really doubt people will want to implement
> it.
> > >
> > >
> > > i had similar feedback from others when support of the
> sipTransactionTable
> > > was being considered by a project.    the effort involved in supporting
> > > the table was a problem.
> > >
> > > we can focus some discussion on the viability of the table during wg
> last call.
> >
> > Well, discussion can happen any time. Its happening now, in this email
> > thread. I dont know what else you are expecting.
> 
> sorry.  didn't mean to imply what came across.  we can definitely discuss
> it now.  i'm just not used to actually having active discussion of the mib.
> i'm used to getting one or two sporatic comments and only having focused
> discussion during last call - when someone is forced to review the mib ;)
> 
> > I don't think the table
> > is viable. You had similar feedback. Seems like a clear cut resolution
> > to me.
> 
> not that i don't take your point of view as very significant, i do.
> i was hoping that others might concur before we decide to remove the table.
> the table has been in every revision of the mib, so it has been reviewed
> multiple times by multiple people... no one - until now has recommended it
> come out.   i guess i had taken that as acceptance of it's potential
> usefulness
> to someone.  my case of one project finding it questionable does not
> necessarily
> make a case for it's removal.   we determined that support was feasible but
> not all that useful for our platform.
> 
> implementation experience with a mib usually decides such things and we
> don't
> have much (any?) feedback in that area (not likely to until the mibs are in
> an
> rfc).   still, if we can resolve the question of usefulness now it would be
> better than
> to have to revise the mib (ie, rfc) and deprecate objects found - through
> implementation experience - to be useless.
> 
> >
> > > i'd rather do that than to yank the table w/out further discussion.
> >
> > *This* is the discussion.
> 
> agreed.
> 
> >
> > > at the very least, we can definitely make the table optional.
> > > right now it appears to be in a mandatory group - which seem wrong to
> me anyway.
> >
> > Mandatory is definitely wrong. I am not even sure optional is useful.
> >
> > So - here is the question. What is the bar for putting something in the
> > MIB? At one extreme, every piece of state managed by a server could be
> > in the mib. At the other extreme, there is nothing, or a single summary
> > statistic. I think a reasonable bar, is that there should be clear value
> > in monitoring that information, from the perspective of fault and
> > performance management in particular. If you cannot demonstrate such a
> > use, it should be out.
> >
> > I cannot think of a reason that I would want, from an operations center,
> > to monitor the current set of transactions held in the system. Thus, I
> > think it should be out.
> 
> it's hard for me to justify or defend the existence of this table as i
> was not the original author of it.  but for the sake of discussion...
> 
> is there no value from a troubleshooting perspective?   you mentioned that
> this information is likely logged for offline analysis - not real-time.
> are transactions generally occuring too fast for real-time analysis to
> be of worth?   it would seem to me that unless you know specifically which
> transactions are of interest to you, it would be hard to use this table
> for anything.  the table is defintely not a logging mechanism as
> transactions
> will not persist for any historically significant time.
> 
> from a performance mgmt point of view...
> could the increase in entries in this table be used to gain insight into
> congestion in this or other parts of the sip network?   even if that were
> the case, the simple sipCurrentTransactions gauge object would provide
> similar insight (to congestion at least).   perhaps the to/from details
> in the table would lend some more insight to problems elsewhere in the
> network?   i'm not sure.
> 
> dave walker was the original author of this table, so maybe
> he can offer some insight.  dave?
> i did find one old comment related to this table squirreled away in some
> old emails....
> 
> Kurt Robohm <kurt@mci.net> wrote:
> >
> > sipCurrentTransacations         Does this reflect number of
> >                                 calls in progress?
> >                                 Intent of objects in table to
> > sipTransactionTable             provide real time info on
> >                                 number of calls being handled
> >                                 real time?
> 
> Dave Walker <drwalker@ss8networks.com> responded:
> > Transactions aren't calls.  The intent, as mentioned in Adelaide,
> > is to evolve the transaction table to a call leg table (probably
> > for UAs only).  This hasn't been resolved yet, so no real changes
> > were made.
> 
> i can't find any further discussion by dave (or others) in subsequent
> revisions of the draft to work out any further evolution of this table.
> only minor comments related to the table - none questioning its existence.
> dave's involvement in the last couple of revisions of the draft have been
> nil.
> so perhaps the table is an articfact of an idea that was not fully
> conceived or fleshed out.  that would lend itself to removal imo.
> 
> kevin
> >
> > Is not our aim to take things out of protocols until we can't take out
> > any more, rather than the other way around?
> >
> > -Jonathan R.
> >
> > --
> > Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > 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
> 
> --
> =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
>  Kevin R. Lingle       919.392.2029
> =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
> _______________________________________________
> 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

-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
_______________________________________________
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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Everist, Giles <Giles.Everist@level3.com> writes:
><html xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns="http://www.w3.org/TR/REC-html40">

	Kewl.

>Can a Cancel have a message body, and if so does a stateful proxy have
>the same responsibility to maintain that message body in a generated
>Cancel request as with other messages?  Also, is the proxy required to
>maintain ALL headers, such as a "Reason" header in the original Cancel
>request?  Thanks for the help.

	CANCEL is a hop-by-hop request, so stateful proxies do not
	forward it. No message-body or headers get forwarded.

					Pekka
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Hi Robert,

> From a constructionist view, the PRACK is clearly associated with the
> early dialog and should use the early dialog route set. From a practical
> point of view, using either the early or confirmed route set gets the
> PRACK to the place that needs it, so what does it matter? (Can you
> construct a case where a proxy would care?)

First, if we use connection protocols, with or without security (TCP, TLS etc), nodes may have to establish two connections (if the next hop is different in the early- and confirmed route sets). This affects, of course, both UAs and proxies.

Second, since we are not allowed to change the route once the dialog has been "confirmed", there may be proxies which checks the route for incoming request. And, those proxies may not know the semantics of PRACK, so they may think the request is for the confirmed dialog, and reject it
because it tries to use another route.

Third, this MAY also affect other early dialog requests (UPDATE, INFO), which due to race conditions are sent after the dialog has been confirmed. >From a proxy point of view, however, I guess this is not different from the PRACK case.

Fourth, but this I haven't thought so much about yet, could this somehow affect the work on connection reuse?

> I think, though, that the idea that the early and established route
> sets can be different is just an artifact of the way we defined
> record-route rewriting instead of a feature we really intended to
> engineer.

Personally I don't see any reason why it should be allowed in the first place. Because, we are NOT allowed to change the Contact header in 18x and 200 responses, are we? So, why should we be allowed to change Record-Route?

Of course, IF someone has reasons why proxies should be involved only in the early state, or in the confirmed state, that is another issue, but in the case the question is if we should have a generic way of changing the route set - also after the dialog has been confirmed (Juha
mentioned cases where such functionality could be useful).

Regards,

Christer Holmberg
Ericsson Finland






> (If someone knows differently, please speak up). The
> double-record-route solution for those cases where rewriting
> is necessary would allow the RR field in a response to be signed.
> That would _prevent_ the early and confirmed route sets from being
> different. I suspect that's a more desired behavior.
>
> So to answer your question more succinctly:
>  - I don't think it matters which you use
>  - The potential difference should go away anyhow
>
> RjS
>
> On Thu, 2003-03-20 at 21:49, Christer Holmberg wrote:
> > Hi Robert,
> >
> > > >From rfc3261 page 82:
> > >
> > >    If the dialog identifier in the 2xx response matches the dialog
> > >    identifier of an existing dialog, the dialog MUST be transitioned to
> > >    the "confirmed" state, and the route set for the dialog MUST be
> > >    recomputed based on the 2xx response using the procedures of Section
> > >    12.2.1.2.  Otherwise, a new dialog in the "confirmed" state MUST be
> > >    constructed using the procedures of Section 12.1.2.
> > >
> > > In context, you'll see the "existing dialog" above is an "early dialog".
> >
> > Chapter 4 in RFC3262 says:
> >
> > "The UAC MAY acknowledge reliable provisional responses received after
> > the final response or MAY discard them."
> >
> > Now, IF the UAC chooses to acknowledge reliable provisional responses after the final response is received, which route set should it use for the PRACK? The route set it used for the other PRACKs in the "early state", or the possible new route set is has for the "confirmed state"?
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland

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Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-05.txt
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There is a delicate balance to be looked for here. I doubt that 'every piece pf state' is useful information, both on grounds of resources, as well quantity of information that needs to be processed. SNMP agents run on the same hosts that run protocols, and the management information carried from SNMP agents to SNMP clients (managers) competes for bandwidth with other data traffic. When designing a MIB one needs to take into account that in order for the MIB to be useful, it needs to run in the context of the host and application resources, and that the management traffic that is generated needs to be a reasonable fraction of the traffic of the protocol or application that is being managed. In the context of the current discussion it looks to me that large state or session tables should be avoided, with more compact indications, or threshold based alarms being put in place instead.

Dan


> So - here is the question. What is the bar for putting 
> something in the 
> MIB? At one extreme, every piece of state managed by a server 
> could be 
> in the mib. At the other extreme, there is nothing, or a 
> single summary 
> statistic. I think a reasonable bar, is that there should be 
> clear value 
> in monitoring that information, from the perspective of fault and 
> performance management in particular. If you cannot 
> demonstrate such a 
> use, it should be out.
> 
> 
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From: "Cullen Jennings" <fluffy@cisco.com>
To: "SVR Anand" <anand@eis.iisc.ernet.in>, <loretosa@vodafone.it>
Cc: <sip@ietf.org>
Subject: RE: Ri: [Sip] SIP proxy network
Date: Sun, 23 Mar 2003 22:58:14 -0800
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You might want to have a read of TRIP (RFC 3219)

 ftp://ftp.rfc-editor.org/in-notes/rfc3219.txt

Is based off of BGP4 instead of OSPF.


> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of SVR
> Anand
> Sent: Wednesday, March 19, 2003 2:34 AM
> To: loretosa@vodafone.it
> Cc: sip@ietf.org
> Subject: Re: Ri: [Sip] SIP proxy network
>
>
> The kind of information the proxies exchange should enable the UA
> or local proxy
> to dynamically locate and choose where to forward say,INVITE to.
> Since the
> information contained within proxies is at the application level
> I am thinking if
> an application level and perhaps text based routing protocol is necessary.
>
>
> >
> > I'm not sure about your question,
> > are you asking about the possibility for UA or an
> > inBound Proxy to choose the best way where send
> > a message (like INVITE)?
> >
> >
> > > Hi,
> > >
> > > Not sure if I am sending my mail to the proper mailing list.
> > >
> > > Just a wild thought. Can I think of network of SIP proxies
> connected in some
> > > way dynamically exchanging messages in order to assist in
> forwarding of say,
> > > INVITES to an appropriate proxy ? Just like OSPF for link
> state information,
> > > proxy network for say proxy configuration information exchange ?
> > >
> > > Please let me know if I am talking sense.
> > >
> > > Anand
> > > _______________________________________________
> > > 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
> > >
> >
> >  Per registrarti gratuitamente a Vodafone Mail vai su www.190.it
> >
> >
>
> _______________________________________________
> 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
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>

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From: "Cullen Jennings" <fluffy@cisco.com>
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Subject: [Sip] SIP MIB implementations?
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I'm wondering about actual implementation experience with the SIP MIB. Has
anyone implemented any version or portion of it? Any operational experience
folks could provide on what turned out to be useful?



> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> Internet-Drafts@ietf.org
> Sent: Friday, March 07, 2003 3:55 AM
> To: IETF-Announce:
> Cc: sip@ietf.org
> Subject: [Sip] I-D ACTION:draft-ietf-sip-mib-05.txt
>
>
> 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		: Management Information Base for Session
> Initiation
>                           Protocol
> 	Author(s)	: K. Lingle, J. Maeng, J. Mule, D. Walker
> 	Filename	: draft-ietf-sip-mib-05.txt
> 	Pages		: 97
> 	Date		: 2003-3-6
>
> This memo defines a portion of the Management Information Base (MIB)
> for use with network management protocols in the Internet community.
> In particular, it describes a set of managed objects that are used
> to manage Session Initiation Protocol (SIP) entities, which include
> User Agents, Proxy servers, Redirect servers and Registrars.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-05.txt
>
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From: "James Ford" <james_s_ford@hotmail.com>
To: sip@ietf.org
Cc: eronstein@hotmail.com
Subject: Re: [Sip] TLS post connection verification
Date: Mon, 24 Mar 2003 12:51:48 +0000
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Hi Eron,
The standard does not elaborate on that, but here is what I think:

For the second question: for sending responses on broken connections, I 
don't think that there is a need to do post connection assertion. for two 
reasons:
a - The server does not need to authenticate the client.
b - allot of applications put their specific IP address on the via field 
while a certificate is more likely issued to an FQDN.

The first question is more tricky. My guess is the "incoming" connection 
should only check for the correctness of the certificate and not do a post 
connection assertion. This might be a security issue though. Maybe someone 
with some hands on experience can would be more helpful here.

Regards,
James S. Ford


>Hi,
>
>When sending a request (ie REGISTER) to a server I can compare the request
>URI to the common name (or the alt dns name) in the certificate. If the
>names match, I can conclude that the certificate is OK.
>(I'm using OpenSSL, and they recommend this post connection assertion).
>
>I have two questions thou:
>
>1 - What name should I use for comparison when accepting a connection?
>Usually only the UAC will demand certificate, I am concerned with te case 
>of
>
>two proxies trying to connect using TLS and the UAS proxy asking for client
>certificates. (what uri will the UAS proxy has, there is no message yet).
>
>2 - how should broken connection be handled? lets say UAC1 sent a request
>over TLS to UAS1. the handshake went well and the request sent. than for
>some reason, the connection was broken and UAS1 now needs to reestablish 
>the
>
>connection. What should UAS1 do? use TLS w/out certificates?
>
>
>Regards,
>Eron Stein
>
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To: "Cullen Jennings" <fluffy@cisco.com>
Cc: sip@ietf.org, sip-admin@ietf.org
Subject: Re: [Sip] SIP MIB implementations?
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Yes - indeed we have - developed and deployed -> you may want to follow 
some previous threads of some real-life issues
we have been facing wrt. dynamic data (see mail by sshetti@hss.hns.com in 
archives as a starter). We'd be interested in hearing
from deployment experience from others too.

regds
arjun

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860  (M): 240 997 0066{@vtext.com}




"Cullen Jennings" <fluffy@cisco.com>
Sent by: sip-admin@ietf.org
03/24/2003 02:05 AM

 
        To:     <sip@ietf.org>
        cc: 
        Subject:        [Sip] SIP MIB implementations?



I'm wondering about actual implementation experience with the SIP MIB. Has
anyone implemented any version or portion of it? Any operational 
experience
folks could provide on what turned out to be useful?



> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> Internet-Drafts@ietf.org
> Sent: Friday, March 07, 2003 3:55 AM
> To: IETF-Announce:
> Cc: sip@ietf.org
> Subject: [Sip] I-D ACTION:draft-ietf-sip-mib-05.txt
>
>
> 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                           : Management Information 
Base for Session
> Initiation
>                           Protocol
>                Author(s)               : K. Lingle, J. Maeng, J. Mule, 
D. Walker
>                Filename                : draft-ietf-sip-mib-05.txt
>                Pages                           : 97
>                Date                            : 2003-3-6
>
> This memo defines a portion of the Management Information Base (MIB)
> for use with network management protocols in the Internet community.
> In particular, it describes a set of managed objects that are used
> to manage Session Initiation Protocol (SIP) entities, which include
> User Agents, Proxy servers, Redirect servers and Registrars.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-05.txt
>
> To remove yourself from the IETF Announcement list, send a message to
> ietf-announce-request with the word unsubscribe in the body of
> the message.
>
> Internet-Drafts are also available by anonymous FTP. Login with
> the username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
>                "get draft-ietf-sip-mib-05.txt".
>
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
> Internet-Drafts can also be obtained by e-mail.
>
> Send a message to:
>                mailserv@ietf.org.
> In the body type:
>                "FILE /internet-drafts/draft-ietf-sip-mib-05.txt".
>
> NOTE:          The mail server at ietf.org can return the document in
>                MIME-encoded form by using the "mpack" utility.  To use 
this
>                feature, insert the command "ENCODING mime" before the 
"FILE"
>                command.  To decode the response(s), you will need 
"munpack" or
>                a MIME-compliant mail reader.  Different MIME-compliant 
mail readers
>                exhibit different behavior, especially when dealing with
>                "multipart" MIME messages (i.e. documents which have been 
split
>                up into multiple messages), so check your local 
documentation on
>                how to manipulate these messages.
>
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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<br><font size=2 face="sans-serif">Yes - indeed we have - developed and deployed -&gt; you may want to follow some previous threads of some real-life issues</font>
<br><font size=2 face="sans-serif">we have been facing wrt. dynamic data (see mail by sshetti@hss.hns.com in archives as a starter). We'd be interested in hearing</font>
<br><font size=2 face="sans-serif">from deployment experience from others too.</font>
<br>
<br><font size=2 face="sans-serif">regds</font>
<br><font size=2 face="sans-serif">arjun</font>
<br>
<br><font size=2 face="sans-serif">--<br>
Arjun Roychowdhury @ Hughes Software Systems<br>
11717 Exploration Lane, Germantown MD 20876<br>
(O): 301 212 7860 &nbsp;(M): 240 997 0066{@vtext.com}</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td><font size=1 face="sans-serif"><b>&quot;Cullen Jennings&quot; &lt;fluffy@cisco.com&gt;</b></font>
<br><font size=1 face="sans-serif">Sent by: sip-admin@ietf.org</font>
<p><font size=1 face="sans-serif">03/24/2003 02:05 AM</font>
<br>
<td><font size=1 face="Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;&lt;sip@ietf.org&gt;</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;[Sip] SIP MIB implementations?</font></table>
<br>
<br>
<br><font size=2 face="Courier New"><br>
I'm wondering about actual implementation experience with the SIP MIB. Has<br>
anyone implemented any version or portion of it? Any operational experience<br>
folks could provide on what turned out to be useful?<br>
<br>
<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of<br>
&gt; Internet-Drafts@ietf.org<br>
&gt; Sent: Friday, March 07, 2003 3:55 AM<br>
&gt; To: IETF-Announce:<br>
&gt; Cc: sip@ietf.org<br>
&gt; Subject: [Sip] I-D ACTION:draft-ietf-sip-mib-05.txt<br>
&gt;<br>
&gt;<br>
&gt; A New Internet-Draft is available from the on-line<br>
&gt; Internet-Drafts directories.<br>
&gt; This draft is a work item of the Session Initiation Protocol<br>
&gt; Working Group of the IETF.<br>
&gt;<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Title &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: Management Information Base for Session<br>
&gt; Initiation<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Protocol<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Author(s) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : K. Lingle, J. Maeng, J. Mule, D. Walker<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Filename &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : draft-ietf-sip-mib-05.txt<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Pages &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: 97<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Date &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: 2003-3-6<br>
&gt;<br>
&gt; This memo defines a portion of the Management Information Base (MIB)<br>
&gt; for use with network management protocols in the Internet community.<br>
&gt; In particular, it describes a set of managed objects that are used<br>
&gt; to manage Session Initiation Protocol (SIP) entities, which include<br>
&gt; User Agents, Proxy servers, Redirect servers and Registrars.<br>
&gt;<br>
&gt; A URL for this Internet-Draft is:<br>
&gt; http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-05.txt<br>
&gt;<br>
&gt; To remove yourself from the IETF Announcement list, send a message to<br>
&gt; ietf-announce-request with the word unsubscribe in the body of<br>
&gt; the message.<br>
&gt;<br>
&gt; Internet-Drafts are also available by anonymous FTP. Login with<br>
&gt; the username<br>
&gt; &quot;anonymous&quot; and a password of your e-mail address. After logging in,<br>
&gt; type &quot;cd internet-drafts&quot; and then<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&quot;get draft-ietf-sip-mib-05.txt&quot;.<br>
&gt;<br>
&gt; A list of Internet-Drafts directories can be found in<br>
&gt; http://www.ietf.org/shadow.html<br>
&gt; or ftp://ftp.ietf.org/ietf/1shadow-sites.txt<br>
&gt;<br>
&gt;<br>
&gt; Internet-Drafts can also be obtained by e-mail.<br>
&gt;<br>
&gt; Send a message to:<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;mailserv@ietf.org.<br>
&gt; In the body type:<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&quot;FILE /internet-drafts/draft-ietf-sip-mib-05.txt&quot;.<br>
&gt;<br>
&gt; NOTE: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; The mail server at ietf.org can return the document in<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;MIME-encoded form by using the &quot;mpack&quot; utility. &nbsp;To use this<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;feature, insert the command &quot;ENCODING mime&quot; before the &quot;FILE&quot;<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;command. &nbsp;To decode the response(s), you will need &quot;munpack&quot; or<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;a MIME-compliant mail reader. &nbsp;Different MIME-compliant mail readers<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;exhibit different behavior, especially when dealing with<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&quot;multipart&quot; MIME messages (i.e. documents which have been split<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;up into multiple messages), so check your local documentation on<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;how to manipulate these messages.<br>
&gt;<br>
&gt;<br>
&gt; Below is the data which will enable a MIME compliant mail reader<br>
&gt; implementation to automatically retrieve the ASCII version of the<br>
&gt; Internet-Draft.<br>
&gt;<br>
<br>
_______________________________________________<br>
Sip mailing list &nbsp;https://www1.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 questions on current sip<br>
Use sipping@ietf.org for new developments on the application of sip<br>
</font>
<br>
<br>
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_______________________________________________
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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



From mailnull@www1.ietf.org  Mon Mar 24 18:27:31 2003
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Date: Mon, 24 Mar 2003 15:23:29 -0800 (PST)
From: Amit Bhadoria <bhadoria@cisco.com>
To: James Ford <james_s_ford@hotmail.com>
cc: sip@ietf.org, eronstein@hotmail.com
Subject: Re: [Sip] TLS post connection verification
In-Reply-To: <F15FKQRwgF0Z9hvPlmk00004a2b@hotmail.com>
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answer to these questions depend on whether the connection is intra domain
or inter domain. for an intra domain connection, when proxy is talking to
its clients, then yes it does'nt need to do post connection authentication
unless the clients are capable of it. however, it is recommended that for
this intra domain signalling, digest authentication should be used with
TLS. hence, this digest over TLS presents a pretty secured model.

things are different for inter domain connection though! it makes most
sense to perform mutual authentication (post-connection) for inter domain
signalling. for client side, its straignt forward. however, RFC also
mentions recommendations (section 26.3.2.2) for server side to verify the
certificate for domainname portion of the From header. this makes much
sense, coz interdomain signalling SHOULD be going thru proxies (which
should be able to present certificates), and there should'nt be more than
one interdomain hop involved in a sip session establishement.

as a side note, its highly unlikely (atleast inefficient) to have parsed
the message (for via) while TLS connection is still happening. hence, it
makes more sense to authenticate the certificate against the domain
of source IP addr in the connection instead of IP addr mention in Via
header.

regards,
-amit.
On Mon, 24 Mar 2003, James Ford wrote:

> Hi Eron,
> The standard does not elaborate on that, but here is what I think:
> 
> For the second question: for sending responses on broken connections, I 
> don't think that there is a need to do post connection assertion. for two 
> reasons:
> a - The server does not need to authenticate the client.
> b - allot of applications put their specific IP address on the via field 
> while a certificate is more likely issued to an FQDN.
> 
> The first question is more tricky. My guess is the "incoming" connection 
> should only check for the correctness of the certificate and not do a post 
> connection assertion. This might be a security issue though. Maybe someone 
> with some hands on experience can would be more helpful here.
> 
> Regards,
> James S. Ford
> 
> 
> >Hi,
> >
> >When sending a request (ie REGISTER) to a server I can compare the request
> >URI to the common name (or the alt dns name) in the certificate. If the
> >names match, I can conclude that the certificate is OK.
> >(I'm using OpenSSL, and they recommend this post connection assertion).
> >
> >I have two questions thou:
> >
> >1 - What name should I use for comparison when accepting a connection?
> >Usually only the UAC will demand certificate, I am concerned with te case 
> >of
> >
> >two proxies trying to connect using TLS and the UAS proxy asking for client
> >certificates. (what uri will the UAS proxy has, there is no message yet).
> >
> >2 - how should broken connection be handled? lets say UAC1 sent a request
> >over TLS to UAS1. the handshake went well and the request sent. than for
> >some reason, the connection was broken and UAS1 now needs to reestablish 
> >the
> >
> >connection. What should UAS1 do? use TLS w/out certificates?
> >
> >
> >Regards,
> >Eron Stein
> >
> >_________________________________________________________________
> >The new MSN 8: advanced junk mail protection and 2 months FREE*
> >http://join.msn.com/?page=features/junkmail
> >
> >_______________________________________________
> >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
> 
> 
> _________________________________________________________________
> Add photos to your e-mail with MSN 8. Get 2 months FREE*. 
> http://join.msn.com/?page=features/featuredemail
> 
> _______________________________________________
> 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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as an income tax refund.
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Date: Mon, 24 Mar 2003 19:04:17 -0500
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From: Richard Shockey <richard@shockey.us>
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Subject: [Sip] Eating our own Dog Food...could the IAB and IESG use SIP for
 conference calls
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Like many of us I was moved my Harald's appeal for suggestions for helping 
to cut down costs in the IETF.

I certainly endorse the idea of considering Canada or Mexico as possible 
sites for future IETF meetings, but I suspect that the weekly teleconferece 
calls that the IAB and IESG have represent a significant line item for the 
Secretariat.

In case anyone has not heard, SIP is quite capable of handling this type of 
task and there are a variety of commercial as well as open source Client 
User Agents as well as commercial products and services that could help 
reduce this cost.

I'm sure the SIP working group could help the Secretariat identify products 
and services that could make this essential function more productive and 
operate at less cost.




 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Senior Manager, Strategic Technology Initiatives
NeuStar Inc.
46000 Center Oak Plaza  -   Sterling, VA  20166
Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 815.333.1237
<mailto:richard@shockey.us> or <mailto:richard.shockey@neustar.biz>
  <http://www.neustar.biz> ; <http://www.enum.org>
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

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From mailnull@www1.ietf.org  Mon Mar 24 19:24:51 2003
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From: Adam Roach <adam@dynamicsoft.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Date: Mon, 24 Mar 2003 18:21:51 -0600
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Subject: [Sip] RE: [Simple] [Fwd: RE: SIP WG bugzilla instance]
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I like it. A lot. I would strongly endorse its use by any and all
working groups for tracking bugs in published RFCs and working
group items.

/a

> -----Original Message-----
> From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
> Sent: Wednesday, March 19, 2003 14:56
> To: sip@ietf.org; simple@ietf.org; sipping@ietf.org
> Subject: [Simple] [Fwd: RE: SIP WG bugzilla instance]
> 
> 
> All -
> 
> I apologize now for violating the no-crosspost policy, but
> this spans all the groups. Please, please, reply _ONLY_ to
> sip@ietf.org (which is what should happen if you hit your
> reply button if I've built this message correctly).
> 
> If you have any opinions, positive or negative, about our
> use of bugzilla for issue tracking and whether other groups
> should follow the model, now would be a good time to comment.
> 
> Thanks!
> 
> RjS
> 
> -----Forwarded Message-----
> 
> > From: john.loughney@nokia.com
> > To: rsparks@dynamicsoft.com, wgchairs@ietf.org
> > Subject: RE: SIP WG bugzilla instance
> > Date: 19 Mar 2003 22:46:52 +0200
> > 
> > Robert,
> > 
> > Any comments on how the WG & WG Chairs like it?
> > 
> > John
> > 
> > > At the session on improving working group chair training
> > > (and chair effectiveness), tools came up. The SIP related
> > > working groups have been effectively using a bugzilla variant
> > > to track document issues and help drive open issues to
> > > conclusion.
> > > 
> > > It's available at http://www.sipwg.org (or http://bugs.sipit.net).
> > > 
> > > One point to bring forth at the beginning - we use this as
> > > a tracking tool, not an alternative venue for list discussion.
> > > Document/issue owners are able to make changes to a bug -
> > > people ask the owner to add things to the tracking tool on
> > > the appropriate list. The chairs make sure that the owners
> > > accurately capture threads on the tracking tool.
> > > 
> > > RjS
> > > 
> > > 
> > > 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 
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From mailnull@www1.ietf.org  Tue Mar 25 00:19:52 2003
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From: "Cullen Jennings" <fluffy@cisco.com>
To: "Eron Stein" <eronstein@hotmail.com>, <sip@ietf.org>
Subject: RE: [Sip] TLS post connection verification
Date: Mon, 24 Mar 2003 21:17:59 -0800
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If element A wants to send a SIP message over TLS to a next hop called
foo.com, A needs to check that the SubjectAltName in the certificate
provided over the TLS connection to A matches foo.com. Openssl can not do
this for you automatically, A needs to call the SSL_get_peer_certificate
after A has connected. foo.com may request, and look at, the certificate of
A or may not depending on if mutual TLS is being done or not.

There is very little point in using TLS if you don't check that the server
you connected to is the server you meant to connect to. If all you know is
the IP address of the server and the server presents a certificate with a
FQDN, you are pretty much out of luck.

Cullen


> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Eron
> Stein
> Sent: Wednesday, March 19, 2003 6:14 AM
> To: sip@ietf.org
> Subject: [Sip] TLS post connection verification
>
>
> Hi,
>
> When sending a request (ie REGISTER) to a server I can compare
> the request
> URI to the common name (or the alt dns name) in the certificate. If the
> names match, I can conclude that the certificate is OK.
> (I'm using OpenSSL, and they recommend this post connection assertion).
>
> I have two questions thou:
>
> 1 - What name should I use for comparison when accepting a connection?
> Usually only the UAC will demand certificate, I am concerned with
> te case of
> two proxies trying to connect using TLS and the UAS proxy asking
> for client
> certificates. (what uri will the UAS proxy has, there is no message yet).
>
> 2 - how should broken connection be handled? lets say UAC1 sent a request
> over TLS to UAS1. the handshake went well and the request sent. than for
> some reason, the connection was broken and UAS1 now needs to
> reestablish the
> connection. What should UAS1 do? use TLS w/out certificates?
>
>
> Regards,
> Eron Stein
>
> _________________________________________________________________
> The new MSN 8: advanced junk mail protection and 2 months FREE*
> http://join.msn.com/?page=features/junkmail
>
> _______________________________________________
> 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 mailnull@www1.ietf.org  Tue Mar 25 01:27:28 2003
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To: "Cullen Jennings" <fluffy@cisco.com>
Cc: "Eron Stein" <eronstein@hotmail.com>, <sip@ietf.org>
Subject: Re: [Sip] TLS post connection verification
References: <DLEHICEBMNEIPCACNLPCOEJDCHAA.fluffy@cisco.com>
Reply-To: EKR <ekr@rtfm.com>
From: Eric Rescorla <ekr@rtfm.com>
Date: 24 Mar 2003 22:30:08 -0800
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"Cullen Jennings" <fluffy@cisco.com> writes:

> If element A wants to send a SIP message over TLS to a next hop called
> foo.com, A needs to check that the SubjectAltName in the certificate
> provided over the TLS connection to A matches foo.com. Openssl can not do
> this for you automatically, A needs to call the SSL_get_peer_certificate
> after A has connected. foo.com may request, and look at, the certificate of
> A or may not depending on if mutual TLS is being done or not.
Exactly.

> There is very little point in using TLS if you don't check that the server
> you connected to is the server you meant to connect to. If all you know is
> the IP address of the server and the server presents a certificate with a
> FQDN, you are pretty much out of luck.
Just to be clear, it depends on your threat model. If you believe
that only passive attacks will be mounted, then TLS is worth something
even without checking the certificates. However, active attacks
are getting easier to mount...

-Ekr
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Date: Tue, 25 Mar 2003 08:46:23 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] Eating our own Dog Food...could the IAB and IESG use SIP for
 conference calls
In-reply-to: <5.2.0.9.2.20030324185638.016696c8@popd.ix.netcom.com>
To: "'Richard Shockey'" <richard@shockey.us>, ietf@ietf.org
Cc: sip@ietf.org, Scott Petrack <scott.petrack@edial.com>,
        Christian Stredicke <stredicke@snom.de>
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There are excellent SIP voice conferencing bridges available, such as
from snom AG and eDial. They can be used with various soft clients such
as the Windows Messenger, HotSIP Active Contacts or the Pingtel instant
expressa, or any SIP phone.

I have taken the liberty of copying here the contacts for snom AG and
eDial, in case this will be pursued. I wonder if snom AG or eDial would
just give the IESG and IAB access to their conference servers?

Would there be interest to have access to a WorldCom SIP conference
bridge?

Thanks, Henry

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Richard Shockey
> Sent: Monday, March 24, 2003 6:04 PM
> To: ietf@ietf.org
> Cc: sip@ietf.org
> Subject: [Sip] Eating our own Dog Food...could the IAB and 
> IESG use SIP for conference calls
> 
> 
> 
> Like many of us I was moved my Harald's appeal for 
> suggestions for helping 
> to cut down costs in the IETF.
> 
> I certainly endorse the idea of considering Canada or Mexico 
> as possible 
> sites for future IETF meetings, but I suspect that the weekly 
> teleconferece 
> calls that the IAB and IESG have represent a significant line 
> item for the 
> Secretariat.
> 
> In case anyone has not heard, SIP is quite capable of 
> handling this type of 
> task and there are a variety of commercial as well as open 
> source Client 
> User Agents as well as commercial products and services that 
> could help 
> reduce this cost.
> 
> I'm sure the SIP working group could help the Secretariat 
> identify products 
> and services that could make this essential function more 
> productive and 
> operate at less cost.
> 
> 
> 
> 
>  >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> Richard Shockey, Senior Manager, Strategic Technology 
> Initiatives NeuStar Inc.
> 46000 Center Oak Plaza  -   Sterling, VA  20166
> Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 
> 815.333.1237 > <mailto:richard@shockey.us> or 
> 
<mailto:richard.shockey@neustar.biz>
  <http://www.neustar.biz> ; <http://www.enum.org>
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

_______________________________________________
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



From mailnull@www1.ietf.org  Tue Mar 25 10:03:34 2003
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Date: Tue, 25 Mar 2003 16:01:10 +0100
To: Henry Sinnreich <Henry.Sinnreich@wcom.com>,
        "'Richard Shockey'" <richard@shockey.us>, ietf@ietf.org
From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: RE: [Sip] Eating our own Dog Food...could the IAB and IESG use
  SIP for conference calls
Cc: sip@ietf.org, Scott Petrack <scott.petrack@edial.com>,
        Christian Stredicke <stredicke@snom.de>
In-Reply-To: <000001c2f2dd$4f03d2a0$bc6032a6@hsinnreich2>
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	<mailto:sip-request@ietf.org?subject=subscribe>

That's great idea. We can help with our free SIP server (www.iptel.org/ser/) 
and hosting the services.

-Jiri

At 03:46 PM 3/25/2003, Henry Sinnreich wrote:
>There are excellent SIP voice conferencing bridges available, such as
>from snom AG and eDial. They can be used with various soft clients such
>as the Windows Messenger, HotSIP Active Contacts or the Pingtel instant
>expressa, or any SIP phone.
>
>I have taken the liberty of copying here the contacts for snom AG and
>eDial, in case this will be pursued. I wonder if snom AG or eDial would
>just give the IESG and IAB access to their conference servers?
>
>Would there be interest to have access to a WorldCom SIP conference
>bridge?
>
>Thanks, Henry
>
>> -----Original Message-----
>> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
>> Behalf Of Richard Shockey
>> Sent: Monday, March 24, 2003 6:04 PM
>> To: ietf@ietf.org
>> Cc: sip@ietf.org
>> Subject: [Sip] Eating our own Dog Food...could the IAB and 
>> IESG use SIP for conference calls
>> 
>> 
>> 
>> Like many of us I was moved my Harald's appeal for 
>> suggestions for helping 
>> to cut down costs in the IETF.
>> 
>> I certainly endorse the idea of considering Canada or Mexico 
>> as possible 
>> sites for future IETF meetings, but I suspect that the weekly 
>> teleconferece 
>> calls that the IAB and IESG have represent a significant line 
>> item for the 
>> Secretariat.
>> 
>> In case anyone has not heard, SIP is quite capable of 
>> handling this type of 
>> task and there are a variety of commercial as well as open 
>> source Client 
>> User Agents as well as commercial products and services that 
>> could help 
>> reduce this cost.
>> 
>> I'm sure the SIP working group could help the Secretariat 
>> identify products 
>> and services that could make this essential function more 
>> productive and 
>> operate at less cost.
>> 
>> 
>> 
>> 
>>  >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
>> Richard Shockey, Senior Manager, Strategic Technology 
>> Initiatives NeuStar Inc.
>> 46000 Center Oak Plaza  -   Sterling, VA  20166
>> Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 
>> 815.333.1237 > <mailto:richard@shockey.us> or 
>> 
><mailto:richard.shockey@neustar.biz>
>  <http://www.neustar.biz> ; <http://www.enum.org>
><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
>
>_______________________________________________
>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 

_______________________________________________
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 mailnull@www1.ietf.org  Tue Mar 25 10:40:54 2003
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From: Orit Levin <orit@radvision.com>
To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>,
        "'Richard Shockey'"
	 <richard@shockey.us>, ietf@ietf.org
Cc: sip@ietf.org
Subject: RE: [Sip] Eating our own Dog Food...could the IAB and IESG use SI
	P for conference calls
Date: Tue, 25 Mar 2003 10:36:44 -0500
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	<mailto:sip-request@ietf.org?subject=subscribe>

We can provide a voice and/or VIDEO conferencing service interconnecting all
kinds of voice and video applications in a single conference from/to our NJ
"production lab". That includes SIP voice-only and multimedia (such as
Messenger) end points.

BR,
Orit Levin
Chief Architect
RADVISION
Phone: +1.201.6896330
Video: +1.201.6896430


> -----Original Message-----
> From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> Sent: Tuesday, March 25, 2003 9:46 AM
> To: 'Richard Shockey'; ietf@ietf.org
> Cc: sip@ietf.org; Scott Petrack; Christian Stredicke
> Subject: RE: [Sip] Eating our own Dog Food...could the IAB and IESG use
> SIP for conference calls
> 
> There are excellent SIP voice conferencing bridges available, such as
> from snom AG and eDial. They can be used with various soft clients such
> as the Windows Messenger, HotSIP Active Contacts or the Pingtel instant
> expressa, or any SIP phone.
> 
> I have taken the liberty of copying here the contacts for snom AG and
> eDial, in case this will be pursued. I wonder if snom AG or eDial would
> just give the IESG and IAB access to their conference servers?
> 
> Would there be interest to have access to a WorldCom SIP conference
> bridge?
> 
> Thanks, Henry
> 
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On
> > Behalf Of Richard Shockey
> > Sent: Monday, March 24, 2003 6:04 PM
> > To: ietf@ietf.org
> > Cc: sip@ietf.org
> > Subject: [Sip] Eating our own Dog Food...could the IAB and
> > IESG use SIP for conference calls
> >
> >
> >
> > Like many of us I was moved my Harald's appeal for
> > suggestions for helping
> > to cut down costs in the IETF.
> >
> > I certainly endorse the idea of considering Canada or Mexico
> > as possible
> > sites for future IETF meetings, but I suspect that the weekly
> > teleconferece
> > calls that the IAB and IESG have represent a significant line
> > item for the
> > Secretariat.
> >
> > In case anyone has not heard, SIP is quite capable of
> > handling this type of
> > task and there are a variety of commercial as well as open
> > source Client
> > User Agents as well as commercial products and services that
> > could help
> > reduce this cost.
> >
> > I'm sure the SIP working group could help the Secretariat
> > identify products
> > and services that could make this essential function more
> > productive and
> > operate at less cost.
> >
> >
> >
> >
> >  >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> > Richard Shockey, Senior Manager, Strategic Technology
> > Initiatives NeuStar Inc.
> > 46000 Center Oak Plaza  -   Sterling, VA  20166
> > Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1
> > 815.333.1237 > <mailto:richard@shockey.us> or
> >
> <mailto:richard.shockey@neustar.biz>
>   <http://www.neustar.biz> ; <http://www.enum.org>
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> 
> _______________________________________________
> 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
_______________________________________________
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 mailnull@www1.ietf.org  Tue Mar 25 10:43:18 2003
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	Tue, 25 Mar 2003 08:41:58 -0700 (MST)
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Reply-To: <dan@fsa.ca>
From: "Dan Freedman" <dan@fsa.ca>
To: <ietf@ietf.org>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Eating our own Dog Food...could the IAB and IESG use  SIP for conference calls
Date: Tue, 25 Mar 2003 08:43:59 -0700
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Agreed. To enable access from behind firewalls (ie: homes, hotels,
conferences, airports, Asia, ...), we can donate a B2BUA-based NAT traversal
solution for SIP (same one that Pulver's Free World Dialup uses). It sits
near the proxy, with nothing needed on the client end.

Bottom line: In addition to helping the IESG and IAB, it will help SIP to
have more IETF participants gaining experience with it.

  -Dan





Dan Freedman, CEO, Jasomi Networks, Inc.
2033 Gateway Place, Suite 500, San Jose, CA, 95110
Phone   +1.403.680.2351
Fax     +1.403.269.2993
Email   dan@jasomi.com
SIP     dan@jasomi.com
Web     www.jasomi.com


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Jiri
Kuthan
Sent: Tuesday, March 25, 2003 8:01 AM
To: Henry Sinnreich; 'Richard Shockey'; ietf@ietf.org
Cc: sip@ietf.org; Scott Petrack; Christian Stredicke

That's great idea. We can help with our free SIP server (www.iptel.org/ser/)

and hosting the services.

-Jiri

At 03:46 PM 3/25/2003, Henry Sinnreich wrote:
>There are excellent SIP voice conferencing bridges available, such as
>from snom AG and eDial. They can be used with various soft clients such
>as the Windows Messenger, HotSIP Active Contacts or the Pingtel instant
>expressa, or any SIP phone.
>
>I have taken the liberty of copying here the contacts for snom AG and
>eDial, in case this will be pursued. I wonder if snom AG or eDial would
>just give the IESG and IAB access to their conference servers?
>
>Would there be interest to have access to a WorldCom SIP conference
>bridge?
>
>Thanks, Henry
>
>> -----Original Message-----
>> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
>> Behalf Of Richard Shockey
>> Sent: Monday, March 24, 2003 6:04 PM
>> To: ietf@ietf.org
>> Cc: sip@ietf.org
>> Subject: [Sip] Eating our own Dog Food...could the IAB and 
>> IESG use SIP for conference calls
>> 
>> 
>> 
>> Like many of us I was moved my Harald's appeal for 
>> suggestions for helping 
>> to cut down costs in the IETF.
>> 
>> I certainly endorse the idea of considering Canada or Mexico 
>> as possible 
>> sites for future IETF meetings, but I suspect that the weekly 
>> teleconferece 
>> calls that the IAB and IESG have represent a significant line 
>> item for the 
>> Secretariat.
>> 
>> In case anyone has not heard, SIP is quite capable of 
>> handling this type of 
>> task and there are a variety of commercial as well as open 
>> source Client 
>> User Agents as well as commercial products and services that 
>> could help 
>> reduce this cost.
>> 
>> I'm sure the SIP working group could help the Secretariat 
>> identify products 
>> and services that could make this essential function more 
>> productive and 
>> operate at less cost.
>> 
>> 
>> 
>> 
>>  >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
>> Richard Shockey, Senior Manager, Strategic Technology 
>> Initiatives NeuStar Inc.
>> 46000 Center Oak Plaza  -   Sterling, VA  20166
>> Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 
>> 815.333.1237 > <mailto:richard@shockey.us> or 
>> 
><mailto:richard.shockey@neustar.biz>
>  <http://www.neustar.biz> ; <http://www.enum.org>
><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
>
>_______________________________________________
>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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>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 sip-implementors@cs.columbia.edu for questions on current sip
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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From mailnull@www1.ietf.org  Tue Mar 25 12:53:16 2003
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From: "Valerie Kenneally" <vkenneally@cit.ie>
To: <sip@ietf.org>
Date: Tue, 25 Mar 2003 17:46:05 -0000
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Hi all,

 Assume a User Agent is registering with a SIP Server.  The SIP Server
challenges the User Agent with regard to Authorization.  The User Agent
replies with the appropriate information.  My question-when the User Agent
responds to the challenge will the Call-ID and From tag or this message be
the same as the first REGISTER message sent?  All info greatly accepted!



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From mailnull@www1.ietf.org  Tue Mar 25 14:23:24 2003
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Date: Tue, 25 Mar 2003 20:13:37 +0100
From: Harald Tveit Alvestrand <harald@alvestrand.no>
To: Richard Shockey <richard@shockey.us>, ietf@ietf.org
Cc: sip@ietf.org
Subject: Re: [Sip] Eating our own Dog Food...could the IAB and IESG use SIP
 for conference calls
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Thanks for the idea!
injecting a slight dash of cold water....

- the actual cost of "Teleconferences and Long Distance charges" in 2002 
were USD 82.210 (unaudited) (vs 54.400 for 2001). A significant fraction of 
that, but far from all, is the IESG teleconferences.
- we've already switched teleconference providers once to reduce costs, 
going from call-everyone to most-people-call-in.
- the costliest part of the IESG teleconferences has been the callout: 
international participants (last year, France, the Netherlands, Norway and 
Sweden when they are at home, hotels literally anywhere in the world when 
they are travelling) are called rather than calling in. You don't want to 
discuss with your boss why you had to make a 2 1/2 hour international call 
at hotel room rates if you can avoid it......
- it's absolutely essential that one be able to participate in the IESG 
telecon from just about anything that one can dial from - we've had 
participants on cellphones from trains, in airport lounges, hotel rooms and 
other places. So SIP-only won't work for some time yet.
- even at work, several of us have problems with firewalls; about half the 
IESG uses Jabber during sessions - the reason many of the others don't is 
that they can't get Jabber through their corporate firewalls. So "pure 
Internet SIP" won't work for all of us any time soon.

So I think this is a good idea, PROVIDED THAT:

- The SIP teleconference bridge provider is able to provide either 
800-number access or callout services to normal telephones in most corners 
of the world
- The voice quality, operator quality and call stability is competitive 
(yes, we've got the "there's an echo on this conference - can you figure 
out who is echoing and fix it" request down to a matter of routine)

A "normal" teleconference provider that *also* allows SIP dialin over the 
Internet would probably be perfect. If you have one - send email to me - 
PRIVATELY - and I will forward to the relevant parties at the secretariat.

               Harald

--On mandag, mars 24, 2003 19:04:17 -0500 Richard Shockey 
<richard@shockey.us> wrote:

>
> Like many of us I was moved my Harald's appeal for suggestions for
> helping to cut down costs in the IETF.
>
> I certainly endorse the idea of considering Canada or Mexico as possible
> sites for future IETF meetings, but I suspect that the weekly
> teleconferece calls that the IAB and IESG have represent a significant
> line item for the Secretariat.
>
> In case anyone has not heard, SIP is quite capable of handling this type
> of task and there are a variety of commercial as well as open source
> Client User Agents as well as commercial products and services that could
> help reduce this cost.
>
> I'm sure the SIP working group could help the Secretariat identify
> products and services that could make this essential function more
> productive and operate at less cost.


_______________________________________________
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 mailnull@www1.ietf.org  Tue Mar 25 17:47:30 2003
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Date: Tue, 25 Mar 2003 16:36:14 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] Eating our own Dog Food...could the IAB and IESG use SIP for
 conference calls
In-reply-to: <711640000.1048619617@askvoll.hjemme.alvestrand.no>
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        "'Richard Shockey'" <richard@shockey.us>, ietf@ietf.org
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>So "pure Internet SIP" won't work for all of us any time soon.

Glad to clear up the confusion on this point. People on the PSTN can
dial in and can be called from the SIP conferencing server by using a
service provider that has standard PSTN-SIP gateways. The typical SIP
voice conference has both PSTN users and SIP users. Works quite well for
everybody.

IM can also be used at the same time, by those who prefer it to real
time voice. Several SIP clients do both, actually video and data as
well.

Thanks, Henry

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Harald Tveit Alvestrand
> Sent: Tuesday, March 25, 2003 1:14 PM
> To: Richard Shockey; ietf@ietf.org
> Cc: sip@ietf.org
> Subject: Re: [Sip] Eating our own Dog Food...could the IAB 
> and IESG use SIP for conference calls
> 
> 
> Thanks for the idea!
> injecting a slight dash of cold water....
> 
> - the actual cost of "Teleconferences and Long Distance 
> charges" in 2002 
> were USD 82.210 (unaudited) (vs 54.400 for 2001). A 
> significant fraction of 
> that, but far from all, is the IESG teleconferences.
> - we've already switched teleconference providers once to 
> reduce costs, 
> going from call-everyone to most-people-call-in.
> - the costliest part of the IESG teleconferences has been the 
> callout: 
> international participants (last year, France, the 
> Netherlands, Norway and 
> Sweden when they are at home, hotels literally anywhere in 
> the world when 
> they are travelling) are called rather than calling in. You 
> don't want to 
> discuss with your boss why you had to make a 2 1/2 hour 
> international call 
> at hotel room rates if you can avoid it......
> - it's absolutely essential that one be able to participate 
> in the IESG 
> telecon from just about anything that one can dial from - we've had 
> participants on cellphones from trains, in airport lounges, 
> hotel rooms and 
> other places. So SIP-only won't work for some time yet.
> - even at work, several of us have problems with firewalls; 
> about half the 
> IESG uses Jabber during sessions - the reason many of the 
> others don't is 
> that they can't get Jabber through their corporate firewalls. 
> So "pure 
> Internet SIP" won't work for all of us any time soon.
> 
> So I think this is a good idea, PROVIDED THAT:
> 
> - The SIP teleconference bridge provider is able to provide either 
> 800-number access or callout services to normal telephones in 
> most corners 
> of the world
> - The voice quality, operator quality and call stability is 
> competitive 
> (yes, we've got the "there's an echo on this conference - can 
> you figure 
> out who is echoing and fix it" request down to a matter of routine)
> 
> A "normal" teleconference provider that *also* allows SIP 
> dialin over the 
> Internet would probably be perfect. If you have one - send 
> email to me - 
> PRIVATELY - and I will forward to the relevant parties at the 
> secretariat.
> 
>                Harald
> 
> --On mandag, mars 24, 2003 19:04:17 -0500 Richard Shockey 
> <richard@shockey.us> wrote:
> 
> >
> > Like many of us I was moved my Harald's appeal for suggestions for 
> > helping to cut down costs in the IETF.
> >
> > I certainly endorse the idea of considering Canada or Mexico as 
> > possible sites for future IETF meetings, but I suspect that 
> the weekly 
> > teleconferece calls that the IAB and IESG have represent a 
> significant 
> > line item for the Secretariat.
> >
> > In case anyone has not heard, SIP is quite capable of handling this 
> > type of task and there are a variety of commercial as well as open 
> > source Client User Agents as well as commercial products 
> and services 
> > that could help reduce this cost.
> >
> > I'm sure the SIP working group could help the Secretariat identify 
> > products and services that could make this essential function more 
> > productive and operate at less cost.
> 
> 
> _______________________________________________
> 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
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Thread-Topic: [Sip] Eating our own Dog Food...could the IAB and IESG use SIP for conference calls
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Subject: [Sip] Question on RFC3329
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RFC3329, section 2.3.1 says:

" The server MUST check that the security mechanisms listed in the
   Security-Verify header field of incoming requests correspond to its
   static list of supported security mechanisms.", and further down:

"   The server can proceed processing a particular request if, and only
   if, the list was not modified.  If modification of the list is
   detected, the server MUST respond to the client with a 494 (Security
   Agreement Required) response."

In case of Digest this makes sense. But in case the agreed security mechanism is IPSec or TLS, then the SIP layer will only receive the request if it was sent according to the previously agreed security rules (otherwise the lower layers discard it). So, even in case the Security Verify header is altered (e.g. mistakenly at insertion by the UA), the request passed the security checks. Then why the server must respond with 494 instead of accepting it?

Shouldn't that MUST be a SHOULD instead, as the server knows how reliable the security mechanism they agreed is without the Security-Verify content? 

/Gabor
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From: Jari Arkko <jari.arkko@piuha.net>
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To: Gabor.Bajko@nokia.com
Cc: sip@ietf.org
Subject: Re: [Sip] Question on RFC3329
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Gabor.Bajko@nokia.com wrote:
> RFC3329, section 2.3.1 says:
> 
> " The server MUST check that the security mechanisms listed in the
>    Security-Verify header field of incoming requests correspond to its
>    static list of supported security mechanisms.", and further down:
> 
> "   The server can proceed processing a particular request if, and only
>    if, the list was not modified.  If modification of the list is
>    detected, the server MUST respond to the client with a 494 (Security
>    Agreement Required) response."
> 
> In case of Digest this makes sense. But in case the agreed security mechanism is IPSec or TLS, then the SIP layer will only receive the request if it was sent according to the previously agreed security rules (otherwise the lower layers discard it). So, even in case the Security Verify header is altered (e.g. mistakenly at insertion by the UA), the request passed the security checks. Then why the server must respond with 494 instead of accepting it?
> 
> Shouldn't that MUST be a SHOULD instead, as the server knows how reliable the security mechanism they agreed is without the Security-Verify content? 

We discussed this while the draft was being worked on. In principle
you are right, but we felt that it would be better to make the
behaviour the same in all cases without special cases.

Jari

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Hi,
When a single phone is used by several users, do we refer to each user
as a seperate UA or do we refer to the phone as the UA?
Thanks,
James.




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Hi 

Does anybody have any recommendations for a small and cheap SIP-PSTN
gateway? 
 
/P

--- 
Ps. I am Blogger! http://www.parnes.com/blog/
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From: Tiefengraber Roland <Roland.Tiefengraber@atc.co.at>
To: "'Peter Parnes'" <peppar@cdt.luth.se>, sip@ietf.org
Subject: AW: [Sip] PSTN gateway
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Hi

you can take the

Aphona StarLine SIP-PSTN Gateway
StarLine 200 Basic 19inch Gateway: approx.2700 $
E1/LAN extension card (30 PRI channels): approx.2600 $
Up to 8 extension cards for each chassis

Roland

>Hi 
>
>Does anybody have any recommendations for a small and cheap SIP-PSTN
>gateway? 

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Subject: RE: [Sip] PSTN gateway
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
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        "Peter Parnes" <peppar@cdt.luth.se>, <sip@ietf.org>
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I do not think that this type of questions and answers are appropriate for an IETF WG list. 

Dan


> -----Original Message-----
> From: Tiefengraber Roland [mailto:Roland.Tiefengraber@atc.co.at]
> Sent: Thursday, March 27, 2003 2:33 PM
> To: 'Peter Parnes'; sip@ietf.org
> Subject: AW: [Sip] PSTN gateway
> 
> 
> Hi
> 
> you can take the

.....

> 
> >
> >Does anybody have any recommendations for a small and cheap SIP-PSTN
> >gateway? 
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From: Robert Sparks <rsparks@dynamicsoft.com>
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Subject: [Sip] [Fwd: [Simple] Call for availability: Interim meeting]
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All -

Ted Hardie (our new Apps Area AD) asked that I repeat this
request on SIP, SIPPING, and IMPP.

If you haven't already responded and plan to attend this
SIMPLE WG interim meeting, please send a private message
to me as soon as possible.

RjS

-----Forwarded Message-----

> From: Robert Sparks <rsparks@dynamicsoft.com>
> To: simple@ietf.org
> Subject: [Simple] Call for availability: Interim meeting
> Date: 26 Mar 2003 08:50:08 -0600
> 
> Folks -
> 
> As mentioned at our meeting in IETF56 we are going to hold an
> interim meeting between now and IETF57. We're planning
> a two day session, but need to choose the actual dates. 
> Location will depend on the dates, but will probably be in
> eastern Canada.
> 
> If, and _only_ if, you are definitely planning to attend
> please reply to me PRIVATELY by Friday with your availability 
> between May 12 and June 13. We'll find a date that accomodates as
> many responders as possible. 
> 
> Thanks,
> 
> RjS
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple

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CC: Tiefengraber Roland <Roland.Tiefengraber@atc.co.at>, sip@ietf.org
Subject: Re: [Sip] PSTN gateway
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> I do not think that this type of questions and answers are appropriate 
> for an IETF WG list. 

Oh, I am very sorry that I misunderstood the purpose of this list. I
thought discussing deployment and interoperability issues was under the
IETF hat. 

Any suggestions on where such discussions should take place? 


/P
--- 
Ps. I am Blogger! http://www.parnes.com/blog/
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From mailnull@www1.ietf.org  Thu Mar 27 14:14:14 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'James Ford'" <james_s_ford@hotmail.com>, sip@ietf.org
Subject: RE: [Sip] The term UA
Date: Thu, 27 Mar 2003 14:08:54 -0500
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That depends on how the phone is implemented.  

The difference is pretty subtle, and may not matter to anyone.

If you can't tell from the outside if there is more than one
UA in the physical phone then I suppose the phone is the UA.
I think most phones are implemented that way.  Multiple
users are just separate registrations with different AoRs.

You could also say that there is a UA instance for each
active call on a device if you want to.

Brian

> -----Original Message-----
> From: James Ford [mailto:james_s_ford@hotmail.com]
> Sent: Wednesday, March 26, 2003 8:53 AM
> To: sip@ietf.org
> Subject: [Sip] The term UA
> 
> 
> 
> Hi,
> When a single phone is used by several users, do we refer to each user
> as a seperate UA or do we refer to the phone as the UA?
> Thanks,
> James.
> 
> 
> 
> 
> _________________________________________________________________
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> 
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To: Harald Tveit Alvestrand <harald@alvestrand.no>
From: Jiri Kuthan <jiri@iptel.org>
Subject: Re: [Sip] Eating our own Dog Food...could the IAB and IESG use
  SIP for conference calls
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At 08:13 PM 3/25/2003, Harald Tveit Alvestrand wrote:
>A "normal" teleconference provider that *also* allows SIP dialin over the Internet would probably be perfect. If you have one - send email to me - PRIVATELY - and I will forward to the relevant parties at the secretariat.

As said, we can provide tech-support or hosting a SIP server (iptel.org/ser) --
the server itself is free. I think I could get a conferencing server from
a partner of ours. The missing piece then is PSTN connectivity, both dial-in
for out-of-IP callers and callees. We can't unfortunately afford sponsoring that
-- we are non-profit (on purpose ;)). Maybe approaching companies like deltathree 
would make sense?

-Jiri  

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Subject: [Sip] SIP Client Transaction
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Hi,
  RFC 3261 section "17.1.1.1 Overview of INVITE Transaction" specifies the
state machine for Client INVITE transaction. In the diagram if client
transaction state comes to proceeding after receiving 1xx message, no timers
are started. It is not specified that if no message is  received further in
this state, how this transaction will be terminated. There is no timer
specified to clear this transaction. Do we need to start timer B again in
this state and clear the transaction after timer B expiry?
Your feedback is appreciated.
Thanks,
Vijay
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From mailnull@www1.ietf.org  Fri Mar 28 17:30:51 2003
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From: murali <muraliv@bharatmail.com>
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Subject: [Sip] how do you segregate host from token?
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Hi,

In the field generic-param, its mentioned that generic-param = token[EQUAL gen-value}

now gen-value = token/host/quoted-string.
If host is an IPV4 address or IPV6 address, its fine. But, if host is a hostname which in turn is defined as 
hostname = *(domainlabel".")toplabel [.]
token also has the same allowable characters as allowed in domainname.

So, if I am writing a parser for gen-value, when do I recognize it as a token and when do I recognise it as a host with a hostname which comprises of a domainlabel followed by a toplabel?

Is there any particular character or pattern which distinguishes the hostname from the token?

Regards
Murali.

-------
Murali Voleti,
#503,Maheswari complex,
Masab Tank,
Hyderabad
Tele:(040) 6502272 ext:211

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From: "Christopher Allen" <ChristopherA@AlacrityManagement.com>
To: <ietf@ietf.org>, "Richard Shockey" <richard@shockey.us>
Cc: <sip@ietf.org>, "Bill Woodcock" <woody@pch.net>
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Subject: [Sip] Re: Eating our own Dog Food...could the IAB and IESG use SIP for  conference calls
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Richard Shockey <richard@shockey.us> wrote:
> Like many of us I was moved my Harald's appeal for suggestions for
> helping to cut down costs in the IETF.
>
> I certainly endorse the idea of considering Canada or Mexico as
> possible sites for future IETF meetings, but I suspect that the
> weekly teleconferece calls that the IAB and IESG have represent a
> significant line item for the Secretariat.
>
> In case anyone has not heard, SIP is quite capable of handling this
> type of task and there are a variety of commercial as well as open
> source Client User Agents as well as commercial products and services
> that could help reduce this cost.
>
> I'm sure the SIP working group could help the Secretariat identify
> products and services that could make this essential function more
> productive and operate at less cost.

Bill Woodock of Packet Clearing House <woody@pch.net> has set up a SIP network
where you dial an ASN (and optionally an extension) that will ring various NOCs.
Currently they are using Cisco phones, but there is no reason why software SIP
based phones shouldn't work as well.

http://www.pch.net/inoc-dba/

~-~
INTER-NOC HOTLINE PHONE SYSTEM
Packet Clearing House operates the Inter-Network Operations Center Dial-By-ASN
(INOC-DBA) hotline phone system, a global voice telephony network that connects
the network operations centers and security incident response teams of critical
Internet infrastructure providers such as backbone carriers, Internet service
providers, and Internet exchanges as well as critical individuals within the
policy, regulatory, Internet governance, security and vendor communities. The
INOC-DBA is a closed system, ensuring secure and authenticated communications,
and uses a combination of highly redundant directory services and direct
peer-to-peer communications between stations to create a resilient,
high-survivability network. It carries both routine operational traffic and
emergency-response traffic.
~-~

It may make sense to have the IESG members to have access to this emergency
network in general, besides for its use in IESG business.

I don't believe at this point that they have conferencing services available,
only point-to-point calls, but I'm fairly certain from some discussions that
I've had with some SIP conferencing tool vendors that we should be able to get
them to donate those services as well. We might even be able to get Cisco to
donate some phones.

-- Christopher Allen

----------------------------------------------------------------------
.. Christopher Allen                            Alacrity Management ..
.. <ChristopherA at AlacrityManagement.com>   1563 Solano Ave. #353 ..
.. o510/649-4030  f510/649-4034                  Berkeley, CA 94707 ..


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Date: Tue, 25 Mar 2003 08:33:30 -0800 (PST)
From: Bill Woodcock <woody@pch.net>
To: Christopher Allen <ChristopherA@AlacrityManagement.com>
cc: ietf@ietf.org, Richard Shockey <richard@shockey.us>, <sip@ietf.org>
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Subject: [Sip] Re: Eating our own Dog Food...could the IAB and IESG use SIP for
 conference calls
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    > Currently they are using Cisco phones, but there is no reason why software SIP
    > based phones shouldn't work as well.

Actually, only about three-quarters of the phones are Ciscos, to the best
of my knowledge; meaning that there are more than a hundred non-Cisco
(mostly open-source software) phones on the network as well.

A lot of the IAB and IESG folks are already on the INOC network.  Should
we put some effort into getting the remainder (and newly-appointed) folks
on as well?

                                -Bill


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From: "Scott Petrack" <scott.petrack@edial.com>
To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>,
        "'Richard Shockey'" <richard@shockey.us>, <ietf@ietf.org>
Cc: <sip@ietf.org>, "Scott Petrack" <scott.petrack@edial.com>,
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        "Ben Teitelbaum" <ben@internet2.edu>
Subject: RE: [Sip] Eating our own Dog Food...could the IAB and IESG use SIP for conference calls
Date: Tue, 25 Mar 2003 12:31:05 -0500
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eDial can definitely help here, and in the longer term perhaps Internet2
can help:

1. I just made a URL and a dial-in access code for a reservationless
conference which can be dialed in from a PSTN phone or directly from a
SIP phone. (The URL allows people to join the conference call without
punching in any access codes). Since Henry's letter went out to two very
large mailing lists, I'll just ask for the duly authorized person who
cares about IAB and IESG conferences to give me a call or write me and
I'll give you the info.

2. Internet2 recently bought an eDial server and I imagine that as part
of their charter they can help the IAB/IESG with these sorts of
services. I have cc-ed Ben Teitelbaum of internet2.edu on this letter.

I could provide some conferencing in the short term, but that Internet2
would be a very appropriate way to host such conferences going forward.
Since the eDial boxes do mixed SIP and PSTN conferences, it might make
for a smooth transition for people who don't believe in VoIP.....(I
remember Steve Deering once ranting at me about this).

Scott

Scott Petrack, CTO
eDial, Inc.
mobile: +1-617-592-8688
email: scott.petrack@edial.com
IM: scott.petrack@edial.com (MSN) and sbpetrack (AOL)
find me: http://beta.edial.com/call/scott.petrack


> -----Original Message-----
> From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> Sent: Tuesday, March 25, 2003 9:46 AM
> To: 'Richard Shockey'; ietf@ietf.org
> Cc: sip@ietf.org; Scott Petrack; Christian Stredicke
> Subject: RE: [Sip] Eating our own Dog Food...could the IAB and IESG
use
> SIP for conference calls
> 
> There are excellent SIP voice conferencing bridges available, such as
> from snom AG and eDial. They can be used with various soft clients
such
> as the Windows Messenger, HotSIP Active Contacts or the Pingtel
instant
> expressa, or any SIP phone.
> 
> I have taken the liberty of copying here the contacts for snom AG and
> eDial, in case this will be pursued. I wonder if snom AG or eDial
would
> just give the IESG and IAB access to their conference servers?
> 
> Would there be interest to have access to a WorldCom SIP conference
> bridge?
> 
> Thanks, Henry
> 
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On
> > Behalf Of Richard Shockey
> > Sent: Monday, March 24, 2003 6:04 PM
> > To: ietf@ietf.org
> > Cc: sip@ietf.org
> > Subject: [Sip] Eating our own Dog Food...could the IAB and
> > IESG use SIP for conference calls
> >
> >
> >
> > Like many of us I was moved my Harald's appeal for
> > suggestions for helping
> > to cut down costs in the IETF.
> >
> > I certainly endorse the idea of considering Canada or Mexico
> > as possible
> > sites for future IETF meetings, but I suspect that the weekly
> > teleconferece
> > calls that the IAB and IESG have represent a significant line
> > item for the
> > Secretariat.
> >
> > In case anyone has not heard, SIP is quite capable of
> > handling this type of
> > task and there are a variety of commercial as well as open
> > source Client
> > User Agents as well as commercial products and services that
> > could help
> > reduce this cost.
> >
> > I'm sure the SIP working group could help the Secretariat
> > identify products
> > and services that could make this essential function more
> > productive and
> > operate at less cost.
> >
> >
> >
> >
> >  >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> > Richard Shockey, Senior Manager, Strategic Technology
> > Initiatives NeuStar Inc.
> > 46000 Center Oak Plaza  -   Sterling, VA  20166
> > Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1
> > 815.333.1237 > <mailto:richard@shockey.us> or
> >
> <mailto:richard.shockey@neustar.biz>
>   <http://www.neustar.biz> ; <http://www.enum.org>
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> 
> _______________________________________________
> 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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On Mon, 24 Mar 2003, Richard Shockey wrote:

> I certainly endorse the idea of considering Canada or Mexico as possible
> sites for future IETF meetings, but I suspect that the weekly
> teleconferece calls that the IAB and IESG have represent a significant
> line item for the Secretariat.
>
> In case anyone has not heard, SIP is quite capable of handling this type
> of task and there are a variety of commercial as well as open source
> Client User Agents as well as commercial products and services that
> could help reduce this cost.

Are there existence proofs of IETF-style WGs conferencing via SIP these
days?  I don't know anything about these services, but it seems to me that
to be practical, a SIP-capable conf call service would also have to
support plain-old telephony callers as well.

 - RL "Bob"


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Subject: [Sip] Non-Invite adhoc minutes and action plan
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--=-Kw37/HfBI5hw03WNxTHW
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Here are the minutes from the ad-hoc we held last Tuesday
on the Non-Invite problem. Many thanks to Cullen Jennings for 
being the minute taker.

I will be taking the output of this discussion and revising
the non-invite draft reflecting the following input:

- Separate the analysis of problems due to slow responding
  elements from the analysis of problems due to non-responding
  elements.

- Explore mechanisms to let the next hop better estimate how much 
  time it has to complete.

If you have other input to contribute after you look over these
notes, please send it to the list soon. I will begin revising
the non-invite draft within the week.

Thanks to everyone for the thought they've put into this so far.

RjS

--=-Kw37/HfBI5hw03WNxTHW
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Folks:
Adam Roach; Ben Campbell; Cullen Jennings; Daniel Petrie; Dean Willis; Gonzalo 
Camarillo; Jonathan Rosenberg; Robert Sparks

Conclusion :
Identified we need to tear this apart addressing the problem of having to take a long time 
to generate a response separately from the problem of something that was going to 
respond went away.  

Summary:
The "Random Notes" below are the stuff I typed in the meeting. Right here I am going to 
describe what I took out of the meeting - this could be a totally warped impression of 
what happened - I was pretty tired. So read the random notes too and make your own 
summary. 

The problem we are trying to solve is how to clean up state on all the elements. The 
major two cases we need to deal with are 1) something died and will never respond 2) 
something is taking a long time to respond. Case 2 splits into two sub cases. The first 2a 
where it is a automata that is holding up the response and 2b where something is waiting 
for input from a human before responding. 

Some things to keep in mind when evaluating solutions are: systems use different values 
of T1. DNS SRV failures complicate things. A similar problem might be able to happen 
in INVITE transactions where things fork or CANCEL. 

We brainstormed a few possible approaches to this problem. 
-	Set an expire time of some sort on the transaction
 o	If this is absolute, then need synchronized time - Yuck
 o	If we ignore the network time (the hops) and only count the processing 
time at each node (the hipitty) then absolute time would not be needed
 o	Make the time start at 64*T1 for the first node and shrink as the message traverses 
nodes 
 o	won't work
-	Make the time at each node be large enough (i.e. current max hops * 64 * T1) that 
downstream nodes will timeout first. 
 o	64*T1*70 = 37 minutes * 1000 transaction per second is a lot of state 
-	Have any transaction that is going to take awhile, return some code that more or 
less amounts to try back later with the same request and I will give you the answer 
then. 
 o	Not clear how this work when nodes fail instead of just being slow

The reason I came to the meeting was to say that no node should have to hold state 
forever and that it is reasonable to put some upper bound on how long things are allowed 
to take and still be a single transaction. I don't think the problem is solvable without this 
constraint. 



Random Notes:

First question - should we kill proposal B?

Considered  how HTTP deals with the problem. Feeling was HTTP did not deal 
with it yet but would need  to in the future with SOAP. 

Decided that right now we don't have enough information to kill proposal B - 
need to brainstorm some options. 

Jonathan posed the problem as: Imagine the timeout is very long then figure out how to 
clean up state. There is state in user agent, proxies. 

Idea:  allow the proxy to go stateless at some point. 

Consider invite stuff. Proxy may have to maintain state arbitrarily long. Solution 
was to have uas send provisional to refresh timers. Not sure this was true. 

If we make things arbitrarily long, how deal with crashed things...

The ideas of shrinking the timer: The uac will have timer 70*64*t1. Each proxy have 
max hops * 64 * t1. uas has timer 64 *t1 or could possibly max hops *54 *t1.  This 
results in keeping state 37 minutes - all agree 37 minutes does not sound good. 

Invite has this same problem if the cancel fails you can get this cascaded.

Really two problem here - one is failure - other is device is just slow doing it's backend 
processing or something. 

Important point - not all systems have t1 at 500 ms - this can mess up things and we 
have to take it into account. 

The DNS SRV fail over needs to be considered.

Disused the idea of saying you have this much time to process this message. It can get 
decremented along the way. 

Is the whole discussion purely that the timer number is too low. 

If we want to deal with human interaction,  then it is a different problem that time for 
automata to respond. 

If the UAS timer is large enough, then the only time that there is no response is when 
something failed. 

One proposal is make timer bigger than t1*64 and then toss out the 408 somehow. 

Do we let the client continues to allow the transaction to processed using some sort of 
message back or  do we redo the request at some future time when the far side as 
completed the answer?


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From: Deuringer Wolfgang <wolfgang.deuringer@siemens.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Date: Thu, 27 Mar 2003 13:18:15 +0100
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Subject: [Sip] RFC 3325 - Private Extensions to SIP for Asserted Identity within
 Trusted Networks
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Dear all,

the RFC3325 states in Chapter 11.2 Authentication requirements that "Users must be 
authenticated using SIP Digest Authentication". This contradicts the statement in the SIP discussion
list that the P-Asserted-Identity should be available in response messages, too. However I see no
way to authenticate response messages with SIP mechanisms. Additionally the RFC shows the usage 
of these headers only for Requests.

If there is a common understanding that the P-Asserted-Identity header is required for responses,
will there be an update of the RFC?
*	What happens in a case where a proxy receives a response from a node he does not trust? He cannot
authenticate the originator of the response. This means that he can only set up the P-Asserted-Identity
for responsens of users which are stored within the database which is accessible by the proxy.
*	Does this mean that response messages from users not available in the database would be skipped by the
proxy. 
*	Which responses should contain this P-Asserted-Identity Header? Only final responses or even provisional ones? 
 

Best regards,

	Wolfgang Deuringer


____________________________________________________
Wolfgang Deuringer
ICN CP D NA D12
Mch H/Sc8
Tel.  +49 89 722 28236
Fax  + 49 89 722 48697



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From: Y01317@njupt.edu.cn
To: Vijay Gaur <vgaur@aastra.com>
Cc: sip@ietf.org
Date: Sun, 30 Mar 2003 14:28:10 GMT
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Subject: [Sip] Re: SIP Client Transaction
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Vijay Gaur writes: 

> Hi,
>   RFC 3261 section "17.1.1.1 Overview of INVITE Transaction" specifies the
> state machine for Client INVITE transaction. In the diagram if client
> transaction state comes to proceeding after receiving 1xx message, no timers
> are started. It is not specified that if no message is  received further in
> this state, how this transaction will be terminated. There is no timer
> specified to clear this transaction. Do we need to start timer B again in
> this state and clear the transaction after timer B expiry?
> Your feedback is appreciated.
> Thanks,
> Vijay
> _______________________________________________
> 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

Timer C can do this thing. If Timer C fires, the transaction will be killed. 

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Subject: [Sip] Event header in responses
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Thread-Topic: [Sip] Event header in responses
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Hi

Bug 677 (http://www.sipwg.org/sipwg/show_bug.cgi?id=677) says that there is confilicting text in RFC 3265 about Event header in responses. The wording of the bug is not 100% clear, but I assume that responses should not contain an Event header. The reason for my question is that SIP Service Examples-04 and shows no Event in responses, presence-10 call flow example shows it, event-list-01 draft shows it sometimes. What is the correct interpretation?

		Jussi
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From: "Patrick Lam" <patrickl@astri.org>
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Subject: [Sip] Difference between dialog and session in SIP?
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Dear all:

I am a little confused about the concept of a dialog.  What exactly is the
difference between a dialog and a session?  Does dialog mean a "call has
already been setup" in the signaling sense, while a session mean the "media
has already been setup" in the media sense?

Also, what does "part of a dialog" mean?  Are "INVITE", "ACK" and the
responses considered "part of a dialog"?

What does "outside the dialog" mean then?  Is a request is outside the
dialog, does it still affect or change anything "inside the dialog" then?

Thanks very much in advance,

Regards,

Patrick

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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Deuringer Wolfgang'" <wolfgang.deuringer@siemens.com>,
        "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] RFC 3325 - Private Extensions to SIP for Asserted Ident
	ity within Trusted Networks
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Wolfgang,

Section 11 of the RFC is just an example.

The point of the mechanism in this RFC is that P-Asserted-Identity can only
be used within a 'trusted domain', but that to make this concept meaningful,
you must provide concrete details of the requirements that are met by all
elements of the trusted domain. This is called Spec(T), and section 11 gives
an example.

Nevertheless, this statement simply says that users must be authenticated
using SIP Digest Authentication. I cannot see what relevance this has to
whether P-Asserted-Identity may be used in responses.

The handling of responses from a node outside the Trust Domain is a sub-case
of the handling of messages described in section 5:

  "A proxy in a Trust Domain can receive a message from a node that it
   trusts, or a node that it does not trust.  When a proxy receives a
   message from a node it does not trust and it wishes to add a P-
   Asserted-Identity header field, the proxy MUST authenticate the
   originator of the message, and use the identity which results from
   this authentication to insert a P-Asserted-Identity header field into
   the message."

This does not specify how or when the originator of the message is
authenticated. If you were using the Spec(T) specified in Section 11, then
Digest is the only method of authentication specified, so clearly this
cannot be carried out at the point at which the response is received. The
proxy would need some means to ascertain that the message (response) came
from the same user it had previously authenticated with Digest (such means
being a form of authentication in themselves).

This 'some means' ought to be specified in Spec(T), but as Section 11 is
just an example, I don't think it is necessary to update the RFC.

Regards,

Mar

> -----Original Message-----
> From: Deuringer Wolfgang [mailto:wolfgang.deuringer@siemens.com]
> Sent: 27 March 2003 12:18
> To: 'sip@ietf.org'
> Subject: [Sip] RFC 3325 - Private Extensions to SIP for Asserted
> Identity within Trusted Networks
> 
> 
> 
> Dear all,
> 
> the RFC3325 states in Chapter 11.2 Authentication 
> requirements that "Users must be 
> authenticated using SIP Digest Authentication". This 
> contradicts the statement in the SIP discussion
> list that the P-Asserted-Identity should be available in 
> response messages, too. However I see no
> way to authenticate response messages with SIP mechanisms. 
> Additionally the RFC shows the usage 
> of these headers only for Requests.
> 
> If there is a common understanding that the 
> P-Asserted-Identity header is required for responses,
> will there be an update of the RFC?
> *	What happens in a case where a proxy receives a 
> response from a node he does not trust? He cannot
> authenticate the originator of the response. This means that 
> he can only set up the P-Asserted-Identity
> for responsens of users which are stored within the database 
> which is accessible by the proxy.
> *	Does this mean that response messages from users not 
> available in the database would be skipped by the
> proxy. 
> *	Which responses should contain this P-Asserted-Identity 
> Header? Only final responses or even provisional ones? 
>  
> 
> Best regards,
> 
> 	Wolfgang Deuringer
> 
> 
> ____________________________________________________
> Wolfgang Deuringer
> ICN CP D NA D12
> Mch H/Sc8
> Tel.  +49 89 722 28236
> Fax  + 49 89 722 48697
> 
> 
> 
> _______________________________________________
> 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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<TITLE>RE: [Sip] RFC 3325 - Private Extensions to SIP for Asserted =
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<BODY>

<P><FONT SIZE=3D2>Wolfgang,</FONT>
</P>

<P><FONT SIZE=3D2>Section 11 of the RFC is just an example.</FONT>
</P>

<P><FONT SIZE=3D2>The point of the mechanism in this RFC is that =
P-Asserted-Identity can only be used within a 'trusted domain', but =
that to make this concept meaningful, you must provide concrete details =
of the requirements that are met by all elements of the trusted domain. =
This is called Spec(T), and section 11 gives an example.</FONT></P>

<P><FONT SIZE=3D2>Nevertheless, this statement simply says that users =
must be authenticated using SIP Digest Authentication. I cannot see =
what relevance this has to whether P-Asserted-Identity may be used in =
responses.</FONT></P>

<P><FONT SIZE=3D2>The handling of responses from a node outside the =
Trust Domain is a sub-case of the handling of messages described in =
section 5:</FONT></P>

<P><FONT SIZE=3D2>&nbsp; &quot;A proxy in a Trust Domain can receive a =
message from a node that it</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; trusts, or a node that it does not =
trust.&nbsp; When a proxy receives a</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; message from a node it does not trust =
and it wishes to add a P-</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; Asserted-Identity header field, the =
proxy MUST authenticate the</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; originator of the message, and use the =
identity which results from</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; this authentication to insert a =
P-Asserted-Identity header field into</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; the message.&quot;</FONT>
</P>

<P><FONT SIZE=3D2>This does not specify how or when the originator of =
the message is authenticated. If you were using the Spec(T) specified =
in Section 11, then Digest is the only method of authentication =
specified, so clearly this cannot be carried out at the point at which =
the response is received. The proxy would need some means to ascertain =
that the message (response) came from the same user it had previously =
authenticated with Digest (such means being a form of authentication in =
themselves).</FONT></P>

<P><FONT SIZE=3D2>This 'some means' ought to be specified in Spec(T), =
but as Section 11 is just an example, I don't think it is necessary to =
update the RFC.</FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Mar</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Deuringer Wolfgang [<A =
HREF=3D"mailto:wolfgang.deuringer@siemens.com">mailto:wolfgang.deuringer=
@siemens.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 27 March 2003 12:18</FONT>
<BR><FONT SIZE=3D2>&gt; To: 'sip@ietf.org'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: [Sip] RFC 3325 - Private Extensions to =
SIP for Asserted</FONT>
<BR><FONT SIZE=3D2>&gt; Identity within Trusted Networks</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Dear all,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; the RFC3325 states in Chapter 11.2 =
Authentication </FONT>
<BR><FONT SIZE=3D2>&gt; requirements that &quot;Users must be </FONT>
<BR><FONT SIZE=3D2>&gt; authenticated using SIP Digest =
Authentication&quot;. This </FONT>
<BR><FONT SIZE=3D2>&gt; contradicts the statement in the SIP =
discussion</FONT>
<BR><FONT SIZE=3D2>&gt; list that the P-Asserted-Identity should be =
available in </FONT>
<BR><FONT SIZE=3D2>&gt; response messages, too. However I see no</FONT>
<BR><FONT SIZE=3D2>&gt; way to authenticate response messages with SIP =
mechanisms. </FONT>
<BR><FONT SIZE=3D2>&gt; Additionally the RFC shows the usage </FONT>
<BR><FONT SIZE=3D2>&gt; of these headers only for Requests.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; If there is a common understanding that the =
</FONT>
<BR><FONT SIZE=3D2>&gt; P-Asserted-Identity header is required for =
responses,</FONT>
<BR><FONT SIZE=3D2>&gt; will there be an update of the RFC?</FONT>
<BR><FONT SIZE=3D2>&gt; *&nbsp;&nbsp;&nbsp;&nbsp; What happens in a =
case where a proxy receives a </FONT>
<BR><FONT SIZE=3D2>&gt; response from a node he does not trust? He =
cannot</FONT>
<BR><FONT SIZE=3D2>&gt; authenticate the originator of the response. =
This means that </FONT>
<BR><FONT SIZE=3D2>&gt; he can only set up the =
P-Asserted-Identity</FONT>
<BR><FONT SIZE=3D2>&gt; for responsens of users which are stored within =
the database </FONT>
<BR><FONT SIZE=3D2>&gt; which is accessible by the proxy.</FONT>
<BR><FONT SIZE=3D2>&gt; *&nbsp;&nbsp;&nbsp;&nbsp; Does this mean that =
response messages from users not </FONT>
<BR><FONT SIZE=3D2>&gt; available in the database would be skipped by =
the</FONT>
<BR><FONT SIZE=3D2>&gt; proxy. </FONT>
<BR><FONT SIZE=3D2>&gt; *&nbsp;&nbsp;&nbsp;&nbsp; Which responses =
should contain this P-Asserted-Identity </FONT>
<BR><FONT SIZE=3D2>&gt; Header? Only final responses or even =
provisional ones? </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Best regards,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Wolfgang =
Deuringer</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
____________________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Wolfgang Deuringer</FONT>
<BR><FONT SIZE=3D2>&gt; ICN CP D NA D12</FONT>
<BR><FONT SIZE=3D2>&gt; Mch H/Sc8</FONT>
<BR><FONT SIZE=3D2>&gt; Tel.&nbsp; +49 89 722 28236</FONT>
<BR><FONT SIZE=3D2>&gt; Fax&nbsp; + 49 89 722 48697</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From mailnull@www1.ietf.org  Mon Mar 31 08:15:09 2003
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Thread-Topic: A couple of nits in RFC3261
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Subject: [Sip] A couple of nits in RFC3261
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- In section 13.2.2.4 Second last paragraph says:

   The UAC core considers the INVITE transaction completed 64*T1 seconds
   after the reception of the first 2xx response.  At this point all the
   early dialogs that have not transitioned to established dialogs are
   terminated.  Once the INVITE transaction is considered completed by
   the UAC core, no more new 2xx responses are expected to arrive

The problem is with the word "established" above. It should say "confirmed", since early dialogs are in fact established.

A related problem in that the words "created" and "established" are interchangeable in the rfc, and cause confusion. We should use just one of them?

- In section 15, Second paragraph:

   However, the callee's UA MUST NOT send a BYE on a confirmed dialog
   until it has received an ACK for its 2xx response or until the server
   transaction times out

For confirmed dialogs, the server transaction is already terminated and therefore cannot time out. It should say "... or until UAS core has the 2xx response for 64*T1 seconds without receiving an ACK.

Regards,
Hisham
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From: "Shane McCormack" <smccormack@tssg.org>
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Date: Mon, 31 Mar 2003 12:05:27 +0100
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Subject: [Sip] SIP servlet location implementation
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Hi, 
I wish to implement a sip servlet that subscribes to different users that
can theoretically receive geographical location NOTIFY reports. Would the
correct scenario be for a servlet to send these SUBSCRIBE messages to each
user or use a non-SIP entity such as a GPS Location Server?

Regards

Shane McCormack

Research Assistant,
TSSG, Confederation House, 
Waterford Business Park, Cork Road, 
Waterford, Co. Waterford. 

Phone: +353 51 302927     Fax: +353 51 302901 
Mobile: +353 87 9955505 
<http://www.tssg.org/> 
MSN: mccormackshane@hotmail.com <mailto:mccormackshane@hotmail.com> 

 


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Does anyone have the SIP client and server program
from the Columbia University?
I really need that program for my Final Project. 
Thanks

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From mailnull@www1.ietf.org  Mon Mar 31 11:28:05 2003
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Jussi.Turunen@nokia.com'" <Jussi.Turunen@nokia.com>, sip@ietf.org
Cc: Adam Roach <adam@dynamicsoft.com>
Subject: RE: [Sip] Event header in responses
Date: Mon, 31 Mar 2003 10:18:42 -0600
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The bug you indicate (#677) is in regards to the question
of how to match responses to their requests, not whether
"Event" is allowed in responses.

The short answer to your question, though, is that "Event"
should not be present in responses. The "Event:" header in
steps 6, 8, 10, and 12 of the example of event-lists-01 is
an oversight. Thanks for finding it.

That said, an implementation that receives an "Event" header
in a response should silently discard it.

/a

> -----Original Message-----
> From: Jussi.Turunen@nokia.com [mailto:Jussi.Turunen@nokia.com]
> Sent: Monday, March 31, 2003 0:29
> To: sip@ietf.org
> Cc: adam@dynamicsoft.com
> Subject: [Sip] Event header in responses
> 
> 
> Hi
> 
> Bug 677 (http://www.sipwg.org/sipwg/show_bug.cgi?id=677) says 
> that there is confilicting text in RFC 3265 about Event 
> header in responses. The wording of the bug is not 100% 
> clear, but I assume that responses should not contain an 
> Event header. The reason for my question is that SIP Service 
> Examples-04 and shows no Event in responses, presence-10 call 
> flow example shows it, event-list-01 draft shows it 
> sometimes. What is the correct interpretation?
> 
> 		Jussi
> 
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Subject: [Sip] Question Caller Preferences 08 draft
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Quoting the 08 draft  section 7.4 (Matching):

Next, the proxy applies the predicates associated with the Accept-Contact header field. For each contact that remains in the target set, the proxy constructs a matching set, Ms. Initially, this set contains all of the Accept-Contact predicates. Each of those predicates is examined. It is matched to the contact predicate using the matching operation of RFC 2533 [2]. If the result is not a match, and the Accept-Contact predicate had its require flag set, the URI corresponding to that contact predicate is discarded from the contact set. If the result is not a match, but the Accept-Contact predicate removed from the target set.

My question relates to the last statement in the above paragraph.
The objective of that step, as I understand it, is to remove from the set all predicates, that don't have an equivalent in the contact.
This step seems to be redundant, as it would not have an impact on the score, since it would not change the score anyway (based on the way the score is calculated)

Is my understanding correct?

My next question relates to the usage of the cancel-directive.
Quoting the Draft :

cancel-directive: This type of directive indicates whether the
             caller would like each proxy server to send a CANCEL
             request downstream ("cancel") in response to a 200 OK from
             the downstream server (which is the normal mode of
             operation, making it somewhat redundant), or whether this
             function should be left to the caller ("no-cancel"). If a
             proxy receives a request with this parameter set to "no-
             cancel", it SHOULD NOT CANCEL any outstanding branches on
             receipt of a 2xx. However, it would still send CANCEL on
             any outstanding branches on receipt of a 6xx.

The strength SHOULD NOT makes it unclear as to how the UAC can decide on what to do, even if the proxy understands caller preferences.

Rgds/gf
 


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3DISO-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
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<TITLE>Question Caller Preferences 08 draft</TITLE>
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<P><FONT FACE=3D"Times New Roman">Quoting the 08 draft&nbsp; section =
7.4 (Matching):</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Next, the proxy applies the =
predicates associated with the Accept-Contact header field. For each =
contact that remains in the target set, the proxy constructs a matching =
set, Ms. Initially, this set contains all of the Accept-Contact =
predicates. Each of those predicates is examined. It is matched to the =
contact predicate using the matching operation of RFC 2533 [2]. If the =
result is not a match, and the Accept-Contact predicate had its require =
flag set, the URI corresponding to that contact predicate is discarded =
from the contact set. If the result is not a match, but the =
Accept-Contact predicate removed from the target set.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">My question relates to the last =
statement in the above paragraph.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">The objective of that step, as =
I understand it, is to remove from the set all predicates, that don't =
have an equivalent in the contact.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">This step seems to be redundant, =
as it would not have an impact on the score, since it would not change =
the score anyway (based on the way the score is calculated)</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Is my understanding =
correct?</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">My next question relates to the =
usage of the cancel-directive.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">Quoting the Draft :</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">cancel-directive: This type of =
directive indicates whether the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; caller would like each proxy server to send a CANCEL</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; request downstream (&quot;cancel&quot;) in response to a 200 OK =
from</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; the downstream server (which is the normal mode of</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; operation, making it somewhat redundant), or whether this</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; function should be left to the caller (&quot;no-cancel&quot;). If =
a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; proxy receives a request with this parameter set to =
&quot;no-</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; cancel&quot;, it SHOULD NOT CANCEL any outstanding branches =
on</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; receipt of a 2xx. However, it would still send CANCEL on</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; any outstanding branches on receipt of a 6xx.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The strength SHOULD NOT makes it =
unclear as to how the UAC can decide on what to do, even if the proxy =
understands caller preferences.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Rgds/gf<I></I></FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;</FONT>
</P>

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