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From sip-bounces@ietf.org  Mon Nov  1 11:02:38 2004
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From: Nils Ohlmeier <lists@ohlmeier.org>
To: Robert Sparks <RjS@xten.com>
Subject: Re: [Sip] LC on nit-actions and nit-problems
Date: Mon, 1 Nov 2004 11:59:35 +0100
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Hi Robert,

as I did not found anything about that in the list archives: how about the UAC 
ignores the 100 Trying response for setting Timer E to T2?

I mean the UAC should not set Timer E to T2 if it receives a 100 Trying while 
Timer E < T2. So the 100 Trying will be treated as "I'm alive" for the 
possible DNS cache, but Timer E will be increased normaly as if no response 
would have been received. I think that could improve the reliability and 
should not harm.

Maybe I'm overseeing something here, so comments are welcome.

Greetings
  Nils Ohlmeier

On Friday 29 October 2004 20:29, Robert Sparks wrote:
> Folks -
>
> WGLC for the nit actions and problems draft started Oct 20 and is
> scheduled to
> end November 3 (next Wednesday).
>
> So far I have received _no_ comments.
>
> I believe that's because these drafts have been thoroughly discussed,
> are well
> understood, and they are ready to go (the points where more discussion
> is needed
> have all been moved to nit-future).
>
> If you agree, please take a moment to send a note to the list saying "I
> read this draft
> and believe it is ready to be published".
>
> If you disagree - now is the time to comment!
>
> Thanks,
> RjS
>
>
>
> _______________________________________________
> 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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--NextPart


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


        RFC 3903

        Title:      Session Initiation Protocol (SIP) Extension
                    for Event State Publication
        Author(s):  A. Niemi, Ed.
        Status:     Standards Track
        Date:       October 2004
        Mailbox:    aki.niemi@nokia.com
        Pages:      32
        Characters: 72062
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-sip-publish-04.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3903.txt


This document describes an extension to the Session Initiation
Protocol (SIP) for publishing event state used within the SIP Events
framework.  The first application of this extension is for the
publication of presence information.

The mechanism described in this document can be extended to support
publication of any event state for which there exists an appropriate
event package.  It is not intended to be a general-purpose mechanism
for transport of arbitrary data, as there are better-suited
mechanisms for this purpose.

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

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
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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
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From sip-bounces@ietf.org  Mon Nov  1 14:25:16 2004
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Date: Mon, 1 Nov 2004 14:02:56 -0500 (EST)
From: Sheetal Khemani <skhemani@cisco.com>
To: sip@ietf.org, sip-implementors@cs.columbia.edu
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Subject: [Sip] SIP AMR Codec (RFC3267) out-of-band bitrate change 
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Hi,

I want to know if AMR mode changes are supported via SIP/SDP (possibly
using a new off in a reINVITE).

RFC3267 doesn't mention any parameter in the SDP to change the bitrate
during the call using SIP. It allows the mode-set parameter that tells the
encoder the subset of modes/bitrates it can use.  It doesn't allow the
specification of the bitrate the other end needs to use/switch-to.

I came across some 3GPP docs (3GPP TS 26.236: Packet switched
conversational multimedia applications; Transport protocols) that suggest
using the bandwidth paramter, but that seems to be specific to QoS and I
am not sure if this applies to the SIP standard per se.

Please let me know if anyone has any insight on this topic.

Thanks
Sheetal

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Mon Nov  1 18:47:16 2004
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Cc: chair <rohan@ekabal.com>, Internet Architecture Board <iab@iab.org>,
        sip chair <dean.willis@softarmor.com>, sip mailing list <sip@ietf.org>,
        sip, RFC Editor <rfc-editor@rfc-editor.org>
Subject: [Sip] Protocol Action: 'Update to the Session Initiation Protocol 
 (SIP) Preconditions Framework' to Proposed Standard 
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The IESG has approved the following document:

- 'Update to the Session Initiation Protocol (SIP) Preconditions Framework '
   <draft-ietf-sip-rfc3312-update-03.txt> as a Proposed Standard

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

The IESG contact persons are Allison Mankin and Jon Peterson.

Technical Summary

   This document updates the framework for preconditions in SIP.  It
   clarifies RFC 3312; it does not obsolete RFC 3312, but just explains
   it better and adds some rules that update it.  It provides guidelines for
   authors who plan to write new precondition types, beyond the qos
   precondition.  

   In addition, this document describes and solves some issues concerning
   the interaction between session mobility and preconditions. 
 
Working Group Summary

  The working group supported advancement of this document.

 Protocol Quality
 
 The concerns about session mobility were identified in implementation
 experiences.  The document was reviewed for the IESG by Allison Mankin
 and by Spencer Dawkins of the General Area Review Team.

RFC Editor Notes

Abstract:
OLD:
This document updates the framework for preconditions in SIP.

NEW:
This document updates RFC 3312, which defines the framework for
preconditions in SIP.

Add the following paragraph to the end of the Introduction:

Specifically, we now allow answers to downgrade current status values
(this was disallowed by RFC 3312). We consider moving an existing stream
to a new location as equivalent to establishing a new stream. Therefore,
answerers moving streams to new locations set all the current status
values in their answers to "No" and start a new precondition negotiation
from scratch.

Section 4.1
OLD:
   Table 3 of RFC 3312 [3] needs to be updated to allow answers to
   downgrade current status values. 

NEW:
   Table 3 of RFC 3312 needs to be updated to allow answerers to
   downgrade current status values. 

Section 6
IANA Considerations

OLD:
This document has no IANA considerations.

NEW:
The IANA registration requirements for the preconditions framework are
defined in RFC 3312.  Any new preconditions are governed by the IANA
Considerations there.

Throughout the document: 
     After the first reference to [3], omit the [3].


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From sip-bounces@ietf.org  Mon Nov  1 21:14:30 2004
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Subject: [Sip] IPR disclosure on MESSAGE and related use of SIP
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I've just noticed that Nortel has recently been issued a patent for 
which they have made an IPR disclosure relative to RFC 3428, the SIP 
MESSAGE method.

The disclosure document is listed at:

http://www.ietf.org/ietf/IPR/nortel-ipr-rfc-3428.txt

The issued patent is US patent number 6,757,732

and you can retrieve its full text from:

http://patft.uspto.gov/netahtml/srchnum.htm


--
Dean Willis
SIP co-chair

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From sip-bounces@ietf.org  Mon Nov  1 21:54:18 2004
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Subject: [Sip] SIP Agenda, IETF 61
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The IETF agenda has not yet jelled, so times are relative. We're 
assuming two 2-hour sessions.

Every slot has received slightly MORE time than requested by the topic 
leader. Consequently, discussion will be cut off WITH A GUILLOTINE if 
required . . .

First Session
+0        Agenda bash and Status
             Chairs
+15      Connection Reuse
             Rohan Mahy, Cullen Jennings
             draft-ietf-sip-connect-reuse-03.txt
+30      Identity
             Jon Peterson
             draft-ietf-sip-identity-03.txt
+60      GRUU
             Jonathan Rosenberg
             draft-ietf-sip-gruu-02.txt
+80      Content Indirection
             Eric Burger
             draft-ietf-sip-content-indirect-mech-05.txt
+95      History Info
             Mary Barnes
             draft-ietf-sip-history-info-04.txt

Second Session
+0       Agenda Bash
            Chairs
+5       Event List
            Adam Roach
            draft-ietf-simple-event-list-06
+30     REFER Without Subscriptions
            Orit Levin
            draft-ietf-sip-refer-with-norefersub-00
+40     REFER Feature Parameters
            Orit Levin
            draft-ietf-sip-refer-with-feature-param-00
+50     Resource Priority
            James Polk
            draft-ietf-sip-resource-priority-05.txt
+70     Cert Usage
            Cullen jennings
            draft-ietf-sipping-certs-00.txt
+100   SIP With SAML
            Hannes Tschofenig
            draft-tschofenig-sip-saml-01.txt

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From sip-bounces@ietf.org  Mon Nov  1 22:15:50 2004
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] SIP Extensions
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Egil C. =D8sthus wrote:
> Hi,
>=20
> I have a question about section 8.1.1.9 in rfc3261.
> It is said in this section that "The option tags listed MUST only refer=
 to
> extensions defined in standards-track RFCs". Does this mean that the on=
ly
> legal extensions to SIP is those how are described in RFCs? If I would
> like to define a method in SIP, is it possible to use this method some =
way
> and at the same time be 100% SIP compliant?

Aboslutely. To make your extension 100% standards compliant, write a=20
standards-track RFC describing your extension.

There's more discussion of this process in RFC 3427

ftp://ftp.rfc-editor.org/in-notes/rfc3427.txt


The basic theory is that it is not standards-compliant to REQUIRE an=20
implementation manifest a non-standard extension.

So, you can invent your own extension and use it to your heart's=20
content, but your implementation should work in such a way that if your=20
extension is not available, it must "fall back" in some fairly graceful=20
manner to standard behavior. That way, if somebody accidentally connects=20
using a "standard" client, the world doesn't self-destruct underneath the=
m.

--
Dean Willis
co-chair, SIP Working Group

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From sip-bounces@ietf.org  Tue Nov  2 03:00:34 2004
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From: Avasarala Ranjit-A20990 <ranjit@motorola.com>
To: Sheetal Khemani <skhemani@cisco.com>, sip@ietf.org,
        sip-implementors@cs.columbia.edu
Subject: RE: [Sip] SIP AMR Codec (RFC3267) out-of-band bitrate change 
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Hi
   Can't the bandwidth attribute b= be used here? 

Regards
Ranjit



-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Sheetal Khemani
Sent: Tuesday, November 02, 2004 12:33 AM
To: sip@ietf.org; sip-implementors@cs.columbia.edu
Subject: [Sip] SIP AMR Codec (RFC3267) out-of-band bitrate change 


Hi,

I want to know if AMR mode changes are supported via SIP/SDP (possibly using a new off in a reINVITE).

RFC3267 doesn't mention any parameter in the SDP to change the bitrate during the call using SIP. It allows the mode-set parameter that tells the encoder the subset of modes/bitrates it can use.  It doesn't allow the specification of the bitrate the other end needs to use/switch-to.

I came across some 3GPP docs (3GPP TS 26.236: Packet switched conversational multimedia applications; Transport protocols) that suggest using the bandwidth paramter, but that seems to be specific to QoS and I am not sure if this applies to the SIP standard per se.

Please let me know if anyone has any insight on this topic.

Thanks
Sheetal

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Adam,

Can I please ask for some clarification on one aspect of RFC3265, Session
Initiation Protocol (SIP)-Specific Event Notification"?  Section 3.2
contains
the following text.

  "If any non-SUBSCRIBE mechanisms are defined to create subscriptions..."

I have always interpreted this as being the "window" required to allow REFER
initiated subscriptions or (perish the thought) subscriptions created by a
new
and hitherto undefined SIP method.  However, we have encountered customers
who
view this text as implying that a subscription can also be created using any
out-of-band mechanism, and therefore that from a SIP perspective, a NOTIFY
can
arrive without any prior SIP exchanges.

We see this as a major source of future interoperability problems as people
create their own, ad-hoc out-of-band subscription creation systems so we
want
to clarify whether this was the intention.  If it were not, we would like to
see a clear statement on the WG and in future updates to RFC3265 clarifying
this situation.

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




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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip


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To: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] IPR disclosure on MESSAGE and related use of SIP
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Suffice it to say, I am very unhappy about this.

This patent was filed in 2000. I do not understand why the disclosure 
was only made just now, some four years after it was filed and two years 
after issuance of the RFC. The only reasons I can think of are not very 
nice.

The licensing terms are also unclear. Does Nortel plan on extracting 
licensing revenues for this?

I think some clarifications and explanations from participants from 
Nortel are in order.

-Jonathan R.


Dean Willis wrote:

> 
> I've just noticed that Nortel has recently been issued a patent for 
> which they have made an IPR disclosure relative to RFC 3428, the SIP 
> MESSAGE method.
> 
> The disclosure document is listed at:
> 
> http://www.ietf.org/ietf/IPR/nortel-ipr-rfc-3428.txt
> 
> The issued patent is US patent number 6,757,732
> 
> and you can retrieve its full text from:
> 
> http://patft.uspto.gov/netahtml/srchnum.htm
> 
> 
> -- 
> Dean Willis
> SIP co-chair
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.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


From sip-bounces@ietf.org  Tue Nov  2 10:30:44 2004
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From: T Satyanarayana-A12694 <Satyanarayana.T@motorola.com>
To: "'Jonathan Rosenberg'" <jdrosen@cisco.com>,
        Dean Willis
	<dean.willis@softarmor.com>
Subject: RE: [Sip] IPR disclosure on MESSAGE and related use of SIP
Date: Tue, 2 Nov 2004 20:31:16 +0530 
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Jonathan etc,

For the benefit of those unaware of IP content in SIP standards, can somebody send to this group the list of patented content in all SIP-related drafts/RFCs? I don't know if such a list exists. Or, should that be extracted from the respective RFCs/drafts?

Regards
Satya T

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Jonathan Rosenberg
Sent: Tuesday, November 02, 2004 7:54 PM
To: Dean Willis
Cc: sip@ietf.org
Subject: Re: [Sip] IPR disclosure on MESSAGE and related use of SIP


Suffice it to say, I am very unhappy about this.

This patent was filed in 2000. I do not understand why the disclosure 
was only made just now, some four years after it was filed and two years 
after issuance of the RFC. The only reasons I can think of are not very 
nice.

The licensing terms are also unclear. Does Nortel plan on extracting 
licensing revenues for this?

I think some clarifications and explanations from participants from 
Nortel are in order.

-Jonathan R.


Dean Willis wrote:

> 
> I've just noticed that Nortel has recently been issued a patent for
> which they have made an IPR disclosure relative to RFC 3428, the SIP 
> MESSAGE method.
> 
> The disclosure document is listed at:
> 
> http://www.ietf.org/ietf/IPR/nortel-ipr-rfc-3428.txt
> 
> The issued patent is US patent number 6,757,732
> 
> and you can retrieve its full text from:
> 
> http://patft.uspto.gov/netahtml/srchnum.htm
> 
> 
> --
> Dean Willis
> SIP co-chair
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip Use 
> sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.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

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Tue Nov  2 11:19:52 2004
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To: T Satyanarayana-A12694 <Satyanarayana.T@motorola.com>
Subject: Re: [Sip] IPR disclosure on MESSAGE and related use of SIP 
In-Reply-To: Message from T Satyanarayana-A12694
	<Satyanarayana.T@motorola.com> of "Tue,
	02 Nov 2004 20:31:16 +0530."
	<C73F06B054B7D6118C040008C7F3613D03FBB6AB@zin09exm01.corp.mot.com>
Date: Tue, 02 Nov 2004 07:52:08 -0800
From: Allison Mankin <mankin@psg.com>
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FYI,

It is now possible, as of about two weeks ago, to do a search of the IETF
IPR claims by working group.  

Note that we hope that lots of you do not know of other IPR claims.

Participants need to go back and re-study the Note Well:

http://www.ietf.org/overview.html

Below is the info on IPR Search, if you didn't see it.  

Allison


P.S In the info below, the presentation of related documents in the search
in no way implies that the IPR claim applies to them.  IPR claims apply only
to the specific i-d named in the claim, since a revision of the document
often changes the technology.

---------

To: IETF Announcement list <ietf-announce@ietf.org>
From: IETF Secretariat <ietf-secretariat-reply@ietf.org>
Subject: Enhanced IETF IPR Disclosure Search Capability 


The IETF Secretariat is pleased to announce the deployment of an enhanced search
capability for the "IETF IPR Disclosure Page"
(https://datatracker.ietf.org/public/ipr_disclosure.cgi).  With this new
implementation, when a user searches on an I-D filename or an RFC number, the
search will return all IPR disclosures associated with the document as well as
all IPR disclosures associated with all related documents.  For example, if the
user searches on an RFC number, then the search will return all IPR disclosures
associated with:


o The RFC
o Any version of the I-D that became the RFC
o Any RFCs that replaced, updated, or obsoleted the RFC and their associated
  I-Ds
o Any RFCs that were replaced by, updated by, or obsoleted by the RFC, and their
  associated I-Ds


In addition, the new implementation will list all of the documents included in
the search (by both title and RFC number or I-D filename) whether or not IPR
disclosures have been submitted for those documents, and will indicate the
relationship of each document to other documents on the list.


If you require assistance in using this enhanced "Search" capability, or wish
to report a bug, then please send a message to ietf-action@ietf.org.


The IETF Secretariat

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IETF-Announce mailing list
IETF-Announce@ietf.org
https://www1.ietf.org/mailman/listinfo/ietf-announce


_______________________________________________
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: T Satyanarayana-A12694 <Satyanarayana.T@motorola.com>
From: Peter Macaulay <peter.macaulay@ZDSL.com>
Subject: RE: [Sip] IPR disclosure on MESSAGE and related use of SIP
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Internet Protocol (IP) is used extensively by SIP!  :-)

Intellectual Property Rights (IPR) can be found at;
https://datatracker.ietf.org/public/ipr_disclosure.cgi

You can search for the keyword SIP and find about 20 IPR disclosures;
https://datatracker.ietf.org/public/ipr_list.cgi

  Regards,
/PETER MACAULAY

At 10:01 AM 11/2/2004, T Satyanarayana-A12694 wrote:
>Jonathan etc,
>
>For the benefit of those unaware of IP content in SIP standards, can 
>somebody send to this group the list of patented content in all 
>SIP-related drafts/RFCs? I don't know if such a list exists. Or, should 
>that be extracted from the respective RFCs/drafts?
>
>Regards
>Satya T
>
>-----Original Message-----
>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of 
>Jonathan Rosenberg
>Sent: Tuesday, November 02, 2004 7:54 PM
>To: Dean Willis
>Cc: sip@ietf.org
>Subject: Re: [Sip] IPR disclosure on MESSAGE and related use of SIP
>
>
>Suffice it to say, I am very unhappy about this.
>
>This patent was filed in 2000. I do not understand why the disclosure
>was only made just now, some four years after it was filed and two years
>after issuance of the RFC. The only reasons I can think of are not very
>nice.
>
>The licensing terms are also unclear. Does Nortel plan on extracting
>licensing revenues for this?
>
>I think some clarifications and explanations from participants from
>Nortel are in order.
>
>-Jonathan R.
>
>
>Dean Willis wrote:
>
> >
> > I've just noticed that Nortel has recently been issued a patent for
> > which they have made an IPR disclosure relative to RFC 3428, the SIP
> > MESSAGE method.
> >
> > The disclosure document is listed at:
> >
> > http://www.ietf.org/ietf/IPR/nortel-ipr-rfc-3428.txt
> >
> > The issued patent is US patent number 6,757,732
> >
> > and you can retrieve its full text from:
> >
> > http://patft.uspto.gov/netahtml/srchnum.htm
> >
> >
> > --
> > Dean Willis
> > SIP co-chair
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip Use
> > sipping@ietf.org for new developments on the application of sip
> >
>
>--
>Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
>Cisco Systems
>jdrosen@cisco.com                              FAX:   (973) 952-5050
>http://www.jdrosen.net                         PHONE: (973) 952-5000
>http://www.cisco.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
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip



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From sip-bounces@ietf.org  Tue Nov  2 11:52:13 2004
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Folks,

FYI: this draft (draft-ietf-mmusic-connection-precon-00.txt) will be 
discussed in the MMUSIC meeting in Washington.

Regards,

Gonzalo

Colin Perkins wrote:

> Folks,
> 
> The MMUSIC working group is considering a draft on 
> Connection-Establishment Preconditions in SIP 
> <draft-ietf-mmusic-connection-precon-00.txt>. As there are obvious 
> implications for SIP, this note is to solicit review by the SIP working 
> group before we advance this draft in MMUSIC.
> 
> We'd be grateful to receive any comments - both for or against this 
> draft - in reply to this message.
> 
> Thanks,
> Colin
> 
> 


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From sip-bounces@ietf.org  Tue Nov  2 12:00:33 2004
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To: Dean Willis <dean.willis@softarmor.com>
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I want to make three observations of fact.

    * Patent 6,757,732 was filed March 16, 2000.

    * The MESSAGE method was used to exchange instant messages using SIP
      in August of 1999, and the source code to the client used to do so
      ("insipid," later renamed "kphone") was published on a publicly
      accessible web site at that time. A screen shot of the client
      performing instant messaging can be seen here:

      http://web.archive.org/web/20010624055115/http://www.billybiggs.com/screenshots/insipid-19-aug-99.gif

    * By April of 2000, there were enough implementations of the SIP
      MESSAGE method to have testing between multiple independent
      implementations at the 4th SIP bakeoff. I was a party to several
      of these tests.


To be clear: I make no claims about the validity of this patent. I am 
simply stating facts that I believe may be of interest to this 
community. Draw your own conclusions.

/a

Dean Willis wrote:

>
> I've just noticed that Nortel has recently been issued a patent for 
> which they have made an IPR disclosure relative to RFC 3428, the SIP 
> MESSAGE method.
>
> The disclosure document is listed at:
>
> http://www.ietf.org/ietf/IPR/nortel-ipr-rfc-3428.txt
>
> The issued patent is US patent number 6,757,732
>
> and you can retrieve its full text from:
>
> http://patft.uspto.gov/netahtml/srchnum.htm
>
>
> -- 
> Dean Willis
> SIP co-chair
>
> _______________________________________________
> 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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Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Tue Nov  2 12:10:15 2004
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From: Srini Krishnamoorthy <ksrini@motorola.com>
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To: Jonathan Rosenberg <jdrosen@cisco.com>
Subject: Re: [Sip] IPR disclosure on MESSAGE and related use of SIP
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Jonathan et.al,

  Agree, and I think a LOT of people will be unhappy too ..

  Do we read this as specific to the use of "INFO + HTML body" or
  in general :-
   "communicating the text-based chat message includes embedding the 
    text-based chat message" (which the the first claim states)?"

  How does this IPR affect the use of MESSAGE method with content-types
  other than text/html  (say plain, xml) - what would be considered 
  infringment? .. anybody Nortel??

  Please help understand the detailed impact of this ... thanks.

  Or should I just wait to see what transpires?? 

-Srini




On Tue, 2 Nov 2004, Jonathan Rosenberg wrote:

jdrose>Suffice it to say, I am very unhappy about this.
jdrose>
jdrose>This patent was filed in 2000. I do not understand why the disclosure 
jdrose>was only made just now, some four years after it was filed and two years 
jdrose>after issuance of the RFC. The only reasons I can think of are not very 
jdrose>nice.
jdrose>
jdrose>The licensing terms are also unclear. Does Nortel plan on extracting 
jdrose>licensing revenues for this?
jdrose>
jdrose>I think some clarifications and explanations from participants from 
jdrose>Nortel are in order.
jdrose>
jdrose>-Jonathan R.
jdrose>
jdrose>
jdrose>Dean Willis wrote:
jdrose>
jdrose>> 
jdrose>> I've just noticed that Nortel has recently been issued a patent for 
jdrose>> which they have made an IPR disclosure relative to RFC 3428, the SIP 
jdrose>> MESSAGE method.
jdrose>> 
jdrose>> The disclosure document is listed at:
jdrose>> 
jdrose>> http://www.ietf.org/ietf/IPR/nortel-ipr-rfc-3428.txt
jdrose>> 
jdrose>> The issued patent is US patent number 6,757,732
jdrose>> 
jdrose>> and you can retrieve its full text from:
jdrose>> 
jdrose>> http://patft.uspto.gov/netahtml/srchnum.htm
jdrose>> 
jdrose>> 
jdrose>> -- 
jdrose>> Dean Willis
jdrose>> SIP co-chair
jdrose>> 
jdrose>> _______________________________________________
jdrose>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
jdrose>> This list is for NEW development of the core SIP Protocol
jdrose>> Use sip-implementors@cs.columbia.edu for questions on current sip
jdrose>> Use sipping@ietf.org for new developments on the application of sip
jdrose>> 
jdrose>
jdrose>

-- 
Srini K
ksrini at motorola 
Finger ksrini@bulb.corp.mot.com for pub key.


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Paul D.Smith wrote:

>Adam,
>
>Can I please ask for some clarification on one aspect of RFC3265, Session
>Initiation Protocol (SIP)-Specific Event Notification"?  Section 3.2
>contains
>the following text.
>
>  "If any non-SUBSCRIBE mechanisms are defined to create subscriptions..."
>
>I have always interpreted this as being the "window" required to allow REFER
>initiated subscriptions or (perish the thought) subscriptions created by a
>new
>and hitherto undefined SIP method.  However, we have encountered customers
>who
>view this text as implying that a subscription can also be created using any
>out-of-band mechanism, and therefore that from a SIP perspective, a NOTIFY
>can
>arrive without any prior SIP exchanges.
>  
>

Such was the intention. This text was a source of much contention during 
the development of this document, and it was repeatedly removed, 
re-added, and re-crafted to try to find a balance.

The paragraph that you cite is the final compromise. It defines some 
very strict guidelines about how any such mechanisms must behave. In 
particular:

>   Designers of such mechanisms are also
>   warned to make a distinction between sending a NOTIFY message to a
>   subscriber who is aware of the subscription, and sending a NOTIFY
>   message to an unsuspecting node.  The latter behavior is invalid, and
>   MUST receive a "481 Subscription does not exist" response.... 
>   In other words, knowledge of a subscription must exist in both the
>   subscriber and the notifier to be valid, even if installed via a
>   non-SUBSCRIBE mechanism.
>

To be clear:

    * It is a violation of the specification to send a NOTIFY to a node
      that you do not know for a fact is expecting that NOTIFY.
    * It is a violation of the specification to send a 200-class
      response to a NOTIFY if you were not told, via some mechanism, in
      advance, that such a NOTIFY would be arriving.


All of that is historical information at this point. For future 
revisions of the specification, it is highly likely that these 
provisions will be removed, as no published mechanisms have taken 
advantage of them (except for REFER, which will likely be revised to 
remove the implicit subscription in favor of an explicit subscription.)

/a

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Srini Krishnamoorthy wrote:

>  Please help understand the detailed impact of this ... thanks.
>
>  Or should I just wait to see what transpires?? 
>  
>

Well, as heinous as all of this is, I think the appropriate response 
would be to put your efforts into developing an MSRP client instead of a 
MESSAGE-based implementation.

/a

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Subject: RE: [Sip] IPR disclosure on MESSAGE and related use of SIP
Date: Tue, 2 Nov 2004 15:10:20 -0700
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Thread-Topic: [Sip] IPR disclosure on MESSAGE and related use of SIP
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From: "Hearty, John" <John.Hearty@Level3.com>
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Perhaps I'm missing something, but I looked over the patent quickly and
all the examples in the text were around using the INFO method, not the
MESSAGE method.  It was of course laced with language making it sound
rather general so perhaps it could be applied to other SIP methods.

John Hearty
Level3

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Adam Roach
Sent: Tuesday, November 02, 2004 12:50 PM
To: ksrini@motorola.com
Cc: sip@ietf.org; Dean Willis
Subject: Re: [Sip] IPR disclosure on MESSAGE and related use of SIP

Srini Krishnamoorthy wrote:

>  Please help understand the detailed impact of this ... thanks.
>
>  Or should I just wait to see what transpires??=20
> =20
>

Well, as heinous as all of this is, I think the appropriate response=20
would be to put your efforts into developing an MSRP client instead of a

MESSAGE-based implementation.

/a

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_______________________________________________
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From sip-bounces@ietf.org  Tue Nov  2 19:23:28 2004
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I'm nothing like a lawyer, so don't take this as legal advice. On the 
other hand, I've assisted with the IPR process (my own IPR and others') 
at two different companies, so I have some reasonable layperson 
background here.

What /really/ matters in a patent are the /independent/ claims. That is, 
the ones that don't refer to any of the other claims.

So, to evaluate the impact of this patent, look at claims 1, 9, 13, 22, 
25, 28, and 32. In a vacuum. Ignore the entire rest of the document.

The other claims are there just in case the independent claims are ruled 
to be too broad. They progressively narrow the scope so the patent can 
"fall back" to one of these narrower scopes. Unless the patent is 
challenged in court, they are just noise.

Anything that's not a claim is there just to provide context for the 
claims; it doesn't affect what can be enforced.

/a

Hearty, John wrote:

>Perhaps I'm missing something, but I looked over the patent quickly and
>all the examples in the text were around using the INFO method, not the
>MESSAGE method.  It was of course laced with language making it sound
>rather general so perhaps it could be applied to other SIP methods.
>
>John Hearty
>Level3
>
>-----Original Message-----
>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>Adam Roach
>Sent: Tuesday, November 02, 2004 12:50 PM
>To: ksrini@motorola.com
>Cc: sip@ietf.org; Dean Willis
>Subject: Re: [Sip] IPR disclosure on MESSAGE and related use of SIP
>
>Srini Krishnamoorthy wrote:
>
>  
>
>> Please help understand the detailed impact of this ... thanks.
>>
>> Or should I just wait to see what transpires?? 
>> 
>>
>>    
>>
>
>Well, as heinous as all of this is, I think the appropriate response 
>would be to put your efforts into developing an MSRP client instead of a
>
>MESSAGE-based implementation.
>
>/a
>
>_______________________________________________
>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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Dear friends,
I have been studying security and privacy in general and I am interested =
in features/works/papers to improve security and privacy in SIP. =20

Someone could give some more information, articles, papers, RFCs/drafts =
related this topic, including something about lawful interception and =
Spam over IP Telephony??

Really thanks in advance for any help.

Sergio

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
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<DIV><FONT face=3D"Century Gothic" color=3D#0000ff size=3D2><FONT=20
face=3D"Century Gothic" color=3D#0000ff size=3D2><FONT size=3D2>Dear =
friends,</DIV>
<DIV>
<DIV>
<P>I have been studying security and privacy in general and I am =
interested in=20
features/works/papers to improve security and privacy in SIP.&nbsp; </P>
<P>Someone could give some more information, articles, papers, =
RFCs/drafts=20
related this topic, including something about lawful interception and =
Spam over=20
IP Telephony??</P>
<P>Really thanks in advance for any help.</P>
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_______________________________________________
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From sip-bounces@ietf.org  Wed Nov  3 06:21:45 2004
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Subject: RE: [Sip] non-SUBSCRIBE subscriptions
Date: Wed, 3 Nov 2004 11:33:57 +0100
Message-ID: <B30D6148F304A743A53B9B44751CDB9A0233CA7D@FTRDMEL2.rd.francetelecom.fr>
Thread-Topic: [Sip] non-SUBSCRIBE subscriptions
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From: "MANSOOR Usama RD-ILAB-LON" <usama.mansoor@rd.francetelecom.com>
To: "Adam Roach" <adam@nostrum.com>,
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So will there no longer be such a thing as 'implicit subscriptions'?

- Usama

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Adam Roach
Sent: 02 November 2004 16:34
To: Paul D.Smith
Cc: SIP WG
Subject: Re: [Sip] non-SUBSCRIBE subscriptions

Paul D.Smith wrote:

>Adam,
>
>Can I please ask for some clarification on one aspect of RFC3265,
Session
>Initiation Protocol (SIP)-Specific Event Notification"?  Section 3.2
>contains
>the following text.
>
>  "If any non-SUBSCRIBE mechanisms are defined to create
subscriptions..."
>
>I have always interpreted this as being the "window" required to allow
REFER
>initiated subscriptions or (perish the thought) subscriptions created
by a
>new
>and hitherto undefined SIP method.  However, we have encountered
customers
>who
>view this text as implying that a subscription can also be created
using any
>out-of-band mechanism, and therefore that from a SIP perspective, a
NOTIFY
>can
>arrive without any prior SIP exchanges.
> =20
>

Such was the intention. This text was a source of much contention during

the development of this document, and it was repeatedly removed,=20
re-added, and re-crafted to try to find a balance.

The paragraph that you cite is the final compromise. It defines some=20
very strict guidelines about how any such mechanisms must behave. In=20
particular:

>   Designers of such mechanisms are also
>   warned to make a distinction between sending a NOTIFY message to a
>   subscriber who is aware of the subscription, and sending a NOTIFY
>   message to an unsuspecting node.  The latter behavior is invalid,
and
>   MUST receive a "481 Subscription does not exist" response....=20
>   In other words, knowledge of a subscription must exist in both the
>   subscriber and the notifier to be valid, even if installed via a
>   non-SUBSCRIBE mechanism.
>

To be clear:

    * It is a violation of the specification to send a NOTIFY to a node
      that you do not know for a fact is expecting that NOTIFY.
    * It is a violation of the specification to send a 200-class
      response to a NOTIFY if you were not told, via some mechanism, in
      advance, that such a NOTIFY would be arriving.


All of that is historical information at this point. For future=20
revisions of the specification, it is highly likely that these=20
provisions will be removed, as no published mechanisms have taken=20
advantage of them (except for REFER, which will likely be revised to=20
remove the implicit subscription in favor of an explicit subscription.)

/a

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Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Wed Nov  3 09:29:38 2004
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To: Jonathan Rosenberg <jdrosen@cisco.com>
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I'm probably going to get into trouble for responding, but I can't sit by and let 
you throw mud at my colleagues.

Jonathan, you once worked for a large company.  Now you do, again.  Can you really 
say you know everything that's going on in Cisco that might relate to your ETF 
activities?

It's getting tougher as we get slimmed down, but Nortel (and BNR before that) has 
had a long tradition of people in odd corners exploring the possibilities of 
technology and coming up with new ideas unrelated to their current products.  And 
like it or not, Nortel views intellectual property as a source of value, so people 
are encouraged to take out patents on new ideas.

Mary was unaware of the patent concerned until a week before the patent 
announcement.  At that point she took the necessary steps to have it announced.

And we who participate at the IETF are not patent lawyers -- we can only refer you 
to our own legal people for further advice.

Jonathan Rosenberg wrote:
> Suffice it to say, I am very unhappy about this.
> 
> This patent was filed in 2000. I do not understand why the disclosure 
> was only made just now, some four years after it was filed and two years 
> after issuance of the RFC. The only reasons I can think of are not very 
> nice.
> 
> The licensing terms are also unclear. Does Nortel plan on extracting 
> licensing revenues for this?
> 
> I think some clarifications and explanations from participants from 
> Nortel are in order.
> 
> -Jonathan R.
> 
> 


-- 
Tom Taylor
Carrier VoIP Standards Development
Nortel Networks
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortelnetworks.com

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From sip-bounces@ietf.org  Wed Nov  3 11:51:07 2004
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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'Takuya Sawada'" <tu-sawada@kddi.com>
Subject: RE: FW: [Sip] I-D ACTION:draft-ietf-sip-history-info-04.txt
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Hi Takuya,

I apologize if I missed this specific point in your previous postings. You
are correct, that index of 2 should have been 1.1 and this  actually affects
the whole series and also affects the examples in  4.5.1 and 4.5.2.  All the
other entries are correct relative to the index of 2, it's just the 2 should
have been a 1.1, unless of course that proxy had a good reason for starting
at 2, which I don't explain so shouldn't use in the example.  

It looks like I introduced that error in the -01 version when I was updating
the example to include the index for all entries (as the index was
originally optional). I traced the change back to changes I made during the
middle of the day, so I can't even blame late nite editting.   I will
definitely make a note of that and update that in the -05 version, which
will hopefully be the one to go to the IESG.

I'm also copying the SIP list, so that folks are aware of this as they
review the document. 

Thanks for your careful review,
Mary


-----Original Message-----
From: Takuya Sawada [mailto:tu-sawada@kddi.com] 
Sent: Tuesday, November 02, 2004 4:29 AM
To: Barnes, Mary [NGC:B601:EXCH]
Subject: Re: FW: [Sip] I-D ACTION:draft-ietf-sip-history-info-04.txt


Hi Mary,

Each example flow in section 4.5 begins with the following messages,

   UA1        Proxy1  Proxy2     UA2      UA3      UA4      UA5 
                
   |            |         |        |        |        |        | 
   |--INVITE -->|         |        |        |        |        | 
   |            |-INVITE->|        |        |        |        | 
                 Supported: Histinfo 
                 History-Info: <sip:Bob@P1.example.com>;index=1,
                               <sip:Bob@P2.example.com>;index=2 

I think this should be

   UA1        Proxy1  Proxy2     UA2      UA3      UA4      UA5 
                
   |            |         |        |        |        |        | 
   |--INVITE -->|         |        |        |        |        | 
   |            |-INVITE->|        |        |        |        | 
                 Supported: Histinfo 
                 History-Info: <sip:Bob@P1.example.com>;index=1,
                               <sip:Bob@P2.example.com>;index=1.1

Note that the index of the second hi-entry is 1.1.
I made the same comment to the list before, but no response to it.

In Appendix C, it shows,

   UA1          Proxy        ACDGRP1 Svr   ACDGRP2 Svr UA2-ACDGRP2

                
   |              |              |             |          | 
   |--INVITE F1-->|              |             |          | 
    Supported:Histinfo 
   |              |              |             |          | 
   |              |--INVITE F2-->|             |          | 
                    Supported:Histinfo 
                    History-Info: <sip:Gold@example.com>; index=1  
                    History-Info: <sip:ACDGRP1@example.com>; index=1.1 

I can not find what is the difference between the two.
Are you saying that the former is "Retargeting within a  Proxy" and the 
latter is "Basic Forwarding"?
Am I missing something?

Thanks.

Regards,
Takuya

> 
> 
> Hi all,
> 
> Since this version should be ready for WGLC, a very detailed list of the
> changes is provided in the document, annotated as to the source, so I
won't
> repeat those details here.   The majority of the changes were the issues
> discussed at IETF-60, along with the agreements there to change the text
to
> non-normative in section 4 (Protocol structure) and to add some detail per
> Rohan's comment on the necessary processing should TLS not be available.
In
> addition, I received some proposed changes on a marked up hardcopy at
> IETF-61 from Eric Burger.  
> 
> The only items not discussed at IETF-60 were 2 items that came up on the
> list (one posted by John Elwell on August 18th on handling of privacy in
> responses) the other on Oct. 15th around the appropriate character format
> for the escaped headers in the URI. And, as always, there are various
minor
> editorial changes here and there while I was "in the area".  So, there
> should be no surprises with any of the changes, although, it is possible
> that there are still errors and nits that can be identified during WGLC.

> 
> Thanks,
> Mary
> 
> 
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Internet-Drafts@ietf.org
> Sent: Monday, October 25, 2004 3:07 PM
> To: i-d-announce@ietf.org
> Cc: sip@ietf.org
> Subject: [Sip] I-D ACTION:draft-ietf-sip-history-info-04.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		: An Extension to the Session Initiation Protocol
> for Request History Information
> 	Author(s)	: M. Barnes
> 	Filename	: draft-ietf-sip-history-info-04.txt
> 	Pages		: 47
> 	Date		: 2004-10-25
> 	
> This draft defines a standard mechanism for capturing the history 
>    information associated with a SIP request.  This capability enables 
>    many enhanced services by providing the information as to how and why 
>    a call arrives at a specific application or user.  This draft defines 
>    a new optional SIP header, History-Info, for capturing the history 
>    information in requests. A new option tag, Histinfo, to be included 
>    in the Supported header, is defined to allow UAs to indicate whether 
>    the History-Info should be returned in responses to a request which 
>    has captured the history information. A new priv-value, history, is 
>    added to the Privacy header to allow for privacy handling of the 
>    History-Info header.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-04.txt
> 
> To remove yourself from the I-D Announcement list, send a message to 
> i-d-announce-request@ietf.org with the word unsubscribe in the body of the
> message.  
> You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce 
> to change your subscription settings.
> 
> 
> 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-history-info-04.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-history-info-04.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
> 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
> 


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10, Iidabashi, 
Chiyoda-ku, Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0286
tu-sawada@kddi.com


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<TITLE>RE: FW: [Sip] I-D =
ACTION:draft-ietf-sip-history-info-04.txt</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Hi Takuya,</FONT>
</P>

<P><FONT SIZE=3D2>I apologize if I missed this specific point in your =
previous postings. You are correct, that index of 2 should have been =
1.1 and this&nbsp; actually affects the whole series and also affects =
the examples in&nbsp; 4.5.1 and 4.5.2.&nbsp; All the other entries are =
correct relative to the index of 2, it's just the 2 should have been a =
1.1, unless of course that proxy had a good reason for starting at 2, =
which I don't explain so shouldn't use in the example.&nbsp; =
</FONT></P>

<P><FONT SIZE=3D2>It looks like I introduced that error in the -01 =
version when I was updating the example to include the index for all =
entries (as the index was originally optional). I traced the change =
back to changes I made during the middle of the day, so I can't even =
blame late nite editting.&nbsp;&nbsp; I will definitely make a note of =
that and update that in the -05 version, which will hopefully be the =
one to go to the IESG.</FONT></P>

<P><FONT SIZE=3D2>I'm also copying the SIP list, so that folks are =
aware of this as they review the document. </FONT>
</P>

<P><FONT SIZE=3D2>Thanks for your careful review,</FONT>
<BR><FONT SIZE=3D2>Mary</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Takuya Sawada [<A =
HREF=3D"mailto:tu-sawada@kddi.com">mailto:tu-sawada@kddi.com</A>] =
</FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, November 02, 2004 4:29 AM</FONT>
<BR><FONT SIZE=3D2>To: Barnes, Mary [NGC:B601:EXCH]</FONT>
<BR><FONT SIZE=3D2>Subject: Re: FW: [Sip] I-D =
ACTION:draft-ietf-sip-history-info-04.txt</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Hi Mary,</FONT>
</P>

<P><FONT SIZE=3D2>Each example flow in section 4.5 begins with the =
following messages,</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp; =
UA1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Proxy1&nbsp; =
Proxy2&nbsp;&nbsp;&nbsp;&nbsp; UA2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
UA3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA4&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA5 =
</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; |--INVITE =
--&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|-INVITE-&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Supported: Histinfo </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; History-Info: =
&lt;sip:Bob@P1.example.com&gt;;index=3D1,</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&lt;sip:Bob@P2.example.com&gt;;index=3D2 </FONT>
</P>

<P><FONT SIZE=3D2>I think this should be</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp; =
UA1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Proxy1&nbsp; =
Proxy2&nbsp;&nbsp;&nbsp;&nbsp; UA2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
UA3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA4&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA5 =
</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; |--INVITE =
--&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|-INVITE-&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Supported: Histinfo </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; History-Info: =
&lt;sip:Bob@P1.example.com&gt;;index=3D1,</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&lt;sip:Bob@P2.example.com&gt;;index=3D1.1</FONT>
</P>

<P><FONT SIZE=3D2>Note that the index of the second hi-entry is =
1.1.</FONT>
<BR><FONT SIZE=3D2>I made the same comment to the list before, but no =
response to it.</FONT>
</P>

<P><FONT SIZE=3D2>In Appendix C, it shows,</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp; =
UA1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Proxy&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ACDGRP1 Svr&nbsp;&nbsp; =
ACDGRP2 Svr =
UA2-ACDGRP2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; |--INVITE =
F1--&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; Supported:Histinfo </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp; |--INVITE =
F2--&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Supported:Histinfo =
</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
History-Info: &lt;sip:Gold@example.com&gt;; index=3D1&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; History-Info: =
&lt;sip:ACDGRP1@example.com&gt;; index=3D1.1 </FONT>
</P>

<P><FONT SIZE=3D2>I can not find what is the difference between the =
two.</FONT>
<BR><FONT SIZE=3D2>Are you saying that the former is &quot;Retargeting =
within a&nbsp; Proxy&quot; and the </FONT>
<BR><FONT SIZE=3D2>latter is &quot;Basic Forwarding&quot;?</FONT>
<BR><FONT SIZE=3D2>Am I missing something?</FONT>
</P>

<P><FONT SIZE=3D2>Thanks.</FONT>
</P>

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

<P><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Hi all,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Since this version should be ready for WGLC, a =
very detailed list of the</FONT>
<BR><FONT SIZE=3D2>&gt; changes is provided in the document, annotated =
as to the source, so I won't</FONT>
<BR><FONT SIZE=3D2>&gt; repeat those details here.&nbsp;&nbsp; The =
majority of the changes were the issues</FONT>
<BR><FONT SIZE=3D2>&gt; discussed at IETF-60, along with the agreements =
there to change the text to</FONT>
<BR><FONT SIZE=3D2>&gt; non-normative in section 4 (Protocol structure) =
and to add some detail per</FONT>
<BR><FONT SIZE=3D2>&gt; Rohan's comment on the necessary processing =
should TLS not be available.&nbsp; In</FONT>
<BR><FONT SIZE=3D2>&gt; addition, I received some proposed changes on a =
marked up hardcopy at</FONT>
<BR><FONT SIZE=3D2>&gt; IETF-61 from Eric Burger.&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The only items not discussed at IETF-60 were 2 =
items that came up on the</FONT>
<BR><FONT SIZE=3D2>&gt; list (one posted by John Elwell on August 18th =
on handling of privacy in</FONT>
<BR><FONT SIZE=3D2>&gt; responses) the other on Oct. 15th around the =
appropriate character format</FONT>
<BR><FONT SIZE=3D2>&gt; for the escaped headers in the URI. And, as =
always, there are various minor</FONT>
<BR><FONT SIZE=3D2>&gt; editorial changes here and there while I was =
&quot;in the area&quot;.&nbsp; So, there</FONT>
<BR><FONT SIZE=3D2>&gt; should be no surprises with any of the changes, =
although, it is possible</FONT>
<BR><FONT SIZE=3D2>&gt; that there are still errors and nits that can =
be identified during WGLC.&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Thanks,</FONT>
<BR><FONT SIZE=3D2>&gt; Mary</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: sip-bounces@ietf.org [<A =
HREF=3D"mailto:sip-bounces@ietf.org">mailto:sip-bounces@ietf.org</A>] =
On Behalf Of</FONT>
<BR><FONT SIZE=3D2>&gt; Internet-Drafts@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Monday, October 25, 2004 3:07 PM</FONT>
<BR><FONT SIZE=3D2>&gt; To: i-d-announce@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: [Sip] I-D =
ACTION:draft-ietf-sip-history-info-04.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 Internet-Drafts</FONT>
<BR><FONT SIZE=3D2>&gt; directories.</FONT>
<BR><FONT SIZE=3D2>&gt; This draft is a work item of the Session =
Initiation Protocol Working Group</FONT>
<BR><FONT SIZE=3D2>&gt; 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; : An =
Extension to the Session Initiation Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; for Request History Information</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : M. Barnes</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-ietf-sip-history-info-04.txt</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Pages&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
47</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Date&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
2004-10-25</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; This draft defines a standard mechanism for =
capturing the history </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; information associated with a =
SIP request.&nbsp; This capability enables </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; many enhanced services by =
providing the information as to how and why </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; a call arrives at a specific =
application or user.&nbsp; This draft defines </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; a new optional SIP header, =
History-Info, for capturing the history </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; information in requests. A =
new option tag, Histinfo, to be included </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; in the Supported header, is =
defined to allow UAs to indicate whether </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; the History-Info should be =
returned in responses to a request which </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; has captured the history =
information. A new priv-value, history, is </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; added to the Privacy header =
to allow for privacy handling of the </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; History-Info header.</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-history-info-=
04.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-sip-his=
tory-info-04.txt</A></FONT>
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<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 =
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&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 =
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<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; =
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<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
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<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>
<BR>

<P><FONT SIZE=3D2>--------</FONT>
<BR><FONT SIZE=3D2>Takuya Sawada</FONT>
<BR><FONT SIZE=3D2>KDDI Corporation (KDDI)</FONT>
<BR><FONT SIZE=3D2>Garden Air Tower, 3-10-10, Iidabashi, </FONT>
<BR><FONT SIZE=3D2>Chiyoda-ku, Tokyo 102-8460, Japan</FONT>
<BR><FONT SIZE=3D2>Tel: +81-3-6678-2997</FONT>
<BR><FONT SIZE=3D2>Fax: +81-3-6678-0286</FONT>
<BR><FONT SIZE=3D2>tu-sawada@kddi.com</FONT>
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Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org  Wed Nov  3 12:35:25 2004
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Date: Tue, 02 Nov 2004 10:35:54 +0100
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: Sheetal Khemani <skhemani@cisco.com>
Subject: Re: [Sip] SIP AMR Codec (RFC3267) out-of-band bitrate change
References: <Pine.GSO.4.58.0411011357160.23116@lookout.cisco.com>
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Hi,

There is no explicit out-of-band signalling for mode changes defined for 
  the RTP payload format for AMR. Mode changes within the active 
mode-set is expected to be made in two different ways.

1. The sender already posses the information that it needs to switch and 
simply does this. The main information source is expected to congestion 
control. Thus information from RTCP or DCCP can be used to allow the 
sender to determine the need for reducing or increasing the rate.

2. The inband Code Mode Request (CMR) field. This field is intended to 
be used in gateway cases. For example a session between a IP client and 
a circuit switched mobile client. The rules and recommendation on how 
the CMR field should be used are present in section 3.9 of RFC 3267.

I would also like to clarify that a sender is able to at any point 
switch codec mode (within the active mode-set) during a session, no 
external signalling is needed.

So to be able to change the bit-rate there is only the need to know that 
  you need to change the rate. Do you have a need for indication that is 
based on external indication, and not based on available bit-rate?

Cheers

Magnus

Sheetal Khemani wrote:

> Hi,
> 
> I want to know if AMR mode changes are supported via SIP/SDP (possibly
> using a new off in a reINVITE).
> 
> RFC3267 doesn't mention any parameter in the SDP to change the bitrate
> during the call using SIP. It allows the mode-set parameter that tells the
> encoder the subset of modes/bitrates it can use.  It doesn't allow the
> specification of the bitrate the other end needs to use/switch-to.
> 
> I came across some 3GPP docs (3GPP TS 26.236: Packet switched
> conversational multimedia applications; Transport protocols) that suggest
> using the bandwidth paramter, but that seems to be specific to QoS and I
> am not sure if this applies to the SIP standard per se.
> 
> Please let me know if anyone has any insight on this topic.
> 
> Thanks
> Sheetal
> 
> _______________________________________________
> 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
> 


-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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To: "'Jonathan Rosenberg'" <jdrosen@cisco.com>,
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Subject: RE: [Sip] IPR disclosure on MESSAGE and related use of SIP
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After having consulted with Nortel's patent law group on an appropriate
response within the bounds of Nortel corporate policy since I am not a
lawyer and cannot legally represent Nortel's position, I'd like to make the
following statement on this topic and hopefully, we can close the discussion
of this topic. 

US patent number 6,757,732 issued on June 29, 2004. My understanding is that
Nortel became aware of this patent's relevance through a routine periodic
review of its newly issued patents shortly thereafter, informed me of the
patent within a few weeks, and then worked with me to ensure the patent was
properly disclosed to the IETF on July 23, 2004 in compliance with the
applicable IETF policy. 

Nortel Networks considers responsibilities to make appropriate disclosures
in accordance with the revised IETF policy seriously.


Regards,
Mary H. Barnes
mary.barnes@nortelnetworks.com

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Jonathan Rosenberg
Sent: Tuesday, November 02, 2004 8:24 AM
To: Dean Willis
Cc: sip@ietf.org
Subject: Re: [Sip] IPR disclosure on MESSAGE and related use of SIP


Suffice it to say, I am very unhappy about this.

This patent was filed in 2000. I do not understand why the disclosure 
was only made just now, some four years after it was filed and two years 
after issuance of the RFC. The only reasons I can think of are not very 
nice.

The licensing terms are also unclear. Does Nortel plan on extracting 
licensing revenues for this?

I think some clarifications and explanations from participants from 
Nortel are in order.

-Jonathan R.


Dean Willis wrote:

> 
> I've just noticed that Nortel has recently been issued a patent for 
> which they have made an IPR disclosure relative to RFC 3428, the SIP 
> MESSAGE method.
> 
> The disclosure document is listed at:
> 
> http://www.ietf.org/ietf/IPR/nortel-ipr-rfc-3428.txt
> 
> The issued patent is US patent number 6,757,732
> 
> and you can retrieve its full text from:
> 
> http://patft.uspto.gov/netahtml/srchnum.htm
> 
> 
> -- 
> Dean Willis
> SIP co-chair
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.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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charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2658.2">
<TITLE>RE: [Sip] IPR disclosure on MESSAGE and related use of =
SIP</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>After having consulted with Nortel's patent law group =
on an appropriate response within the bounds of Nortel corporate policy =
since I am not a lawyer and cannot legally represent Nortel's position, =
I'd like to make the following statement on this topic and hopefully, =
we can close the discussion of this topic. </FONT></P>

<P><FONT SIZE=3D2>US patent number 6,757,732 issued on June 29, 2004. =
My understanding is that Nortel became aware of this patent's relevance =
through a routine periodic review of its newly issued patents shortly =
thereafter, informed me of the patent within a few weeks, and then =
worked with me to ensure the patent was properly disclosed to the IETF =
on July 23, 2004 in compliance with the applicable IETF policy. =
</FONT></P>

<P><FONT SIZE=3D2>Nortel Networks considers responsibilities to make =
appropriate disclosures in accordance with the revised IETF policy =
seriously.</FONT></P>
<BR>

<P><FONT SIZE=3D2>Regards,</FONT>
<BR><FONT SIZE=3D2>Mary H. Barnes</FONT>
<BR><FONT SIZE=3D2>mary.barnes@nortelnetworks.com</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: sip-bounces@ietf.org [<A =
HREF=3D"mailto:sip-bounces@ietf.org">mailto:sip-bounces@ietf.org</A>] =
On Behalf Of Jonathan Rosenberg</FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, November 02, 2004 8:24 AM</FONT>
<BR><FONT SIZE=3D2>To: Dean Willis</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: Re: [Sip] IPR disclosure on MESSAGE and =
related use of SIP</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Suffice it to say, I am very unhappy about =
this.</FONT>
</P>

<P><FONT SIZE=3D2>This patent was filed in 2000. I do not understand =
why the disclosure </FONT>
<BR><FONT SIZE=3D2>was only made just now, some four years after it was =
filed and two years </FONT>
<BR><FONT SIZE=3D2>after issuance of the RFC. The only reasons I can =
think of are not very </FONT>
<BR><FONT SIZE=3D2>nice.</FONT>
</P>

<P><FONT SIZE=3D2>The licensing terms are also unclear. Does Nortel =
plan on extracting </FONT>
<BR><FONT SIZE=3D2>licensing revenues for this?</FONT>
</P>

<P><FONT SIZE=3D2>I think some clarifications and explanations from =
participants from </FONT>
<BR><FONT SIZE=3D2>Nortel are in order.</FONT>
</P>

<P><FONT SIZE=3D2>-Jonathan R.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Dean Willis wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I've just noticed that Nortel has recently been =
issued a patent for </FONT>
<BR><FONT SIZE=3D2>&gt; which they have made an IPR disclosure relative =
to RFC 3428, the SIP </FONT>
<BR><FONT SIZE=3D2>&gt; MESSAGE method.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The disclosure document is listed at:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"http://www.ietf.org/ietf/IPR/nortel-ipr-rfc-3428.txt" =
TARGET=3D"_blank">http://www.ietf.org/ietf/IPR/nortel-ipr-rfc-3428.txt</=
A></FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The issued patent is US patent number =
6,757,732</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; and you can retrieve its full text from:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"http://patft.uspto.gov/netahtml/srchnum.htm" =
TARGET=3D"_blank">http://patft.uspto.gov/netahtml/srchnum.htm</A></FONT>=

<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -- </FONT>
<BR><FONT SIZE=3D2>&gt; Dean Willis</FONT>
<BR><FONT SIZE=3D2>&gt; SIP co-chair</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>

<P><FONT SIZE=3D2>-- </FONT>
<BR><FONT SIZE=3D2>Jonathan D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 600 Lanidex Plaza</FONT>
<BR><FONT SIZE=3D2>Director, Service Provider VoIP =
Architecture&nbsp;&nbsp; Parsippany, NJ 07054-2711</FONT>
<BR><FONT SIZE=3D2>Cisco Systems</FONT>
<BR><FONT =
SIZE=3D2>jdrosen@cisco.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
FAX:&nbsp;&nbsp; (973) 952-5050</FONT>
<BR><FONT SIZE=3D2><A HREF=3D"http://www.jdrosen.net" =
TARGET=3D"_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) =
952-5000</FONT>
<BR><FONT SIZE=3D2><A HREF=3D"http://www.cisco.com" =
TARGET=3D"_blank">http://www.cisco.com</A></FONT>
</P>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>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>This list is for NEW development of the core SIP =
Protocol</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip</FONT>
<BR><FONT SIZE=3D2>Use sipping@ietf.org for new developments on the =
application of sip</FONT>
</P>

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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
--===============1278425503==--



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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'Robert Sparks'" <RjS@xten.com>, sip@ietf.org
Subject: RE: [Sip] LC on nit-actions and nit-problems
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Hi Robert,

Other than the few "nits" on the drafts that I identify below, I think these
drafts are ready to publish:

nit-problems:
-------------
- the term NIT isn't really defined before it's first use(obviously, it's
clear from the document, but...).  I would suggest add "(NIT)" right after
the first occurrence of "non-INVITE transaction" in section 1. 
- even more obviously, you'll need to update your contact info. 

nit-actions:
------------
- ditto the two points above. 

Regards,
Mary


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Robert
Sparks
Sent: Friday, October 29, 2004 1:29 PM
To: sip@ietf.org
Cc: Dean Willis
Subject: [Sip] LC on nit-actions and nit-problems


Folks -

WGLC for the nit actions and problems draft started Oct 20 and is 
scheduled to
end November 3 (next Wednesday).

So far I have received _no_ comments.

I believe that's because these drafts have been thoroughly discussed, 
are well
understood, and they are ready to go (the points where more discussion 
is needed
have all been moved to nit-future).

If you agree, please take a moment to send a note to the list saying "I 
read this draft
and believe it is ready to be published".

If you disagree - now is the time to comment!

Thanks,
RjS



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
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<TITLE>RE: [Sip] LC on nit-actions and nit-problems</TITLE>
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<BODY>

<P><FONT SIZE=3D2>Hi Robert,</FONT>
</P>

<P><FONT SIZE=3D2>Other than the few &quot;nits&quot; on the drafts =
that I identify below, I think these drafts are ready to =
publish:</FONT>
</P>

<P><FONT SIZE=3D2>nit-problems:</FONT>
<BR><FONT SIZE=3D2>-------------</FONT>
<BR><FONT SIZE=3D2>- the term NIT isn't really defined before it's =
first use(obviously, it's clear from the document, but...).&nbsp; I =
would suggest add &quot;(NIT)&quot; right after the first occurrence of =
&quot;non-INVITE transaction&quot; in section 1. </FONT></P>

<P><FONT SIZE=3D2>- even more obviously, you'll need to update your =
contact info. </FONT>
</P>

<P><FONT SIZE=3D2>nit-actions:</FONT>
<BR><FONT SIZE=3D2>------------</FONT>
<BR><FONT SIZE=3D2>- ditto the two points above. </FONT>
</P>

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

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: sip-bounces@ietf.org [<A =
HREF=3D"mailto:sip-bounces@ietf.org">mailto:sip-bounces@ietf.org</A>] =
On Behalf Of Robert Sparks</FONT>
<BR><FONT SIZE=3D2>Sent: Friday, October 29, 2004 1:29 PM</FONT>
<BR><FONT SIZE=3D2>To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Cc: Dean Willis</FONT>
<BR><FONT SIZE=3D2>Subject: [Sip] LC on nit-actions and =
nit-problems</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Folks -</FONT>
</P>

<P><FONT SIZE=3D2>WGLC for the nit actions and problems draft started =
Oct 20 and is </FONT>
<BR><FONT SIZE=3D2>scheduled to</FONT>
<BR><FONT SIZE=3D2>end November 3 (next Wednesday).</FONT>
</P>

<P><FONT SIZE=3D2>So far I have received _no_ comments.</FONT>
</P>

<P><FONT SIZE=3D2>I believe that's because these drafts have been =
thoroughly discussed, </FONT>
<BR><FONT SIZE=3D2>are well</FONT>
<BR><FONT SIZE=3D2>understood, and they are ready to go (the points =
where more discussion </FONT>
<BR><FONT SIZE=3D2>is needed</FONT>
<BR><FONT SIZE=3D2>have all been moved to nit-future).</FONT>
</P>

<P><FONT SIZE=3D2>If you agree, please take a moment to send a note to =
the list saying &quot;I </FONT>
<BR><FONT SIZE=3D2>read this draft</FONT>
<BR><FONT SIZE=3D2>and believe it is ready to be =
published&quot;.</FONT>
</P>

<P><FONT SIZE=3D2>If you disagree - now is the time to comment!</FONT>
</P>

<P><FONT SIZE=3D2>Thanks,</FONT>
<BR><FONT SIZE=3D2>RjS</FONT>
</P>
<BR>
<BR>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>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>This list is for NEW development of the core SIP =
Protocol</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip</FONT>
<BR><FONT SIZE=3D2>Use sipping@ietf.org for new developments on the =
application of sip</FONT>
</P>

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On Nov 3, 2004, at 9:17 AM, Tom Taylor wrote:

> I'm probably going to get into trouble for responding, but I can't sit 
> by and let you throw mud at my colleagues.
>
Don appropriate lead BVDs Tom.

> Jonathan, you once worked for a large company.  Now you do, again.  
> Can you really say you know everything that's going on in Cisco that 
> might relate to your ETF activities?
>
No, not for any single individual, but there is a process for IPR 
disclosures at Cisco, and it is followed rigorously and taken VERY 
seriously. In fact, you can't even file a Patent disclosure into our 
patent system without filling in a statement as to whether the IPR 
might apply to a standard and if so which one(s). As a check/balance, 
each of our patent committees has at least one member who is up to date 
on standardization activities in that technology area.

It is particularly troubling when such disclosures appear on 
submarines. In order to support your point, I believe you have to 
demonstrate that none of the co-inventors, nor any of the internal 
reviewers in the line to filing and issuance, were aware of the ongoing 
IETF activities.

> It's getting tougher as we get slimmed down, but Nortel (and BNR 
> before that) has had a long tradition of people in odd corners 
> exploring the possibilities of technology and coming up with new ideas 
> unrelated to their current products.  And like it or not, Nortel views 
> intellectual property as a source of value, so people are encouraged 
> to take out patents on new ideas.
>
If that is the case here, 'll be happy to cut some slack. Could you 
ascertain, please?

> Mary was unaware of the patent concerned until a week before the 
> patent announcement.  At that point she took the necessary steps to 
> have it announced.
>
Thank you, Mary. What about the other Nortel IETF participants who work 
with SIP?

> And we who participate at the IETF are not patent lawyers -- we can 
> only refer you to our own legal people for further advice.

Thanks. Dave.

> Jonathan Rosenberg wrote:
>> Suffice it to say, I am very unhappy about this.
>> This patent was filed in 2000. I do not understand why the disclosure 
>> was only made just now, some four years after it was filed and two 
>> years after issuance of the RFC. The only reasons I can think of are 
>> not very nice.
>> The licensing terms are also unclear. Does Nortel plan on extracting 
>> licensing revenues for this?
>> I think some clarifications and explanations from participants from 
>> Nortel are in order.
>> -Jonathan R.
>
>
> -- 
> Tom Taylor
> Carrier VoIP Standards Development
> Nortel Networks
> Phone +1 613 763 1496  (ESN 393-1496)
> E-mail: taylor@nortelnetworks.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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Jon,

Just catching up on my reading. A few questions/comments about the draft:

There is allusion to using this mechanism with redirect servers. But 
there are no examples of this. At least one small thing might not work 
great in such a case. The authentication service is required to add a 
Date header if not already present, and then use in in the signature. 
This won't work well for a redirect lacking Date.

  It may have need/requirement to use some other server as first hop, 
and may or may not even have way to know which one is the auth 
server.Requirement for UA to directly connect to auth server seems 
unrealistic. And requirement to send auth request over a tls connection 
seems insufficient if there are other hops. UA can't really be sure that 
the auth server with be the first hop. And it can't necessarily use sips 
to guarantee hop by hop security till it gets there. How about having 
the *UA* generate an identity header (or variant) signed somehow with 
its digest key to protect the message on the say to the auth server? 
(I'm not a crypto guy - maybe there is no way to make this work.)

You strongly discourage use of tel: in from:. But there are advantages 
of tel: - that recipient can render it as a phone number, which isn't 
really kosher if it is a sip url, even with user=phone. And return calls 
to it are obligated to use sip via the stated domain, while with tel the 
caller has more options.

Hash includes digit and method portions of CSeq. Digit portion could 
have leading zeros. As it passes thru other intermediate nodes it is 
possible this would be parsed and regenerated, losing the leading zeros. 
To avoid, value used here should drop redundant leading zeros.

Above are really small stuff. This is looking quite feasible now.

	Paul




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To: Tom Taylor <taylor@nortelnetworks.com>
Subject: Re: [Sip] IPR disclosure on MESSAGE and related use of SIP
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Tom Taylor wrote:

> I'm probably going to get into trouble for responding, but I can't sit 
> by and let you throw mud at my colleagues.

Mud? Where is there mud? I have asked what I think are legitimate 
questions about this patent - why was the disclosure made only now, and 
what are the terms. I have no doubt that you would ask the same if our 
roles were reversed.


> 
> And we who participate at the IETF are not patent lawyers -- we can only 
> refer you to our own legal people for further advice.

One of the things that has driven success for SIP (and the resulting 
business for us all) has been its relatively IPR-free status for the 
core specifications (in terms of the known disclosures made to IETF; I 
can't say anything about patents we don't know about that might be out 
there). MESSAGE is fairly core to SIP, and I would hate to see its usage 
and deployment be derailed as people run from it due to very unclear IPR 
applicability and licensing terms. I suspect that any strong drive to 
extract revenue from it would just result in quicker abandonment and 
usage of MSRP instead. As such, in the end such an attempt would equate 
to an RF license. While at dynamicsoft, I pushed for us to adopt a 
"don't bug us and we won't bug you" stance 
(http://www.ietf.org/ietf/IPR/DYNAMICSOFT-SIP-UNIFY.txt), which brings 
some of the benefits of RF (doesn't scare folks off) but preserves the 
defensive value of the IPR. Cisco has made similar statements 
(http://www.ietf.org/ietf/IPR/cisco-ipr-draft-ietf-nemo-basic-support-03.txt). 
I hope you agree that this kind of stance is good for us all, and if you 
could help your legal folks understand this and make a similar 
statement, it would be outstanding. I'd be happy to help with that in 
any way that I can.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From sip-bounces@ietf.org  Thu Nov  4 01:26:25 2004
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Subject: RE: [Sip] IPR disclosure on MESSAGE and related use of SIP
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For what it is worth, there may be prior art here. Billy Biggs demonstrated
an implementation of MESSAGE based IM at the 4th SIP bake-off hosted by 3Com
(April 17th-19th 2000) and had an even earlier implementation.

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Srini
Krishnamoorthy
Sent: Tuesday, November 02, 2004 8:34 AM
To: Jonathan Rosenberg
Cc: sip@ietf.org; Dean Willis
Subject: Re: [Sip] IPR disclosure on MESSAGE and related use of SIP


Jonathan et.al,

  Agree, and I think a LOT of people will be unhappy too ..

  Do we read this as specific to the use of "INFO + HTML body" or
  in general :-
   "communicating the text-based chat message includes embedding the 
    text-based chat message" (which the the first claim states)?"

  How does this IPR affect the use of MESSAGE method with content-types
  other than text/html  (say plain, xml) - what would be considered
  infringment? .. anybody Nortel??

  Please help understand the detailed impact of this ... thanks.

  Or should I just wait to see what transpires?? 

-Srini




On Tue, 2 Nov 2004, Jonathan Rosenberg wrote:

jdrose>Suffice it to say, I am very unhappy about this.
jdrose>
jdrose>This patent was filed in 2000. I do not understand why the 
jdrose>disclosure was only made just now, some four years after it was 
jdrose>filed and two years after issuance of the RFC. The only reasons I 
jdrose>can think of are not very nice.
jdrose>
jdrose>The licensing terms are also unclear. Does Nortel plan on 
jdrose>extracting licensing revenues for this?
jdrose>
jdrose>I think some clarifications and explanations from participants 
jdrose>from Nortel are in order.
jdrose>
jdrose>-Jonathan R.
jdrose>
jdrose>
jdrose>Dean Willis wrote:
jdrose>
jdrose>> 
jdrose>> I've just noticed that Nortel has recently been issued a patent 
jdrose>> for which they have made an IPR disclosure relative to RFC 
jdrose>> 3428, the SIP MESSAGE method.
jdrose>> 
jdrose>> The disclosure document is listed at:
jdrose>> 
jdrose>> http://www.ietf.org/ietf/IPR/nortel-ipr-rfc-3428.txt
jdrose>> 
jdrose>> The issued patent is US patent number 6,757,732
jdrose>> 
jdrose>> and you can retrieve its full text from:
jdrose>> 
jdrose>> http://patft.uspto.gov/netahtml/srchnum.htm
jdrose>> 
jdrose>> 
jdrose>> --
jdrose>> Dean Willis
jdrose>> SIP co-chair
jdrose>> 
jdrose>> _______________________________________________
jdrose>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
jdrose>> This list is for NEW development of the core SIP Protocol Use 
jdrose>> sip-implementors@cs.columbia.edu for questions on current sip 
jdrose>> Use sipping@ietf.org for new developments on the application of 
jdrose>> sip
jdrose>> 
jdrose>
jdrose>

--
Srini K
ksrini at motorola
Finger ksrini@bulb.corp.mot.com for pub key.


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Thanks for the close reading, Paul - I know you tend to unearth a lot of
details that others miss, so I especially appreciate your taking a look at
this. A few responses inline:

> There is allusion to using this mechanism with redirect servers. But 
> there are no examples of this. At least one small thing might not work 
> great in such a case. The authentication service is required to add a 
> Date header if not already present, and then use in in the signature. 
> This won't work well for a redirect lacking Date.
> 

Well, I've been roughly assuming that the RFC3261 19.1.5 procedures could be
used to insert a Date header, if needed, but that hasn't been articulated in
the draft.

e.g.: sips:alice@atlanta.example.com?Date=...

There's some ugly escaping that would be required to make that work, but I
think it's doable. Of course, it isn't documented, but as you point out,
there is merely an allusion to this today. Ultimately, I intended to take
this up as future work, given some community interest.

>   It may have need/requirement to use some other server as first hop, 
> and may or may not even have way to know which one is the auth 
> server. Requirement for UA to directly connect to auth server seems 
> unrealistic. 

Personally, I have yet to use SIP in any environment in which I have been
forced to go through some proxy as a first hop, where first hop wouldn't
reasonably be the party that should provide identity services for me. What
the draft says now, essentially, is that the mechanism is only applicable to
scenarios where a UA can connect directly to an auth service. While
long-term deployment experience may prove this scope wise or foolish, the
design choice here hasn't been made capriciously - given the security tools
available, if the UA cannot connect directly to the auth service, then we
encounter serious questions about the efficacy of the mechanism (see below).

The manner in which the UAC might determine that it is talking to an auth
service is, however, pretty straightforward. Considering the Digest
authentication case, from the UAC perspective, all it cares about is the
challenge realm. If the challenge realm matches the domain of the UAC's AoR,
and the cert acquired by TLS matches the challenge realm, then as far as the
UAC is concerned, it should authenticate itself to the server (and this is
all just standard RFC3261 behavior). Strictly speaking, there are
architectures where these conditions might be satisfied, but the auth
service would not actually be the first-hop proxy - in these cases, though,
the auth service and the first-hop proxy would be components of a common
administrative domain, and effectively would be acting as a distributed auth
service (though this is undocumented today). This loophole arises because
this mechanism is intended to be useful opportunistically - that is, without
the UAC even being aware that it is talking to an auth service. Accordingly,
the draft makes few assumptions about how UACs behave other than those
already made in RFC3261. So, ultimately it is up to the auth service itself
whether or not it adds an Identity header.

Anyway, in many environments where I imagine one might be forced to go
through a particular proxy as a first hop, that proxy is probably a good
candidate to instantiate the auth service role. If the environment of the
UAC forces a first-hop proxy, and doesn't provide auth services appropriate
for the UAC, then the environment blocks the UAC from using an identity
service. The fact that this is possible is not, I think, a strong argument
to reconsider the mechanism; in some environments, this may be the desired
effect, even.

Of course, it is also the case the UAC can act as an auth service in the
draft as written; in very constrained environments, this may be the best way
to get an Identity header.

> And requirement to send auth request over a tls connection 
> seems insufficient if there are other hops. UA can't really be sure that 
> the auth server with be the firs hopt. And it can't necessarily use sips 
> to guarantee hop by hop security till it gets there. 

Agreed, but I think it would be closer to the truth to say that the UA and
auth service cannot authenticate one another unless they are one hop apart
(I also don't think SIPS will help much). While the identity mechanism may
not be applicable to all deployment possibilities, I think it is useful as
it stands.

> How about having 
> the *UA* generate an identity header (or variant) signed somehow with 
> its digest key to protect the message on the say to the auth server? 
> (I'm not a crypto guy - maybe there is no way to make this work.)
> 

There is some merit to that idea. I mean, essentially that is what the
Proxy-Authorization header can provide (though auth-int wouldn't guarantee
integrity over the same set of headers). What you want to avoid is cases
where the UAC cannot authenticate the auth service, and a rogue challenger
launches some sort of dictionary attack to discover the Digest credentials
once it has a Proxy-Authorization header in hand. I'm not sure there's a way
to structure a UAC-generated digest for this purpose that wouldn't suffer
from this concern. To get there, we'd certainly be reinventing the majority
of what Proxy-Auth headers do. Hence the direct TLS, and the challenge-realm
matching, which insures that the UAC can authenticate the auth service. Just
handing out your digested password to any transitively-trusted proxy that
challenges you is potentially risky.

> You strongly discourage use of tel: in from:. But there are advantages 
> of tel: - that recipient can render it as a phone number, which isn't 
> really kosher if it is a sip url, even with user=phone. And return calls 
> to it are obligated to use sip via the stated domain, while with tel the 
> caller has more options.
> 

The reasons for discouraging the use of tel are given in the draft, and are
compelling, I think. As a recipient verifying the identity header, why
should you trust, say, cisco.com to authorize the use of the telephone
number '+15165551001'? There's certainly no a priori way for a recipient to
determine whether or not cisco.com is entitled to make any such assertion.
It is very clear, though, why you would trust cisco.com to authorize the use
of the name paul.kyzviat@cisco.com.

There's no question that it would be useful to provide identity services for
telephone numbers. While the draft discourages their use, it doesn't forbid
it, and it leaves the door open for subsequent work to define something
here. A number of theories are circulating (like using ENUM in some
fashion), but it'll take a while to shake out.

> Hash includes digit and method portions of CSeq. Digit portion could 
> have leading zeros. As it passes thru other intermediate nodes it is 
> possible this would be parsed and regenerated, losing the leading zeros. 
> To avoid, value used here should drop redundant leading zeros.

Good catch. Will incorporate it.

> Above are really small stuff. This is looking quite feasible now.
> 

Great!

Jon Peterson
NeuStar, Inc.

> 	Paul
> 
> 
> 

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I already asked this in sip-implementors, but I didn't received any 
conclusive answer.

RFC3261 is not clear about when an UAS must be ready to receive a new 
reINVITE in a dialog.

My reading of the RFC is that the UAS must be ready to receive a new INVITE 
in a dialog as soon a response was sent for the previous INVITE. More 
precisely, that the UAS must not wait for the ACK before accepting the next 
INVITE. Accepting in this case means that the INVITE is passed to the 
application - the INVITE can be rejected by the application because of the 
content of the SDP, but not because of a violation of the signalling protocol.

My reasoning is based on the fact that RFC3261 section 14.1 says that "If 
there is an outgoing INVITE client transaction, the TU must wait until the 
transaction reaches the completed or terminated state before initiating the 
new INVITE." As an INVITE transaction reaches the completed or terminated 
state as soon a non-provisional response is received, the next INVITE can 
be sent at the same time the ACK for the 200 response is sent. As the 
packet ordering can change, the INVITE can arrive at the UAS before the 
ACK. So this mean that the server should be prepared to receive the next 
INVITE before receiving the ACK.

Is it correct?

Please note that my question is not about how SIP implementations work in 
this case, but really on what was RFC3261's authors intent.

Thank you.

-- 
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: marc@8x8.com

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From sip-bounces@ietf.org  Thu Nov  4 09:15:12 2004
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From: "Brett Tate" <brett@broadsoft.com>
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Subject: RE: [Sip] reINVITE
Date: Thu, 4 Nov 2004 09:05:10 -0500
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> So this mean that the server should 
> be prepared to receive the next 
> INVITE before receiving the ACK.
> 
> Is it correct?

It should be prepared; however the re-INVITE can be rejected.  One of the
reasons could be associated with the need of an SDP within the ACK.  Other
reasons could be associated service interactions or product limitations
concerning race conditions.  Thus the uac should be prepared to receive 500
response with a retry-after.



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From sip-bounces@ietf.org  Thu Nov  4 09:23:27 2004
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Subject: RE: [Sip] reINVITE
Date: Thu, 4 Nov 2004 09:14:02 -0500
Message-ID: <8019F8581B5FC14DBD94BAF86844868E352DA3@viridis.IPeria.local>
Thread-Topic: [Sip] reINVITE
Thread-Index: AcTCcISncZynQY/sRMCV9igPqM+GpgABy13g
From: "Gordon Ledgard" <gledgard@iperia.com>
To: "Marc Petit-Huguenin" <petithug@acm.org>, <sip@ietf.org>
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Perhaps someone will chime in and correct me if I'm wrong. I believe
I ran into the same issue a while back, concluding that if another
INVITE were accepted before the ACK to the previous invite arrived
it could cause rather serious instability problems in session=20
negotiations, not to mention trying to align ACKS.

The answer is that you should not accept a new INVITE until the previous
INVITE transaction is complete. Complete means the ACK was seen. I
agree the spec is a bit vague there.

Gordon R. Ledgard
IPeria, Inc.



-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Marc Petit-Huguenin
Sent: Thursday, November 04, 2004 8:04 AM
To: sip@ietf.org
Subject: [Sip] reINVITE


I already asked this in sip-implementors, but I didn't received any=20
conclusive answer.

RFC3261 is not clear about when an UAS must be ready to receive a new=20
reINVITE in a dialog.

My reading of the RFC is that the UAS must be ready to receive a new =
INVITE=20
in a dialog as soon a response was sent for the previous INVITE. More=20
precisely, that the UAS must not wait for the ACK before accepting the =
next=20
INVITE. Accepting in this case means that the INVITE is passed to the=20
application - the INVITE can be rejected by the application because of =
the=20
content of the SDP, but not because of a violation of the signalling =
protocol.

My reasoning is based on the fact that RFC3261 section 14.1 says that =
"If=20
there is an outgoing INVITE client transaction, the TU must wait until =
the=20
transaction reaches the completed or terminated state before initiating =
the=20
new INVITE." As an INVITE transaction reaches the completed or =
terminated=20
state as soon a non-provisional response is received, the next INVITE =
can=20
be sent at the same time the ACK for the 200 response is sent. As the=20
packet ordering can change, the INVITE can arrive at the UAS before the=20
ACK. So this mean that the server should be prepared to receive the next =

INVITE before receiving the ACK.

Is it correct?

Please note that my question is not about how SIP implementations work =
in=20
this case, but really on what was RFC3261's authors intent.

Thank you.

--=20
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: marc@8x8.com

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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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Peterson, Jon wrote:
> Thanks for the close reading, Paul - I know you tend to unearth a lot of
> details that others miss, so I especially appreciate your taking a look at
> this. A few responses inline

>>There is allusion to using this mechanism with redirect servers. But 
>>there are no examples of this. At least one small thing might not work 
>>great in such a case. The authentication service is required to add a 
>>Date header if not already present, and then use in in the signature. 
>>This won't work well for a redirect lacking Date.
> 
> Well, I've been roughly assuming that the RFC3261 19.1.5 procedures could be
> used to insert a Date header, if needed, but that hasn't been articulated in
> the draft.
> 
> e.g.: sips:alice@atlanta.example.com?Date=...
> 
> There's some ugly escaping that would be required to make that work, but I
> think it's doable. Of course, it isn't documented, but as you point out,
> there is merely an allusion to this today. Ultimately, I intended to take
> this up as future work, given some community interest.

OK. If the intent was simply to point the way for future work, then 
fine. That wasn't clear to me.

>>  It may have need/requirement to use some other server as first hop, 
>>and may or may not even have way to know which one is the auth 
>>server. Requirement for UA to directly connect to auth server seems 
>>unrealistic. 
> 
> Personally, I have yet to use SIP in any environment in which I have been
> forced to go through some proxy as a first hop, where first hop wouldn't
> reasonably be the party that should provide identity services for me.

What about the IMS architecture, when you are connected to a visited 
network? The P-CSCF may be able to authenticate you, but it won't be 
able to sign your identity with credentials from your home domain.

(That of course is just an example.)

 > What
> the draft says now, essentially, is that the mechanism is only applicable to
> scenarios where a UA can connect directly to an auth service. While
> long-term deployment experience may prove this scope wise or foolish, the
> design choice here hasn't been made capriciously - given the security tools
> available, if the UA cannot connect directly to the auth service, then we
> encounter serious questions about the efficacy of the mechanism (see below).
> 
> The manner in which the UAC might determine that it is talking to an auth
> service is, however, pretty straightforward. Considering the Digest
> authentication case, from the UAC perspective, all it cares about is the
> challenge realm. If the challenge realm matches the domain of the UAC's AoR,
> and the cert acquired by TLS matches the challenge realm, then as far as the
> UAC is concerned, it should authenticate itself to the server (and this is
> all just standard RFC3261 behavior).

Seems like the challenge realm is a red herring above. If the TLS cert 
matches the domain of my AOR then I know I am one hop away from 
something authoritative for my domain, regardless of the challenge. 
Since there could be many proxies in my domain, the challenge could be 
coming from a different one. In any case I have to forumulate the 
request before being challenged.

 > Strictly speaking, there are
> architectures where these conditions might be satisfied, but the auth
> service would not actually be the first-hop proxy - in these cases, though,
> the auth service and the first-hop proxy would be components of a common
> administrative domain, and effectively would be acting as a distributed auth
> service (though this is undocumented today). This loophole arises because
> this mechanism is intended to be useful opportunistically - that is, without
> the UAC even being aware that it is talking to an auth service. Accordingly,
> the draft makes few assumptions about how UACs behave other than those
> already made in RFC3261. So, ultimately it is up to the auth service itself
> whether or not it adds an Identity header.

Yes, this certainly is a possibility. Maybe it would help to soften the 
stance in the document, by inserting some words like those above.

> Anyway, in many environments where I imagine one might be forced to go
> through a particular proxy as a first hop, that proxy is probably a good
> candidate to instantiate the auth service role. If the environment of the
> UAC forces a first-hop proxy, and doesn't provide auth services appropriate
> for the UAC, then the environment blocks the UAC from using an identity
> service. The fact that this is possible is not, I think, a strong argument
> to reconsider the mechanism; in some environments, this may be the desired
> effect, even.

The IMS case I mentioned above fits this. The UAC is forced to use a 
first hop proxy with credentials for a different (visited) domain. But 
the home domain of the UAC has a business relationship with the visited 
domain and trusts it. But I don't see any way to exploit that 
relationship so that the home proxy could safely sign the request.

I guess I agree that this might be a reason to find that connection 
mechanism to be suspect.

> Of course, it is also the case the UAC can act as an auth service in the
> draft as written; in very constrained environments, this may be the best way
> to get an Identity header.

Sure. But then you are back to needing UAs with certs.

>>And requirement to send auth request over a tls connection 
>>seems insufficient if there are other hops. UA can't really be sure that 
>>the auth server with be the firs hopt. And it can't necessarily use sips 
>>to guarantee hop by hop security till it gets there. 
> 
> Agreed, but I think it would be closer to the truth to say that the UA and
> auth service cannot authenticate one another unless they are one hop apart
> (I also don't think SIPS will help much). While the identity mechanism may
> not be applicable to all deployment possibilities, I think it is useful as
> it stands.

Well, if the identity mechanism can become sufficiently compelling, then 
it might rule out architectures that prevent its use.
Perhaps this is the real motivation. :-)

>>How about having 
>>the *UA* generate an identity header (or variant) signed somehow with 
>>its digest key to protect the message on the say to the auth server? 
>>(I'm not a crypto guy - maybe there is no way to make this work.)
> 
> There is some merit to that idea. I mean, essentially that is what the
> Proxy-Authorization header can provide (though auth-int wouldn't guarantee
> integrity over the same set of headers). What you want to avoid is cases
> where the UAC cannot authenticate the auth service, and a rogue challenger
> launches some sort of dictionary attack to discover the Digest credentials
> once it has a Proxy-Authorization header in hand.  I'm not sure there's a way
> to structure a UAC-generated digest for this purpose that wouldn't suffer
> from this concern. To get there, we'd certainly be reinventing the majority
> of what Proxy-Auth headers do. Hence the direct TLS, and the challenge-realm
> matching, which insures that the UAC can authenticate the auth service. Just
> handing out your digested password to any transitively-trusted proxy that
> challenges you is potentially risky.

As I said, I'm not a crypto guy. I was expecting that there would 
probably be some issue like this.

>>You strongly discourage use of tel: in from:. But there are advantages 
>>of tel: - that recipient can render it as a phone number, which isn't 
>>really kosher if it is a sip url, even with user=phone. And return calls 
>>to it are obligated to use sip via the stated domain, while with tel the 
>>caller has more options.
> 
> The reasons for discouraging the use of tel are given in the draft, and are
> compelling, I think. As a recipient verifying the identity header, why
> should you trust, say, cisco.com to authorize the use of the telephone
> number '+15165551001'?  There's certainly no a priori way for a recipient to
> determine whether or not cisco.com is entitled to make any such assertion.

I agree with the concern, but the need is still there.

> It is very clear, though, why you would trust cisco.com to authorize the use
> of the name paul.kyzviat@cisco.com.
> 
> There's no question that it would be useful to provide identity services for
> telephone numbers. While the draft discourages their use, it doesn't forbid
> it, and it leaves the door open for subsequent work to define something
> here. A number of theories are circulating (like using ENUM in some
> fashion), but it'll take a while to shake out.

Yes, enum does seem like it might provide a solution. Maybe this will 
have to do for now.

>>Hash includes digit and method portions of CSeq. Digit portion could 
>>have leading zeros. As it passes thru other intermediate nodes it is 
>>possible this would be parsed and regenerated, losing the leading zeros. 
>>To avoid, value used here should drop redundant leading zeros.
> 
> Good catch. Will incorporate it.

I looked for other similar issues but didn't find any.

	Paul


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One other thing occurred to me. You point out that there is some 
significant cost to the signing operation. I am thinking that this might 
be done on an as-needed basis. For instance, the proxy could first just 
pass the request along without signing it. If a 428 'Use Identity 
Header' is received in response the request could signed and retried. 
But I think there are problems with the proxy doing the retry on its 
own. It could be done under explicit control of the UAC, but then there 
would need to be a way for the UAC to request that signing be done, or not.

This might be especially interesting in an environment where 
P-Asserted-Identity is often sufficient.

	Paul


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From sip-bounces@ietf.org  Thu Nov  4 11:00:32 2004
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Subject: [Sip] WGLC for App Interact
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Hi Folks,

we would like to working group last call the following draft in SIP in  
parallel with SIPPING:

http://www.ietf.org/internet-drafts/draft-ietf-sipping-app-interaction- 
framework-03.txt

This WGLC will finish on December 3rd. Please, send you comments to the  
authors and to the *SIP* list.

thanks,
-rohan


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From sip-bounces@ietf.org  Thu Nov  4 12:43:21 2004
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Subject: [Sip] Invitation to register to the 2nd ENUM Workshop at ETSI on
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Invitation to register to the 2nd ENUM Workshop at ETSI on Tuesday 30 =
November 2004

Dear All,

2nd ENUM Technical Workshop Event description & information
www.etsi.org/plugtests/ENUM.htm

Agenda
www.etsi.org/plugtests/Upcoming/ENUM/ENUMAgenda.htm

Registration
webapp.etsi.org/meetingcalendar/MeetingDetails.asp?mid=3D24578=20

Please forward in your preferred lists.
Best Regards

Patrick GUILLEMIN=20
ETSI - Plugtests Technical Coordinator=20
www.etsi.org/plugtests




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From sip-bounces@ietf.org  Thu Nov  4 19:08:36 2004
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Date: Thu, 04 Nov 2004 16:04:18 -0800
From: Marc Petit-Huguenin <petithug@acm.org>
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To: Gordon Ledgard <gledgard@iperia.com>
Subject: Re: [Sip] reINVITE
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As the offer/answer cannot overlap, there is 4 different cases:

1.

UAC             UAS
  --INVITE offer->
  <---200 answer--
  --ACK---------->
  --INVITE------->
  <----200 offer--
  --ACK answer--->

2.

UAC             UAS
  --INVITE offer->
  <---200 answer--
  --ACK---------->
  --INVITE offer->
  <---200 answer--
  --ACK---------->

3.

UAC             UAS
  --INVITE------->
  <----200 offer--
  --ACK answer--->
  --INVITE------->
  <----200 offer--
  --ACK answer--->

4.

UAC             UAS
  --INVITE------->
  <----200 offer--
  --ACK answer--->
  --INVITE offer->
  <---200 answer--
  --ACK---------->

The only problem here is the fourth case, because the INVITE offer can
arrive before the previous answer.


Gordon Ledgard wrote:
> 
> Perhaps someone will chime in and correct me if I'm wrong. I believe
> I ran into the same issue a while back, concluding that if another
> INVITE were accepted before the ACK to the previous invite arrived
> it could cause rather serious instability problems in session 
> negotiations, not to mention trying to align ACKS.
> 
> The answer is that you should not accept a new INVITE until the previous
> INVITE transaction is complete. Complete means the ACK was seen. 

No, the ACK for the 2xx is not part of the INVITE transaction. The 
transaction is complete as soon the 2xx is sent. See Figure 7 in RFC3261.

> I
> agree the spec is a bit vague there.
> 
> Gordon R. Ledgard
> IPeria, Inc.
> 
> 
> 
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
> Marc Petit-Huguenin
> Sent: Thursday, November 04, 2004 8:04 AM
> To: sip@ietf.org
> Subject: [Sip] reINVITE
> 
> 
> I already asked this in sip-implementors, but I didn't received any 
> conclusive answer.
> 
> RFC3261 is not clear about when an UAS must be ready to receive a new 
> reINVITE in a dialog.
> 
> My reading of the RFC is that the UAS must be ready to receive a new INVITE 
> in a dialog as soon a response was sent for the previous INVITE. More 
> precisely, that the UAS must not wait for the ACK before accepting the next 
> INVITE. Accepting in this case means that the INVITE is passed to the 
> application - the INVITE can be rejected by the application because of the 
> content of the SDP, but not because of a violation of the signalling protocol.
> 
> My reasoning is based on the fact that RFC3261 section 14.1 says that "If 
> there is an outgoing INVITE client transaction, the TU must wait until the 
> transaction reaches the completed or terminated state before initiating the 
> new INVITE." As an INVITE transaction reaches the completed or terminated 
> state as soon a non-provisional response is received, the next INVITE can 
> be sent at the same time the ACK for the 200 response is sent. As the 
> packet ordering can change, the INVITE can arrive at the UAS before the 
> ACK. So this mean that the server should be prepared to receive the next 
> INVITE before receiving the ACK.
> 
> Is it correct?
> 
> Please note that my question is not about how SIP implementations work in 
> this case, but really on what was RFC3261's authors intent.
> 
> Thank you.
> 


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Subject: [Sip] WiMAX Summit: Call for proposals
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. What is the business model for WiMAX? 
. What do we learn from earlier deployments? 
. What about the future extensions of the standard? 
. How is addressed the interoperability challenge? 

These questions, among others, will be addressed during the second edition
of the WiMAX Summit, next April 5-8 2005, by distinguished experts and key
players in the field.

 

The call for proposal dead line has been extended to November 30.

 

Details at:

 <http://www.upperside.fr/wimax05/wimax2005intro.htm>
http://www.upperside.fr/wimax05/wimax2005intro.htm

 


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<p class=3DMsoNormal><strong><b><font size=3D2 face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial'>&#8226; =
</span></font></b></strong><font
size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:10.0pt;font-family:Arial'>What
is the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>business
model</span></font></b></strong> for WiMAX? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>What
do we learn from <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>earlier
deployments</span></font></b></strong>? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>What
about the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>future
extensions</span></font></b></strong> of the standard? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>How
is addressed the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>interoperability</span></font></b></strong>
challenge? <br>
<br>
These questions, among others, will be addressed during the second =
edition of
the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>WiMAX Summit,
next April 5-8 2005</span></font></b></strong>,&nbsp;by distinguished =
experts
and key players in the field.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>The call for proposal dead line has been =
extended to
November 30.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>Details at:<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'><a =
href=3D"http://www.upperside.fr/wimax05/wimax2005intro.htm"
title=3D"http://www.upperside.fr/wimax05/wimax2005intro.htm"><span =
lang=3DEN-GB>http://www.upperside.fr/wimax05/wimax2005intro.htm</span></a=
></span></font><font
size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p=
>

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

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From: "Elwell, John" <john.elwell@siemens.com>
To: "'jon.peterson@neustar.biz'" <jon.peterson@neustar.biz>
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Subject: [Sip] Comments on draft-ietf-sip-identity-03
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Jon,

A couple of comments:

1. The last requirement "It must be possible, in cases where a request has
been retargeted to a different AoR than the one designated in the To header
field, for the UAC to ascertain the AoR to which the request has been sent"
seems to a  remnant from the previous draft. I believe it relates to
identity in responses.

2. It would be useful to include some statements about how the new mechanism
might be used in situations where the UAS is a gateway to a legacy network.
Here the gateway may provide an authentication service for numbers supplied
by the legacy network, e.g.,
for +441159434989@gateway.example.com.
Alternatively a proxy that has authenticated the gateway might provide an
authentication service, e.g., for +441159434989@example.com.

John (john.elwell@siemens.com)


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<P><FONT SIZE=2 FACE="Arial">Jon,</FONT>
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<P><FONT SIZE=2 FACE="Arial">A couple of comments:</FONT>
</P>

<P><FONT SIZE=2 FACE="Arial">1. The last requirement &quot;It must be possible, in cases where a request has been retargeted to a different AoR than the one designated in the To header field, for the UAC to ascertain the AoR to which the request has been sent&quot; seems to a&nbsp; remnant from the previous draft. I believe it relates to identity in responses.</FONT></P>

<P><FONT SIZE=2 FACE="Arial">2. It would be useful to include some statements about how the new mechanism might be used in situations where the UAS is a gateway to a legacy network. Here the gateway may provide an authentication service for numbers supplied by the legacy network, e.g.,</FONT></P>

<P><FONT SIZE=2 FACE="Arial">for +441159434989@gateway.example.com.</FONT>
<BR><FONT SIZE=2 FACE="Arial">Alternatively a proxy that has authenticated the gateway might provide an authentication service, e.g., for +441159434989@example.com.</FONT></P>

<P><FONT SIZE=2 FACE="Arial">John (john.elwell@siemens.com)</FONT>
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Please note that the it is important to consider the "other half" of
connection reuse, and I've added the Jenning's outbound draft to the
readling list.<br>
<br>
This modified the following agenda item:<br>
<br>
+15&nbsp; &nbsp;&nbsp;&nbsp; Connection Reuse<br>
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp; &nbsp;&nbsp; &nbsp; Rohan Mahy, Cullen
Jennings<br>
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp; &nbsp;&nbsp; &nbsp; <a
 href="http://www.ietf.org/internet-drafts/draft-ietf-sip-connect-reuse-03.txt">draft-ietf-sip-connect-reuse-03.txt</a><br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a
 href="http://www.ietf.org/internet-drafts/draft-jennings-sipping-outbound-00.txt">draft-jennings-sipping-outbound-00.txt</a><br>
<br>
<br>
--<br>
Dean Willis<br>
SIP co-chair<br>
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From sip-bounces@ietf.org  Fri Nov  5 15:10:19 2004
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Subject: [Sip] draft-ietf-sip-refer-with-feature-param-00
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This draft seem innocuous enough, but why do you feel that it is 
necessary? RFC3840 was itself a backward compatible extension, in that 
all the paramters it defines fit the generic-parameter syntax. So it 
seems to me that the desired usage is already present without this 
proposed extension.

	Paul


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
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Subject: [Sip] RE: draft-ietf-sip-identity-03
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Looks like we're on the same page about most of these matters. A few more
notes.

> > Personally, I have yet to use SIP in any environment in which I have
been
> > forced to go through some proxy as a first hop, where first hop wouldn't
> > reasonably be the party that should provide identity services for me.
> 
> What about the IMS architecture, when you are connected to a visited 
> network? The P-CSCF may be able to authenticate you, but it won't be 
> able to sign your identity with credentials from your home domain.
> 

Well, I can't say that I've used SIP in an IMS environment yet, but, your
point is taken. I have a further action item, just as an aside, to describe
interworking between RFC3325 (PAI) environments and sip-identity-0x
environments, once we've agreed on exactly what sip-identity-0x entails. I
think that in many such cases, there might be some sort of edge gateway that
provides the Identity header on behalf of the whole transitively-trusting
federation. 

While I would love to see sip-identity-0x implemented for IMS, as you point
out the whole concept of 'roaming' is somewhat antithetical to the idea that
a user agent associates with their own domain. The fact that some entity
outside your local domain is authenticating you is the first warning sign,
the presence of transitive trust is the second. Roaming is based on
transitive trust. sip-identity-0x is not. Gatewaying may be the best we can
do.

> > If the challenge realm matches the domain of the UAC's AoR,
> > and the cert acquired by TLS matches the challenge realm, then as far as
the
> > UAC is concerned, it should authenticate itself to the server (and this
is
> > all just standard RFC3261 behavior).
> 
> Seems like the challenge realm is a red herring above. If the TLS cert 
> matches the domain of my AOR then I know I am one hop away from 
> something authoritative for my domain, regardless of the challenge. 
> Since there could be many proxies in my domain, the challenge could be 
> coming from a different one. In any case I have to forumulate the 
> request before being challenged.

The realm is significant because it is possible that your request will be
proxied to some other server that issues a challenge for a realm other than
your local domain. The realm tells you, essentially, which credentials you
need to provide. So, the fact that you received a 407 over a TLS connection
doesn't necessarily mean that you are being challenged by your local domain.
Further to this point, you local domain could institute a policy to inspect
the realm of challenges resulting from proxied requests in order to guard
from malicious domains down the pike fishing for passwords in your domain.

> > So, ultimately it is up to the auth service itself
> > whether or not it adds an Identity header.
> 
> Yes, this certainly is a possibility. Maybe it would help to soften the 
> stance in the document, by inserting some words like those above.

Maybe you're right; it could be worthwhile incorporating a little language
along these lines to explain that given the need to rely on opportunistic
UAC behavior, authentication services necessarily have a bit of leeway. It
may help in some architectures, though I'm not sure it'll help a great deal
in others.

> > While the identity mechanism may
> > not be applicable to all deployment possibilities, I think it is useful
as
> > it stands.
> 
> Well, if the identity mechanism can become sufficiently compelling, then 
> it might rule out architectures that prevent its use.
> Perhaps this is the real motivation. :-)
> 

In the Machiavellian sense of pulling a fast one on the community, this
isn't the motivation. But there is a deeper sense in which genuine security
for SIP is possible in some deployment architectures and not possible in
others. We would do ourselves a disservice, from a security perspective, by
building an identity mechanism that is applicable to every imaginable
deployment, since in many imaginable deployments, identity cannot be
asserted securely. I'd rather have the identity assertion when it's
possible, and know that the assertion is meaningful, than allow the
assertion to appear in absolutely every case, but never really know whether
or not it is actually meaningful.

Jon Peterson
NeuStar, Inc.

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Elwell, John'" <john.elwell@siemens.com>
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Subject: [Sip] RE: Comments on draft-ietf-sip-identity-03
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Thanks for the read, John. Agreed that the last requirement is left over
from previous versions and should be removed. 

In terms of your second point, I agree that would be useful. I was planning,
providing that sip-identity-0x passes, on doing yet still more further work
to show how it could be used in PSTN interworking in a separate draft. Some
details of the SIP-ISUP mapping, no doubt, are impacted by this. The focus
of that draft would be on preventing Caller-ID spoofing from SIP-originated
calls (which is the SIP->PSTN case), though I agree there are probably a few
things to say there about the PSTN->SIP case. So, I don't specifically think
we need that to be in this draft, but I agree it needs to be done at some
point.

Jon Peterson
NeuStar, Inc.


-----Original Message-----
From: Elwell, John [mailto:john.elwell@siemens.com]
Sent: Friday, November 05, 2004 1:48 AM
To: 'jon.peterson@neustar.biz'
Cc: 'sip@ietf.org'
Subject: Comments on draft-ietf-sip-identity-03


Jon, 

A couple of comments: 

1. The last requirement "It must be possible, in cases where a request has
been retargeted to a different AoR than the one designated in the To header
field, for the UAC to ascertain the AoR to which the request has been sent"
seems to a  remnant from the previous draft. I believe it relates to
identity in responses.

2. It would be useful to include some statements about how the new mechanism
might be used in situations where the UAS is a gateway to a legacy network.
Here the gateway may provide an authentication service for numbers supplied
by the legacy network, e.g.,

for +441159434989@gateway.example.com. Alternatively a proxy that has
authenticated the gateway might provide an authentication service, e.g., for
+441159434989@example.com.

John (john.elwell@siemens.com) 

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Subject: [Sip] Strict,
	Semi-Strict and Loose mode in RPH - not a good fit for ets and wps
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One of the changes in the draft-ietf-sip-resource-priority-05 is the
definition of 3 modes, strict, semi-strict, and loose, and the assignment
of each defined namespaces to a particular mode.

The definition of the mode involves several distinct behaviors

1 RPH is mandatory vs RPH is optional.
(In strict mode, every relevant SIP message must have the RPH, with the
specific namespace.  In semi-strict and loose mode, some messages may have
the RPH, with the specific namespace, and other messages may have no RPH.)

2 Behavior of a SIP element when it receives an RPH with a namespace it
does not recognize.
(In strict and semi-strict mode, the SIP element returns a 417 error
message if it gets a SIP message with an RPH namespace it does recognize
(or a namespace it does recognize, but an invalid value).  In loose mode,
the SIP element simply ignores namespaces it doesn't recognize - treats it
as if the RPH were not present.  (In loose mode, it will return an error
message if it receives a known namespace with an unknown, or invalid,
value))

3 Content of the error message.
(In strict mode, if the error message contains the "Accept Resource
Priority" Header (which indicates the namespaces and values that WOULD be
accepted) that information is stripped off before it reaches the final
user.  In semi-strict mode and loose mode, it appears that  the information
is not stripped off.)

4  .Inclusion of authorization header (or equivalent functionality from
Trait-Based Authorization).
(In semi-strict mode, SIP messages that include the RPH must also include
the authorization header.  In strict and loose mode, there is no mention of
the authorization header)

5  Preemption vs precedence (i.e., queueing) behavior
(At an IP/PSTN gateway, with all PSTN -side circuits in use, in strict mode
one of the lower priority circuits would be freed using pre-emption.  In
semi strict mode, the SIP request  may be granted access to the next
available circuit based on the header's presence, regardless of how many
other "regular" requests are received at the gateway.  The pre-emption vs.
precedence behavior in loose mode is not described.)

6 Use of the "Require" header field
( in strict and semi-strict modes, a SIP message with the RPH also has the
"Require" header with the "Resource-Priority" option tag.  Loose mode does
no include the Require header)

Of the namespaces described in the document, dsn, drsn and q735 (all
variants of "MLPP") are defined as "strict".  ets and wps are defined as
"semi-strict".  None are defined as loose.

Unfortunately, this  (semi-strict) is NOT the behavior desired for the ets
and wps namespaces.

The behavior desired for the ets and wps namespaces are
1 RPH is optional (semi-strict and loose)
2 SIP element ignores a namespace it does not recognize (loose)
3 The content of the error message should not reveal (to the end user) the
valid values for that namespace (strict)
4 SIP messages that include the RPH with the ets or wps namespaces should
include the authorization header (or equivalent mechanism) (semi-strict)
5 Precedence (queueing) is generally the means of providing priority, but
exemption from network management controls, for instance  call admission
within an IP domain, when a "regular" call would not be admitted, is also
needed.  Pre-emption is not used.  (semi-strict "plus more")
6 There is no a priori requirement to use, or not use, the "Require"
header field.

So the desired behavior is a combination of what is defined for strict,
semi-strict, and loose modes.

It would be much better if one of the modes described the desired behavior
for the ets and wps namespaces.


Thanks,

Janet

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I tired to read this whole thing again and it made my head want to explode -
I think getting a bunch of people to read it again would be really good.
There has been some nice work done on it since the previous rev.




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I think I understand that often the subscriptions is of no use and what you
want to do here. 

It seems like putting a tag in the uri of the refer like ";norefersub" could
be used as a hint not to create the implicit subscription. It does not seem
like the Supported Require stuff is needed.

Would this work and do what you want?

Cullen




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I'm vague on all this stuff but ... Do you need anything normative for this
at all - seems like this is already supported and perhaps we just need some
example call flows demonstrating it.

Cullen



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Subject: Re: [Sip] RE: draft-ietf-sip-identity-03
From: Cullen Jennings <fluffy@cisco.com>
To: Jon Peterson <jon.peterson@neustar.biz>,
        Paul H Kyzivat <pkyzivat@cisco.com>
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On 11/5/04 12:20 PM, "Peterson, Jon" <jon.peterson@neustar.biz> wrote:

>>> Personally, I have yet to use SIP in any environment in which I have
> been
>>> forced to go through some proxy as a first hop, where first hop wouldn't
>>> reasonably be the party that should provide identity services for me.
>> 
>> What about the IMS architecture, when you are connected to a visited
>> network? The P-CSCF may be able to authenticate you, but it won't be
>> able to sign your identity with credentials from your home domain.
>> 

Thought I mostly agree, it might be possible that the roaming operator does
have the private key from the home operator. Once you have theses operators
that trust each other and delegate certain operations back and forth,
something like this might work.




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From: "Ken Carlberg" <carlberg@g11.org.uk>
To: "'Janet P Gunn'" <jgunn6@csc.com>, <sip@ietf.org>, <jmpolk@cisco.com>
Subject: RE: [Sip] Strict,
	Semi-Strict and Loose mode in RPH - not a good fit for ets and wps
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> Unfortunately, this  (semi-strict) is NOT the behavior desired for the ets
> and wps namespaces.
> 
> The behavior desired for the ets and wps namespaces are
> 1 RPH is optional (semi-strict and loose)
> 2 SIP element ignores a namespace it does not recognize (loose)

I would prefer a strict or semi-strict mode because these provide feedback in
case: a) a misconfigured UA sends the wrong namespace.priority, and/or b)
initial selection of a server/proxy that does not support the desired
namespace.  The former is quite possible given an ability (and possibly,
necessity) of the user to configure their own device to support the given
namespace.value combination.

> 3 The content of the error message should not reveal (to the end user) the
> valid values for that namespace (strict)

section 4.5 of the drafts states:

   If the UAS understands the resource value, but refuses to honor the
   request with elevated priority for this particular user, it returns
   the 403 (Forbidden) response code.  It MAY include the list of
   resource values that the user is allowed to use in the
   'Accept-Resource-Priority' response header field.

The "MAY" is the critical term.  My understanding would be that this decision
is a local policy issue.  Outside of that I personally don't see a problem with
providing "hints" of accepted values under the context that they can only acted
upon by authorized users.  Informing the user of the acceptable set of
priorities does not bypass or circumvent the need for proper authentication
credentials.

> 4 SIP messages that include the RPH with the ets or wps namespaces should
> include the authorization header (or equivalent mechanism) (semi-strict)
> 5 Precedence (queueing) is generally the means of providing priority, but
> exemption from network management controls, for instance call admission
> within an IP domain, when a "regular" call would not be admitted, is also
> needed.  Pre-emption is not used.  (semi-strict "plus more")

the "plus more" would seem to be a product of local policy

> 6 There is no a priori requirement to use, or not use, the "Require"
> header field.

isn't this addressed in section 4.3.2 below?  or am I missing something?

   If the request includes a 'Require' header field with the 
   'Resource-Priority' option tag, a UAS MUST follow the strict or 
   semi-strict mode rules, otherwise UAS and proxies MUST operate in 
   loose mode.

-ken


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In a message dated 11/6/2004 12:38:18 AM Eastern Standard Time, 
jgunn6@csc.com writes:


> Of the namespaces described in the document, dsn, drsn and q735 (all
> variants of "MLPP") are defined as "strict".  ets and wps are defined as
> "semi-strict".  None are defined as loose.
> 
> Unfortunately, this  (semi-strict) is NOT the behavior desired for the ets
> and wps namespaces.
> 
> 

Likewise, "strict" is not the behavior desired for dsn, drsn, or q735. 
"Strict" says that it requires the RP header in a bunch of messages. Since these 
three namespaces are based on Q.735 (MLPP), it is expected that the procedures 
should be based on that standard. Therefore, the RP header only goes in the 
INVITE. The procedures do not require it in any other messages.

Further, DSN may choose to allow the absence of any RP header to mean to use 
the lowest (Routine), thus the purpose for specifying a "default level to be 
assumed in the absence of the priority value" as version -01 of this draft 
stated. I don't remember any discussion/agreement to remove this statement about 
"default", but maybe there was. The description of "Loose" mode in 4.3.5 still 
seems to require that there be a known default.

This draft is going way beyond what is required to define a simple header to 
carry a simple piece of information.

Where was the discussion on the list or elsewhere about adding the third 
mode? In the previous draft, as agreed in previous discussions, "strict" vs. 
"loose" modes only pertained to specifying "handling requests with unknown 
namespaces or priority values". This latest version greatly expands the meaning and 
use of this "mode" to specify other aspects of operation, such as relating 
"strict" mode with the use of preemption (section 4.5.1). The previous drafts 
stayed away from specifying the operation for specific cases, since this should be 
the subject of other documents.

In 4.3.4 for Semi-Strict Mode, whether or not the messages "must" contain an 
authorization header or [something unknown from the Trait-based Authorization 
effort] depends on the security architecture of the network in which this is 
being used. There may be networks that use a different approach. It is 
inappropriate for this RP draft to require specific authorization methods. In any 
case, this reference to "trait-based authorization" is normative, not informative.

The basic definitions of "strict" and "loose" modes in version -04 should be 
retained, not all this new material in -05. In fact, all of -04 should be 
accepted as the basis for further discussions, not -05.

Mike Pierce


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/6/2004 12:38:18 AM Eastern Standard Time, jgunn6@csc.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Of the namespaces described=
 in the document, dsn, drsn and q735 (all
<BR>variants of "MLPP") are defined as "strict". &nbsp;ets and wps are defin=
ed as
<BR>"semi-strict". &nbsp;None are defined as loose.
<BR>
<BR>Unfortunately, this &nbsp;(semi-strict) is NOT the behavior desired for=20=
the ets
<BR>and wps namespaces.
<BR>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>Likewise, "strict" is not the behavior desired for dsn, drsn, or q735. "=
Strict" says that it requires the RP header in a bunch of messages. Since th=
ese three namespaces are based on Q.735 (MLPP), it is expected that the proc=
edures should be based on that standard. Therefore, the RP header only goes=20=
in the INVITE. The procedures do not require it in any other messages.
<BR>
<BR>Further, DSN may choose to allow the absence of any RP header to mean to=
 use the lowest (Routine), thus the purpose for specifying a "default level=20=
to be assumed in the absence of the priority value" as version -01 of this d=
raft stated. I don't remember any discussion/agreement to remove this statem=
ent about "default", but maybe there was. The description of "Loose" mode in=
 4.3.5 still seems to require that there be a known default.
<BR>
<BR>This draft is going way beyond what is required to define a simple heade=
r to carry a simple piece of information.
<BR>
<BR>Where was the discussion on the list or elsewhere about adding the third=
 mode? In the previous draft, as agreed in previous discussions, "strict" vs=
. "loose" modes only pertained to specifying "handling requests with unknown=
 namespaces or priority values". This latest version greatly expands the mea=
ning and use of this "mode" to specify other aspects of operation, such as r=
elating "strict" mode with the use of preemption (section 4.5.1). The previo=
us drafts stayed away from specifying the operation for specific cases, sinc=
e this should be the subject of other documents.
<BR>
<BR>In 4.3.4 for Semi-Strict Mode, whether or not the messages "must" contai=
n an authorization header or [something unknown from the Trait-based Authori=
zation effort] depends on the security architecture of the network in which=20=
this is being used. There may be networks that use a different approach. It=20=
is inappropriate for this RP draft to require specific authorization methods=
. In any case, this reference to "trait-based authorization" is normative, n=
ot informative.
<BR>
<BR>The basic definitions of "strict" and "loose" modes in version -04 shoul=
d be retained, not all this new material in -05. In fact, all of -04 should=20=
be accepted as the basis for further discussions, not -05.
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

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From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>, "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] refer-with-feature-param
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Do we need to distinguish the generic-param from the feature-param?

Seems to me that the feature-param is IANA defined (we probably need
to mention it in the IANA section), but the generic-param is not.

So, for example, you can't really know if Refer-To: bof@example.com;audio
is implying the IANA-registered audio as per 3840/this draft or
anybody elses generic parameter which could mean something different.

I'm not sure if it is an issue of not. (I guess it depends if we expect that
the recipient of the REFER is supposed to look at the parameter or not.)

If so, it might be better to have an explicit indication like
;feature-param=audio or
something like that.

If not, then I don't understand why we need to alter the ABNF, because
feature-param is
a generic-param in the first place. 

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Cullen Jennings
> Sent: Friday, November 05, 2004 19:10
> To: sip@ietf.org
> Cc: Orit Levin
> Subject: [Sip] refer-with-feature-param
> 
> 
> 
> I'm vague on all this stuff but ... Do you need anything 
> normative for this at all - seems like this is already 
> supported and perhaps we just need some example call flows 
> demonstrating it.
> 
> Cullen
> 
> 
> 
> _______________________________________________
> 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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<P><FONT SIZE=3D2>Do we need to distinguish the generic-param from the =
feature-param?</FONT>
</P>

<P><FONT SIZE=3D2>Seems to me that the feature-param is IANA defined =
(we probably need</FONT>
<BR><FONT SIZE=3D2>to mention it in the IANA section), but the =
generic-param is not.</FONT>
</P>

<P><FONT SIZE=3D2>So, for example, you can't really know if Refer-To: =
bof@example.com;audio</FONT>
<BR><FONT SIZE=3D2>is implying the IANA-registered audio as per =
3840/this draft or</FONT>
<BR><FONT SIZE=3D2>anybody elses generic parameter which could mean =
something different.</FONT>
</P>

<P><FONT SIZE=3D2>I'm not sure if it is an issue of not. (I guess it =
depends if we expect that</FONT>
<BR><FONT SIZE=3D2>the recipient of the REFER is supposed to look at =
the parameter or not.)</FONT>
</P>

<P><FONT SIZE=3D2>If so, it might be better to have an explicit =
indication like ;feature-param=3Daudio or</FONT>
<BR><FONT SIZE=3D2>something like that.</FONT>
</P>

<P><FONT SIZE=3D2>If not, then I don't understand why we need to alter =
the ABNF, because feature-param is</FONT>
<BR><FONT SIZE=3D2>a generic-param in the first place. </FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: sip-bounces@ietf.org [<A =
HREF=3D"mailto:sip-bounces@ietf.org">mailto:sip-bounces@ietf.org</A>] =
On </FONT>
<BR><FONT SIZE=3D2>&gt; Behalf Of Cullen Jennings</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Friday, November 05, 2004 19:10</FONT>
<BR><FONT SIZE=3D2>&gt; To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: Orit Levin</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: [Sip] refer-with-feature-param</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I'm vague on all this stuff but ... Do you need =
anything </FONT>
<BR><FONT SIZE=3D2>&gt; normative for this at all - seems like this is =
already </FONT>
<BR><FONT SIZE=3D2>&gt; supported and perhaps we just need some example =
call flows </FONT>
<BR><FONT SIZE=3D2>&gt; demonstrating it.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Cullen</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 </FONT>
<BR><FONT SIZE=3D2>&gt; sip Use sipping@ietf.org for new developments =
on the </FONT>
<BR><FONT SIZE=3D2>&gt; application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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Use sipping@ietf.org for new developments on the application of sip
--===============1871162487==--



From sip-bounces@ietf.org  Sat Nov  6 19:18:36 2004
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To: Janet P Gunn <jgunn6@csc.com>
Subject: Re: [Sip] Strict,	Semi-Strict and Loose mode in RPH - not a good
	fit for ets and wps
References: <OF8725CF41.D45329BB-ON85256F44.0016317F-85256F44.001DFBD9@csc.com>
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Cc: fonashp@ncs.gov, Darren E Pado <dpado@csc.com>, mosleyv@ncs.gov,
        Saud Negash <snegash@csc.com>, a.ephrath@ieee.org, sip@ietf.org,
        nyquetek@msn.com, Richard F Kaczmarek <rkaczmarek@csc.com>,
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Janet P Gunn wrote:
> 
> 
> 
> One of the changes in the draft-ietf-sip-resource-priority-05 is the
> definition of 3 modes, strict, semi-strict, and loose, and the assignment
> of each defined namespaces to a particular mode.
> 
> The definition of the mode involves several distinct behaviors
> 

Would it be better to have each namespace definition fully define the 
behavior of that namesapce, rather than trying to have some "templates" 
that could be re-used by multiple namespaces?


--
Dean


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From sip-bounces@ietf.org  Sat Nov  6 19:33:01 2004
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Subject: [Sip] WGLC for Application Interaction
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The WGLC that Gonzalo announced for this draft in SIPPING will run 
concurrently in SIP.

Thanks, y'all.

--
Dean Willis
SIP cochair

-------- Original Message --------
Subject: [Sipping] WGLC: Application Interaction
Date: Wed, 03 Nov 2004 09:50:14 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
To: sipping <sipping@ietf.org>
CC: Rohan Mahy <rohan@ekabal.com>, Dean Willis <dean.willis@softarmor.com>

Folks,

we would like to working group last call the following draft:

http://www.ietf.org/internet-drafts/draft-ietf-sipping-app-interaction-framework-03.txt

This WGLC will finish on December 3rd. Please, send you comments to the
authors and to the list.

As a suggestion, you may want to review this document and KPML, which is
also under WGLC, together.

Thanks,

Gonzalo



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This list is for NEW development of the core SIP Protocol
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Date: Sat, 06 Nov 2004 14:37:18 -0800
Subject: Re: [Sip] Re: GRUU and things that rewrite contacts
From: Cullen Jennings <fluffy@cisco.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Jon Peterson <jon.peterson@neustar.biz>
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I was playing with a SBC from one of my favorite SBC vendors and don't see
why GRUU will cause lots problems. They do muck with contact but they do it
in a stateless and can pass instance and grid parameters right thought. They
know how to map changes done in outgoing contact and apply them to an
incoming request URI.

Yes, SBC will need to be GRU aware, or already be transparent to this type
of thing, but I guess I'm not seeing the issue.


On 9/21/04 12:13 AM, "Jonathan Rosenberg" <jdrosen@dynamicsoft.com> wrote:

> Jon,
> 
> I agree with you and Paul that we cannot, and should not, try here or
> anywhere else to define an SBC and consider its impacts. I think we can
> and should discuss the impact of contact-rewriting elements, and point
> out, as you say, that the identity draft would detect such a case.
> 
>  From a practical matter, I will say that the fact that SBCs will likely
> not work with GRUU is a real concern. It is going to make it hard to
> deploy gruu, and generally I am a really big fan of easy-to-deploy
> solutions. For better or worse, there ARE SBCs deployed in many existing
> SIP networks, and in those networks, getting gruu deployed will be
> harder. It will require coordination now between not just the client
> vendor and SIP server vendor, but now also the SBC vendor. In my
> experience, the more parts of the machine that all need to be
> simultaneously upgraded for a new feature, the less likely,
> exponentially I think, it is that such an upgrade will occur. If you
> couple this with the fact that, while gruu is definitely a key part of
> our specs, its an infrastructure improvement with no new features per
> se, it becomes a hard sell.
> 
> I still believe that the only alternative on the table - using a new
> header field for conveying gruu in register and invite/200 - has an even
> worse deployment path.
> 
> So, absent a better suggestion, and I don't have one, I think our only
> choice is to document the problems per above.
> 
> -Jonathan R.
> 
> 
> 
> Peterson, Jon wrote:
> 
>> I agree with Paul that SBC behavior is probably defined too poorly to refer
>> to it directly in the GRUU draft, and personally I'm very skeptical of
>> taking on that definition as a deliverable for GRUU. At the risk of
>> complicating a discussion about non-standard entities with standards:
>> 
>> RFC3261, Table 2:
>> 
>>       Header field          where   proxy ACK BYE CAN INV OPT REG
>>       ___________________________________________________________
>>       Contact                 R            o   -   -   m   o   o
>> 
>> 
>> I don't see an "m" under "proxy" there... but of course we're not talking
>> about proxy servers as such. That much said, if Contacts were something that
>> an entity other than a user agent could safely set, there would be an "m" in
>> the table up there. If an intermediary wants to request that it be in the
>> path of further SIP transactions in a dialog in a safe way, well, there's
>> another header for that. Furthermore, the GRUU acquisition mechanism
>> inherently provides a way for intermediaries to give a Contact header to a
>> user agent in order to strongly bind request routing to an intermediary. So,
>> there are reasonable alternatives to rewriting the header in transit.
>> 
>> Furthmore, I would like to point out that sip-identity provides integrity
>> over the addr-spec of the Contact header field. If the Contact header field
>> has been modified by an intermediary after integrity has been applied, it
>> should be considered an integrity violation by the recipient.
>> 
>> This is of course not to say that there are no architectures where SBCs and
>> sip-identity might live together in harmony. The point is that SIP can't be
>> expected to differentiate between friendly and unfriendly intermediaries
>> that break the rules, and that we certainly can't enable arbitary entities
>> to modify the Contact header or we open ourselves up to all sorts of nasty
>> attacks. Whatever security gains people think they are getting from
>> rewriting the Contact header at an intermediary, I'm sure they pale in
>> comparison to the threats that the arise if Contact header can be arbitarily
>> changed by anyone. If we want to prevent something from rewriting a header
>> that shouldn't be rewritten, we should use some form of integrity to prevent
>> that.
>> 
>> So, I do think it is reasonable for the GRUU doc to discuss the impact of an
>> intermediary rewriting the Contact header in general (without defining
>> SBCs), and point out that integrity mechanisms like sip-identity could be
>> used to detect a rewrite of a Contact header. But, on some level, I guess I
>> feel rewriting Contacts at an intermediary is already acknowledged to be
>> problematic by the core SIP standard. It'd also be a problem for GRUU if an
>> SBC removed the Call-ID header field, or did any of a number of things it
>> shouldn't do.
>> 
>> Jon Peterson
>> NeuStar, Inc.
>> 
>> 
>>> -----Original Message-----
>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>> Sent: Thursday, September 16, 2004 4:32 PM
>>> To: Jonathan Rosenberg
>>> Cc: sip@ietf.org; Robert Sparks
>>> Subject: Re: [Sip] Re: GRUU and things that rewrite contacts
>>> 
>> 
>> [snip]
>> 
>>> I don't want to comment on the above right now because we are clearly
>>> have different assumptions here.
>>> 
>>> I don't think we can meaningfully comment on the impact on SBCs when we
>>> have no definition of such a beast. And trying to generalize to all
>>> forms of B2BUA makes things worse - we repeatedly state that about all
>>> you can say in general about a B2BUA is that it is two connected UAs -
>>> you can't say anything about how the two calls are related.
>>> 
>>> If this is an important problem for us to mention at all, (and I'm not
>>> yet convinced it is), then I think the first step is to define an SBC,
>>> along with the kinds of transformations it performs. Since that has
>>> never been written down, when it is written it can deal
>>> properly with GRUUs.
>>> 
>>> Paul
>>> 
>> 
>> 



_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Sat Nov  6 23:19:46 2004
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From: "Medhavi Bhatia" <mbhatia@nextone.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Subject: RE: [Sip] Re: GRUU and things that rewrite contacts
Date: Sat, 6 Nov 2004 23:15:22 -0500
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I may have caught up to this discussion too late, but here are some
thoughts. I probably need to dig deeper into the GRUU draft too.

There seem to be two cases.

1) SBC sits between the UA and the proxy/registrar, where the UA sends
requests for the proxy addressed to the SBC. In this case, I don't see much
of a concern as contact parameters are not overwritten for REGISTER. For the
INVITE, since the SBC knows that the Contact is a GRUU, it is not
overwritten. The last part needs SBCs to understand the GRUUs. 

2) The same as above, except the UA does not know who the Registrar is. This
is a tricky application and I guess not part of the usual suspects. In this
case, the SBC will need to be aware of GRUUs on the REGISTER also.

The SBC also may have an option to create a GRUU and map one obtained from
the Registrar to this one. 

-Medhavi.

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Paul
Kyzivat
> Sent: Tuesday, September 21, 2004 9:13 AM
> To: Jonathan Rosenberg
> Cc: sip@ietf.org; Robert Sparks; Peterson, Jon
> Subject: Re: [Sip] Re: GRUU and things that rewrite contacts
> 
> This would be a good time for those that build SBCs to take note and
> action, and to speak up if they have difficulty with the deployment of
> GRUUs.
> 
> 	Paul
> 
> Jonathan Rosenberg wrote:
> > Jon,
> >
> > I agree with you and Paul that we cannot, and should not, try here or
> > anywhere else to define an SBC and consider its impacts. I think we can
> > and should discuss the impact of contact-rewriting elements, and point
> > out, as you say, that the identity draft would detect such a case.
> >
> >  From a practical matter, I will say that the fact that SBCs will likely
> > not work with GRUU is a real concern. It is going to make it hard to
> > deploy gruu, and generally I am a really big fan of easy-to-deploy
> > solutions. For better or worse, there ARE SBCs deployed in many existing
> > SIP networks, and in those networks, getting gruu deployed will be
> > harder. It will require coordination now between not just the client
> > vendor and SIP server vendor, but now also the SBC vendor. In my
> > experience, the more parts of the machine that all need to be
> > simultaneously upgraded for a new feature, the less likely,
> > exponentially I think, it is that such an upgrade will occur. If you
> > couple this with the fact that, while gruu is definitely a key part of
> > our specs, its an infrastructure improvement with no new features per
> > se, it becomes a hard sell.
> >
> > I still believe that the only alternative on the table - using a new
> > header field for conveying gruu in register and invite/200 - has an even
> > worse deployment path.
> >
> > So, absent a better suggestion, and I don't have one, I think our only
> > choice is to document the problems per above.
> >
> > -Jonathan R.
> >
> >
> >
> > Peterson, Jon wrote:
> >
> >> I agree with Paul that SBC behavior is probably defined too poorly to
> >> refer
> >> to it directly in the GRUU draft, and personally I'm very skeptical of
> >> taking on that definition as a deliverable for GRUU. At the risk of
> >> complicating a discussion about non-standard entities with standards:
> >>
> >> RFC3261, Table 2:
> >>
> >>       Header field          where   proxy ACK BYE CAN INV OPT REG
> >>       ___________________________________________________________
> >>       Contact                 R            o   -   -   m   o   o
> >>
> >>
> >> I don't see an "m" under "proxy" there... but of course we're not
talking
> >> about proxy servers as such. That much said, if Contacts were
> >> something that
> >> an entity other than a user agent could safely set, there would be an
> >> "m" in
> >> the table up there. If an intermediary wants to request that it be in
the
> >> path of further SIP transactions in a dialog in a safe way, well,
there's
> >> another header for that. Furthermore, the GRUU acquisition mechanism
> >> inherently provides a way for intermediaries to give a Contact header
> >> to a
> >> user agent in order to strongly bind request routing to an
> >> intermediary. So,
> >> there are reasonable alternatives to rewriting the header in transit.
> >>
> >> Furthmore, I would like to point out that sip-identity provides
integrity
> >> over the addr-spec of the Contact header field. If the Contact header
> >> field
> >> has been modified by an intermediary after integrity has been applied,
it
> >> should be considered an integrity violation by the recipient.
> >>
> >> This is of course not to say that there are no architectures where
> >> SBCs and
> >> sip-identity might live together in harmony. The point is that SIP
> >> can't be
> >> expected to differentiate between friendly and unfriendly
intermediaries
> >> that break the rules, and that we certainly can't enable arbitary
> >> entities
> >> to modify the Contact header or we open ourselves up to all sorts of
> >> nasty
> >> attacks. Whatever security gains people think they are getting from
> >> rewriting the Contact header at an intermediary, I'm sure they pale in
> >> comparison to the threats that the arise if Contact header can be
> >> arbitarily
> >> changed by anyone. If we want to prevent something from rewriting a
> >> header
> >> that shouldn't be rewritten, we should use some form of integrity to
> >> prevent
> >> that.
> >>
> >> So, I do think it is reasonable for the GRUU doc to discuss the impact
> >> of an
> >> intermediary rewriting the Contact header in general (without defining
> >> SBCs), and point out that integrity mechanisms like sip-identity could
be
> >> used to detect a rewrite of a Contact header. But, on some level, I
> >> guess I
> >> feel rewriting Contacts at an intermediary is already acknowledged to
be
> >> problematic by the core SIP standard. It'd also be a problem for GRUU
> >> if an
> >> SBC removed the Call-ID header field, or did any of a number of things
it
> >> shouldn't do.
> >>
> >> Jon Peterson
> >> NeuStar, Inc.
> >>
> >>
> >>> -----Original Message-----
> >>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >>> Sent: Thursday, September 16, 2004 4:32 PM
> >>> To: Jonathan Rosenberg
> >>> Cc: sip@ietf.org; Robert Sparks
> >>> Subject: Re: [Sip] Re: GRUU and things that rewrite contacts
> >>>
> >>
> >> [snip]
> >>
> >>> I don't want to comment on the above right now because we are clearly
> >>> have different assumptions here.
> >>>
> >>> I don't think we can meaningfully comment on the impact on SBCs when
> >>> we have no definition of such a beast. And trying to generalize to
> >>> all forms of B2BUA makes things worse - we repeatedly state that
> >>> about all you can say in general about a B2BUA is that it is two
> >>> connected UAs - you can't say anything about how the two calls are
> >>> related.
> >>>
> >>> If this is an important problem for us to mention at all, (and I'm
> >>> not yet convinced it is), then I think the first step is to define an
> >>> SBC, along with the kinds of transformations it performs. Since that
> >>> has never been written down, when it is written it can deal properly
> >>> with GRUUs.
> >>>
> >>>     Paul
> >>>
> >>
> >>
> >
> 
> 
> _______________________________________________
> 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 sipping@ietf.org for new developments on the application of sip


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Yes, it would be better to have the namespace define the behaviors.

However, I accept that there may be some general bounds on the behaviors.

It is linking them together in fixed combinations that I object to.

Janet


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                      Dean Willis                                                                                              
                      <dean.willis             To:      Janet P Gunn/FED/CSC@CSC                                               
                      @softarmor.com>          cc:      sip@ietf.org, jmpolk@cisco.com, fonashp@ncs.gov, Darren E              
                                               Pado/FED/CSC@CSC, Saud Negash/FED/CSC@CSC, mosleyv@ncs.gov, Richard F           
                      11/06/2004 08:12         Kaczmarek/FED/SC/CSC@CSC, nyquetek@msn.com, a.ephrath@ieee.org,                 
                      PM                       suracif@ncs.gov, KENNETH.R.ERNEY@saic.com, Dennis Q Berg/FED/CSC@CSC            
                                               Subject: Re: [Sip] Strict, Semi-Strict and Loose mode in RPH - not a good fit   
                                               for ets and wps                                                                 
                                                                                                                               




Janet P Gunn wrote:
>
>
>
> One of the changes in the draft-ietf-sip-resource-priority-05 is the
> definition of 3 modes, strict, semi-strict, and loose, and the assignment
> of each defined namespaces to a particular mode.
>
> The definition of the mode involves several distinct behaviors
>

Would it be better to have each namespace definition fully define the
behavior of that namesapce, rather than trying to have some "templates"
that could be re-used by multiple namespaces?


--
Dean






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Subject: RE: [Sip] Strict,
	Semi-Strict and Loose mode in RPH - not a good fit for ets and wps
To: "Ken Carlberg" <carlberg@g11.org.uk>
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comments in line


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                      "Ken Carlberg"                                                                                           
                      <carlberg                To:      Janet P Gunn/FED/CSC@CSC, <sip@ietf.org>, <jmpolk@cisco.com>           
                      @g11.org.uk>             cc:      <fonashp@ncs.gov>, Darren E Pado/FED/CSC@CSC, Saud Negash/FED/CSC@CSC, 
                                               <mosleyv@ncs.gov>, Richard F Kaczmarek/FED/SC/CSC@CSC, <nyquetek@msn.com>,      
                      11/06/2004 09:26         <a.ephrath@ieee.org>, <suracif@ncs.gov>, <KENNETH.R.ERNEY@saic.com>, Dennis Q   
                      AM                       Berg/FED/CSC@CSC                                                                
                                               Subject: RE: [Sip] Strict,Semi-Strict and Loose mode in RPH - not a good fit    
                                               for ets and wps                                                                 
                                                                                                                               




> Unfortunately, this  (semi-strict) is NOT the behavior desired for the
ets
> and wps namespaces.
>
> The behavior desired for the ets and wps namespaces are
> 1 RPH is optional (semi-strict and loose)
> 2 SIP element ignores a namespace it does not recognize (loose)

I would prefer a strict or semi-strict mode because these provide feedback
in
case: a) a misconfigured UA sends the wrong namespace.priority, and/or b)
initial selection of a server/proxy that does not support the desired
namespace.  The former is quite possible given an ability (and possibly,
necessity) of the user to configure their own device to support the given
namespace.value combination.

[JG] Before posting this, I checked with Dr. Fonash (head of NCS, which
"owns" GETS and WPS).  He was very clear about wanting "If the SIP element
does not recognize the namespace (wps or ets) it should pass it on without
acting on it, rather than rejecting it."

> 3 The content of the error message should not reveal (to the end user)
the
> valid values for that namespace (strict)

section 4.5 of the drafts states:

   If the UAS understands the resource value, but refuses to honor the
   request with elevated priority for this particular user, it returns
   the 403 (Forbidden) response code.  It MAY include the list of
   resource values that the user is allowed to use in the
   'Accept-Resource-Priority' response header field.

The "MAY" is the critical term.  My understanding would be that this
decision
is a local policy issue.  Outside of that I personally don't see a problem
with
providing "hints" of accepted values under the context that they can only
acted
upon by authorized users.  Informing the user of the acceptable set of
priorities does not bypass or circumvent the need for proper authentication
credentials.

[JG]I am not as concerned about this case (valid namespace.valid value -
just not authorized).  I am more concerned about the  error message when
the namespace, or value are not recognized.  But if ets and wps operate
with the behavior "SIP element ignores a namespace it does not recognize",
half of that goes away.

[JG]I am still not entirely sure about the desired behavior for the other
half- (valid namespace.invalid value).  But I suspect that it would be
better to
a) Reject the message
b) NOT return the list of valid values.

[JG] I do not think that the full loose mode as defined (if the SIP element
gets (valid namespace.invalid value) it processes the message as if it had
no RPH) is the right approach.  I think these messages should be rejected,
and get an error message.

[JG]But I am open to other approaches.


> 4 SIP messages that include the RPH with the ets or wps namespaces should
> include the authorization header (or equivalent mechanism) (semi-strict)
> 5 Precedence (queueing) is generally the means of providing priority, but
> exemption from network management controls, for instance call admission
> within an IP domain, when a "regular" call would not be admitted, is also
> needed.  Pre-emption is not used.  (semi-strict "plus more")

the "plus more" would seem to be a product of local policy

[JG] Agree.  While it doesn't explicitly state it, the document leads you
to believe that queueing is the only behavior permitted in semi-strict
mode.

> 6 There is no a priori requirement to use, or not use, the "Require"
> header field.

isn't this addressed in section 4.3.2 below?  or am I missing something?

   If the request includes a 'Require' header field with the
   'Resource-Priority' option tag, a UAS MUST follow the strict or
   semi-strict mode rules, otherwise UAS and proxies MUST operate in
   loose mode.


[JG]I guess I wasn't clear.   I simply meant that there wasn't a concern
about the "require" header itself, independent of the
strict/semi-strict/loose mode.  Once that is "right", the approach to the
"require" header will follow

-ken






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From sip-bounces@ietf.org  Sun Nov  7 09:20:38 2004
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From: "Ken Carlberg" <carlberg@g11.org.uk>
To: "'Janet P Gunn'" <jgunn6@csc.com>
Subject: RE: [Sip] Strict,
	Semi-Strict and Loose mode in RPH - not a good fit for ets and wps
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> I would prefer a strict or semi-strict mode because these provide feedback in
> case: a) a misconfigured UA sends the wrong namespace.priority, and/or b)
> initial selection of a server/proxy that does not support the desired
> namespace.  The former is quite possible given an ability (and possibly,
> necessity) of the user to configure their own device to support the given
> namespace.value combination.
> 
> [JG] Before posting this, I checked with Dr. Fonash (head of NCS, which
> "owns" GETS and WPS).  He was very clear about wanting "If the SIP element
> does not recognize the namespace (wps or ets) it should pass it on without
> acting on it, rather than rejecting it."

ok.  Dr Fonash now has additional input on the subject given my previous email;
if there is no feedback, then the sender hopes everything is working fine.
further, it will continue to use an R-P capable proxy with no feedback from the
INVITE indicating that it does not support that particular namespace/value.  

Keep in mind that as it states earlier in the draft, the R-P header is ignored
(assuming proper implementation) by those proxies (eg, the installed base) that
do not support the new R-P header.  As a sidenote, our current implementation
of the R-P header follows the MAY portion of the draft and does not send back
hints of what are acceptable priorities.

Also, when you say the NCS "owns" GETS and WPS, are you refering to the systems
or the namespace?  If its both, then I have my doubts about the latter.  I see
no association of the NCS with WPS or GETS in the draft and it would seem that
initial "ownership" of those namespaces, as well as the others like ETS, lies
with the authors of the draft.  (I'll be happy to be corrected on that
assumption).

-ken


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From sip-bounces@ietf.org  Sun Nov  7 09:48:39 2004
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my apologies for the non-technical comment, and perhaps this thread should go
to the IETF list if there are responses.

private comments to authors is as we all know a time honored tradition in the
IETF.  its very beneficial in lowering the signal-to-noise ratio for minor
comments, word-smithing, and conveying the opinions of those people that are
either shy or feel that privacy will help clarify things.

however, private comments as a *primary and consistent* form of input to
authors leading to *significant* changes is detrimental and contrary to the
open process of the IETF.  It contributes to confusion for those on the list
and can be regressive and unnecesarily delays progression of a draft.

I ask that people keep this in mind and not send private comments beyond
clarification or word-smithing.

cheers,

-ken
  


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Subject: Re: [Sip] Strict,
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In a message dated 11/6/2004 7:21:20 PM Eastern Standard Time, 
dean.willis@softarmor.com writes:


> Would it be better to have each namespace definition fully define the 
> behavior of that namesapce, rather than trying to have some "templates" 
> that could be re-used by multiple namespaces?
> 
Absolutely, and as described in earlier versions of this draft, that 
documentation does not have to be in an RFC. That is also why I object to this new 
version -05. It requires an RFC for every new namespace. Future uses can not, nor 
should they have to, go through this excruciating process of producing an RFC 
just to register a new namespace.


------------------------------------------------------------------------------
---------------------
In a message dated 11/7/2004 9:24:23 AM Eastern Standard Time, 
carlberg@g11.org.uk writes:


> Also, when you say the NCS "owns" GETS and WPS, are you refering to the 
> systems
> or the namespace?  If its both, then I have my doubts about the latter.  I 
> see
> no association of the NCS with WPS or GETS in the draft and it would seem 
> that
> initial "ownership" of those namespaces, as well as the others like ETS, lies
> with the authors of the draft.  (I'll be happy to be corrected on that
> assumption).
> 
In the case of the DSN namespace, the previous version of this draft had the 
following for the definition of this namespace:

9.5.2  Namespace dsn

   Namespace: dsn
   Description: United States Defense Switched Network.  The values are
      adopted from RFC 791 [RFC0791], omitting the levels "critic-ecp",
      "network control" and "internetwork control", as these are
      inappropriate here.
   Documentation: ANSI T1.619, Section B1
   Organization: United States Department of Defense, Defense
      Information Systems Agency (DISA).

So, in a way, DISA does "own" this namespace. At least, DISA is responsible 
for defining its operation within its own network.
This material was removed from the latest version, which also is why I object 
to accepting version -05.

And, no, the "ownership" of these namespaces does not lie with the "authors" 
at this point. It is a Working Group draft, meaning that the "ownership" 
currently lies with the WG as a whole, and changes to this draft should not be made 
unless agreed to by the WG. There was never any discussion about many of 
these changes in -05.

Mike Pierce



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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/6/2004 7:21:20 PM Eastern Standard Time, dean.willis@softarmor.com=20=
writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Would it be better to have=20=
each namespace definition fully define the=20
<BR>behavior of that namesapce, rather than trying to have some "templates"=20
<BR>that could be re-used by multiple namespaces?
<BR></BLOCKQUOTE>
<BR>Absolutely, and as described in earlier versions of this draft, that doc=
umentation does not have to be in an RFC. That is also why I object to this=20=
new version -05. It requires an RFC for every new namespace. Future uses can=
 not, nor should they have to, go through this excruciating process of produ=
cing an RFC just to register a new namespace.
<BR>
<BR>
<BR>------------------------------------------------------------------------=
---------------------------
<BR>In a message dated 11/7/2004 9:24:23 AM Eastern Standard Time, carlberg@=
g11.org.uk writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Also, when you say the NCS=20=
"owns" GETS and WPS, are you refering to the systems
<BR>or the namespace? &nbsp;If its both, then I have my doubts about the lat=
ter. &nbsp;I see
<BR>no association of the NCS with WPS or GETS in the draft and it would see=
m that
<BR>initial "ownership" of those namespaces, as well as the others like ETS,=
 lies
<BR>with the authors of the draft. &nbsp;(I'll be happy to be corrected on t=
hat
<BR>assumption).
<BR></BLOCKQUOTE>
<BR>In the case of the DSN namespace, the previous version of this draft had=
 the following for the definition of this namespace:
<BR>
<BR>9.5.2 &nbsp;Namespace dsn
<BR>
<BR> &nbsp;&nbsp;Namespace: dsn
<BR> &nbsp;&nbsp;Description: United States Defense Switched Network. &nbsp;=
The values are
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;adopted from RFC 791 [RFC0791], omitting=20=
the levels "critic-ecp",
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;"network control" and "internetwork contr=
ol", as these are
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;inappropriate here.
<BR> &nbsp;&nbsp;Documentation: ANSI T1.619, Section B1
<BR> &nbsp;&nbsp;Organization: United States Department of Defense, Defense
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Information Systems Agency (DISA).
<BR>
<BR>So, in a way, DISA does "own" this namespace. At least, DISA is responsi=
ble for defining its operation within its own network.
<BR>This material was removed from the latest version, which also is why I o=
bject to accepting version -05.
<BR>
<BR>And, no, the "ownership" of these namespaces does not lie with the "auth=
ors" at this point. It is a Working Group draft, meaning that the "ownership=
" currently lies with the WG as a whole, and changes to this draft should no=
t be made unless agreed to by the WG. There was never any discussion about m=
any of these changes in -05.
<BR>
<BR>Mike Pierce
<BR>
<BR></FONT></HTML>

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Subject: Re: [Sip] Re: GRUU and things that rewrite contacts
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	Jon'" <jon.peterson@neustar.biz>
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Medhavi Bhatia wrote:
> I may have caught up to this discussion too late, but here are some
> thoughts. I probably need to dig deeper into the GRUU draft too.
> 
> There seem to be two cases.
> 
> 1) SBC sits between the UA and the proxy/registrar, where the UA sends
> requests for the proxy addressed to the SBC.

Can you clarify what you mean above - perhaps give an example?
In general, the UA doesn't know much about what is going on. At best, it 
knows how to construct the To-uri and derive the request-uri from it, 
and then may have a loose route to insert in the message. Perhaps the 
loose route transits an SBC, and maybe it acts as a B2BUA. Or do you 
have something else in mind?

 > In this case, I don't see much
> of a concern as contact parameters are not overwritten for REGISTER.

You state this as fact. But this goes to the heart of the problem that 
we have no definition of what an SBC does. I guess yours must not 
overwrite contact parameters in a REGISTER.

 > For the
> INVITE, since the SBC knows that the Contact is a GRUU,

It does? How? I guess it can know in some cases, if it saw the register 
that requested the GRUU it might remember the returned gruu and 
recognize its use in future requests, but that would be a lot of work. 
And it won't work for gruu values that it didnt' see being assigned.

In general you just can't recognize a gruu by inspection.

 > it is not
> overwritten. The last part needs SBCs to understand the GRUUs. 

Please explain what kind of understanding you think you require, and how 
it is even theoretically possible to have it.

> 2) The same as above, except the UA does not know who the Registrar is. This
> is a tricky application and I guess not part of the usual suspects. In this
> case, the SBC will need to be aware of GRUUs on the REGISTER also.

I have no idea what you mean here.

> The SBC also may have an option to create a GRUU and map one obtained from
> the Registrar to this one. 

I think that was more or less what Jonathan was suggesting it might do 
in his original posting. It is pretty ugly.

I have the distressing feeling that SBCs are just doing random stuff, 
which has lots of potential unintended consequences.

	Paul

> -Medhavi.
> 
> 
>>-----Original Message-----
>>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Paul
> 
> Kyzivat
> 
>>Sent: Tuesday, September 21, 2004 9:13 AM
>>To: Jonathan Rosenberg
>>Cc: sip@ietf.org; Robert Sparks; Peterson, Jon
>>Subject: Re: [Sip] Re: GRUU and things that rewrite contacts
>>
>>This would be a good time for those that build SBCs to take note and
>>action, and to speak up if they have difficulty with the deployment of
>>GRUUs.
>>
>>	Paul
>>
>>Jonathan Rosenberg wrote:
>>
>>>Jon,
>>>
>>>I agree with you and Paul that we cannot, and should not, try here or
>>>anywhere else to define an SBC and consider its impacts. I think we can
>>>and should discuss the impact of contact-rewriting elements, and point
>>>out, as you say, that the identity draft would detect such a case.
>>>
>>> From a practical matter, I will say that the fact that SBCs will likely
>>>not work with GRUU is a real concern. It is going to make it hard to
>>>deploy gruu, and generally I am a really big fan of easy-to-deploy
>>>solutions. For better or worse, there ARE SBCs deployed in many existing
>>>SIP networks, and in those networks, getting gruu deployed will be
>>>harder. It will require coordination now between not just the client
>>>vendor and SIP server vendor, but now also the SBC vendor. In my
>>>experience, the more parts of the machine that all need to be
>>>simultaneously upgraded for a new feature, the less likely,
>>>exponentially I think, it is that such an upgrade will occur. If you
>>>couple this with the fact that, while gruu is definitely a key part of
>>>our specs, its an infrastructure improvement with no new features per
>>>se, it becomes a hard sell.
>>>
>>>I still believe that the only alternative on the table - using a new
>>>header field for conveying gruu in register and invite/200 - has an even
>>>worse deployment path.
>>>
>>>So, absent a better suggestion, and I don't have one, I think our only
>>>choice is to document the problems per above.
>>>
>>>-Jonathan R.
>>>
>>>
>>>
>>>Peterson, Jon wrote:
>>>
>>>
>>>>I agree with Paul that SBC behavior is probably defined too poorly to
>>>>refer
>>>>to it directly in the GRUU draft, and personally I'm very skeptical of
>>>>taking on that definition as a deliverable for GRUU. At the risk of
>>>>complicating a discussion about non-standard entities with standards:
>>>>
>>>>RFC3261, Table 2:
>>>>
>>>>      Header field          where   proxy ACK BYE CAN INV OPT REG
>>>>      ___________________________________________________________
>>>>      Contact                 R            o   -   -   m   o   o
>>>>
>>>>
>>>>I don't see an "m" under "proxy" there... but of course we're not
>>>
> talking
> 
>>>>about proxy servers as such. That much said, if Contacts were
>>>>something that
>>>>an entity other than a user agent could safely set, there would be an
>>>>"m" in
>>>>the table up there. If an intermediary wants to request that it be in
>>>
> the
> 
>>>>path of further SIP transactions in a dialog in a safe way, well,
>>>
> there's
> 
>>>>another header for that. Furthermore, the GRUU acquisition mechanism
>>>>inherently provides a way for intermediaries to give a Contact header
>>>>to a
>>>>user agent in order to strongly bind request routing to an
>>>>intermediary. So,
>>>>there are reasonable alternatives to rewriting the header in transit.
>>>>
>>>>Furthmore, I would like to point out that sip-identity provides
>>>
> integrity
> 
>>>>over the addr-spec of the Contact header field. If the Contact header
>>>>field
>>>>has been modified by an intermediary after integrity has been applied,
>>>
> it
> 
>>>>should be considered an integrity violation by the recipient.
>>>>
>>>>This is of course not to say that there are no architectures where
>>>>SBCs and
>>>>sip-identity might live together in harmony. The point is that SIP
>>>>can't be
>>>>expected to differentiate between friendly and unfriendly
>>>
> intermediaries
> 
>>>>that break the rules, and that we certainly can't enable arbitary
>>>>entities
>>>>to modify the Contact header or we open ourselves up to all sorts of
>>>>nasty
>>>>attacks. Whatever security gains people think they are getting from
>>>>rewriting the Contact header at an intermediary, I'm sure they pale in
>>>>comparison to the threats that the arise if Contact header can be
>>>>arbitarily
>>>>changed by anyone. If we want to prevent something from rewriting a
>>>>header
>>>>that shouldn't be rewritten, we should use some form of integrity to
>>>>prevent
>>>>that.
>>>>
>>>>So, I do think it is reasonable for the GRUU doc to discuss the impact
>>>>of an
>>>>intermediary rewriting the Contact header in general (without defining
>>>>SBCs), and point out that integrity mechanisms like sip-identity could
>>>
> be
> 
>>>>used to detect a rewrite of a Contact header. But, on some level, I
>>>>guess I
>>>>feel rewriting Contacts at an intermediary is already acknowledged to
>>>
> be
> 
>>>>problematic by the core SIP standard. It'd also be a problem for GRUU
>>>>if an
>>>>SBC removed the Call-ID header field, or did any of a number of things
>>>
> it
> 
>>>>shouldn't do.
>>>>
>>>>Jon Peterson
>>>>NeuStar, Inc.
>>>>
>>>>
>>>>
>>>>>-----Original Message-----
>>>>>From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>>>Sent: Thursday, September 16, 2004 4:32 PM
>>>>>To: Jonathan Rosenberg
>>>>>Cc: sip@ietf.org; Robert Sparks
>>>>>Subject: Re: [Sip] Re: GRUU and things that rewrite contacts
>>>>>
>>>>
>>>>[snip]
>>>>
>>>>
>>>>>I don't want to comment on the above right now because we are clearly
>>>>>have different assumptions here.
>>>>>
>>>>>I don't think we can meaningfully comment on the impact on SBCs when
>>>>>we have no definition of such a beast. And trying to generalize to
>>>>>all forms of B2BUA makes things worse - we repeatedly state that
>>>>>about all you can say in general about a B2BUA is that it is two
>>>>>connected UAs - you can't say anything about how the two calls are
>>>>>related.
>>>>>
>>>>>If this is an important problem for us to mention at all, (and I'm
>>>>>not yet convinced it is), then I think the first step is to define an
>>>>>SBC, along with the kinds of transformations it performs. Since that
>>>>>has never been written down, when it is written it can deal properly
>>>>>with GRUUs.
>>>>>
>>>>>    Paul
>>>>>
>>>>
>>>>
>>
>>_______________________________________________
>>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: [Sip] draft-ietf-sip-content-indirect-mech 
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URI scheme negotiation:

Section 5.2
I am not sure I understand the URI scheme mechanizm. As I read
the draft it seems the party providing the content gets to
specify the alternative URI schemes.  How does the retriever
of the indirect content get to indicate what it supports.

It is not always a UAS that is providing content indirection.
In the configuration framework an event package is proposed
where the NOTIFY request would use content indirection.

So how does a subscriber indicate that it supports the
FOO URI scheme.

Can multipart/alternative be used by the notifier to provide
alternative URI scheme URIs for content indirection (e.g.
provide a FTP or XCAP alternative to the HTTP URI that is
also included)?

etag and content-id interaction:
This draft describes a requirement for relating the HTTP etag to
the content, but no binding is proposed.  When using content indirection
with XCAP it is very important that the content-id be somehow bound
to the etag.  Otherwise there is a race condition where the receiver
of the content indirection URI does not know if the last retrived content
(label by the content-id) corresponds to the etag advertised in a
NOTIFY of content change.  The simplest form of binding of the etag
would be to simply use the etag as the content-id.

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From: Adam Roach <adam@nostrum.com>
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Subject: [Sip] Re: I-D ACTION:draft-ietf-sip-identity-03.txt
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This looks really solid. In response to Dean's earlier query, I suspect 
this is quickly reaching "ready for WGLC" status.

When looking over this, though, there are two issues that might need a 
bit more text (although one is really more of a nit).

The first issue revolves around proxy validation of received requests 
which contain authenticated identity headers, and how these proxies 
communicate with the user who is the target of the request, and has two 
sub-issues.

Primary among these is a question of how the proxies communicate 
discrepancies to the recipient. Section 7 requires:

>    Additionally, the Date, Contact and Call-ID headers MUST be analyzed
>    in the manner described in Section 14; recipients that wish to verify
>    Identity signatures MUST support all of the operations described
>    there.  Any discrepancies or violations MUST be reported to the user.


If I implement a proxy, it is not clear how I can satisfy this second MUST.

A secondary sub-issue arises from the mechanism by which the proxy 
decides to trust certificates. There is nothing in the current draft 
that suggests, for example, that trusting self-signed domain certs for 
the purposes of user authentication might be a Really Bad Idea. (Obvious 
to security wonks, maybe not so much for your average implementor). The 
reason this ties into user interaction is that it becomes unclear how a 
proxy should react if it receives a request; that request has an 
authenticated identity asserted by a domain; and the root CA for the 
domain's certificate is not known. With user applications, this is 
typically handled by saying, "I don't know who blackhelicopers.org is, 
but they signed this certificate. Click here to examine the cert 
yourself, and let me know if I should accept this cert as invalid, 
temporarily valid, or permanently valid." When the check is performed by 
a proxy, such interaction is really quite difficult. The best a proxy 
can do is reject the request with a 428 (which seems a bit extreme), or 
let it through (in which case the proxy's check serves little to no purpose)

So, I suggest that the draft probably needs a little more text around 
proxy handling of requests; further, given the all-or-nothing nature of 
proxy validation of authenticated identity, we might consider 
discouraging its use in favor of client validation.

The second issue is a tiny nit in that the current draft doesn't specify 
what to do if you receive a request with an Identity-Info header which 
contains a URI with a scheme other than sips: or https:. Presumably, 
recipients should 428 such requests? 400?

/a


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From: Adam Roach <adam@nostrum.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>
Subject: Re: [Sip] GRUU and sips
References: <BD293668.BBEF%fluffy@cisco.com>	<5.2.1.1.0.20040806093622.01542d40@pop.mcilink.com>
	<4113A68C.60704@cisco.com>
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Cc: Alan Johnston <alan.johnston@mci.com>, Cullen Jennings <fluffy@cisco.com>,
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        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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I'm getting in a bit late on this thread (sorry), but I've been under 
the impression for quite some time that sip:foo and sips:foo are 
*defined* to be the same resource under all circumstances. Section 19.1 
of 3261 seems rather clear on this point:

>    Any resource described by a SIP URI can be
>    "upgraded" to a SIPS URI by just changing the scheme, if it is
>    desired to communicate with that resource securely.


So, if we just hand out sip: URIs, then full 3261 implementations *know* 
that they can use it with a sips: scheme and get the same resource. 
Perhaps it is worth reiterating this point in the GRUU document, but it 
seems like this is a non-problem.

/a


Paul Kyzivat wrote:

> Alan,
>
> Did you see my later message on this subject?
>
> As far as bid down, that is only a problem if the UAS isn't aware that 
> it happened. Note that no matter what we say, bad guys can make the 
> change, so it isn't the change that matters, but how it is handled 
> along the path.
>
> In a direct connection, the UAS can see whether the r-uri is sip/sips. 
> If the r-uri is sip it may conclude that the session isn't secure. For 
> indirect connections it becomes the obligation of the proxies along 
> the path to use a sip r-uri if the connection can't be guaranteed 
> secure to them.
>
> My concern, beyond understanding how GRUU should work, is that a UAS 
> that wants to support both secure and insecure communication should 
> only need to make one registration. I have feeling people widely think 
> things should work this way, but the standard isn't clear if/how this 
> can be achieved.
>
>     Paul
>
> Alan Johnston wrote:
>
>> At 10:35 AM 7/27/2004 -0400, Paul Kyzivat wrote:
>>
>>
>>> Cullen Jennings wrote:
>>>
>>>> Sounds like a possibility. Can you think of a way where I do not 
>>>> have to
>>>> Register twice to get both a sip and sips URL?
>>>> Perhaps we should just say that is a proxy hands out a sip GRUU 
>>>> sip:foo then
>>>> sips:foo MUST also be a valid GRUU with the contact moved to sips 
>>>> from sip.
>>>
>>>
>>>
>>> That would rule out proxies that support gruu but not TLS.
>>>
>>> I think the converse might make more sense - if the proxy hands out 
>>> sips:foo then sip:foo MUST also be a valid GRUU.
>>
>>
>>
>> Are you sure this is a good idea?  Doesn't this open a bid down 
>> attack for the GRUU + grid dialog identifier described by Robert in 
>> his dialog reuse draft?
>>
>> Here's the example - I have a secure SIP dialog with you.  You are 
>> using a secure SIP GRUU + grid dialog identifier.  I pass this GRUU 
>> to another party, such as Robert with a Replaces header field in a 
>> REFER for the intent to transfer you to Robert.  Robert changes the 
>> URI to a regular SIP GRUU and sends the INVITE with Replaces.  If you 
>> accept this, you have replaced a secure SIP session with an insecure 
>> session - not a good idea.  Better for this transfer to fail if 
>> Robert tries to downgrade the security on the connection.
>>
>> I would say a sips:foo GRUU MUST NOT be valid as sip:foo to avoid 
>> this kind of attack.
>>
>> Thanks,
>> Alan Johnston
>>
>>
>>>         Paul
>>>
>>>> If I get a sip GRUU, others can still use TLS to contact it. I 
>>>> would like to
>>>> make sure that the following case works:  a UA only does UDP to 
>>>> it's proxy,
>>>> however, it want communications from outside the domain to use TLS 
>>>> to the
>>>> proxy. I think this works fine with what you are proposing.
>>>> On 7/23/04 12:27 PM, "Jonathan Rosenberg" <jdrosen@dynamicsoft.com> 
>>>> wrote:
>>>>
>>>>> The current GRUU spec says very little about interactions between 
>>>>> GRUU
>>>>> and sips, and I think its sufficiently non-obvious that a 
>>>>> discussion is
>>>>> warranted.
>>>>>
>>>>> Specifically, under what conditions would the registrar provide the
>>>>> client with a gruu that is a sips URI?
>>>>>
>>>>> I think the right answer here is that if teh Contact URI is a sips 
>>>>> URI,
>>>>> then the registrar MUST return a gruu which is a sips URI. If the
>>>>> contact URI is not a sips URI, the registrar MUST NOT return a 
>>>>> sips URI.
>>>>>
>>>>> Other thoughts?
>>>>>
>>>>> Thanks,
>>>>> Jonathan R.
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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] Strict,
	Semi-Strict and Loose mode in RPH - not a good fit for ets and wps
To: "Ken Carlberg" <carlberg@g11.org.uk>
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        Richard F Kaczmarek <rkaczmarek@csc.com>, sip@ietf.org,
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To clarify, NCS "owns" (and it is in quotes on purpose) the GETS and WPS
programs.  As such, it will presumably be the first user of the ets and wps
namespaces.  But, no, it doesn't "own" (either literally or figuratively)
the namespaces.

Janet

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                      "Ken Carlberg"                                                                                           
                      <carlberg                To:      Janet P Gunn/FED/CSC@CSC                                               
                      @g11.org.uk>             cc:      <a.ephrath@ieee.org>, Darren E Pado/FED/CSC@CSC, Dennis Q              
                                               Berg/FED/CSC@CSC, <fonashp@ncs.gov>, <jmpolk@cisco.com>,                        
                      11/07/2004 09:16         <KENNETH.R.ERNEY@saic.com>, <mosleyv@ncs.gov>, <nyquetek@msn.com>, Richard F    
                      AM                       Kaczmarek/FED/SC/CSC@CSC, Saud Negash/FED/CSC@CSC, <sip@ietf.org>,              
                                               <suracif@ncs.gov>                                                               
                                               Subject: RE: [Sip] Strict, Semi-Strict and Loose mode in RPH - not a good fit   
                                               for ets and wps                                                                 
                                                                                                                               




> I would prefer a strict or semi-strict mode because these provide
feedback in
> case: a) a misconfigured UA sends the wrong namespace.priority, and/or b)
> initial selection of a server/proxy that does not support the desired
> namespace.  The former is quite possible given an ability (and possibly,
> necessity) of the user to configure their own device to support the given
> namespace.value combination.
>
> [JG] Before posting this, I checked with Dr. Fonash (head of NCS, which
> "owns" GETS and WPS).  He was very clear about wanting "If the SIP
element
> does not recognize the namespace (wps or ets) it should pass it on
without
> acting on it, rather than rejecting it."

ok.  Dr Fonash now has additional input on the subject given my previous
email;
if there is no feedback, then the sender hopes everything is working fine.
further, it will continue to use an R-P capable proxy with no feedback from
the
INVITE indicating that it does not support that particular namespace/value.


Keep in mind that as it states earlier in the draft, the R-P header is
ignored
(assuming proper implementation) by those proxies (eg, the installed base)
that
do not support the new R-P header.  As a sidenote, our current
implementation
of the R-P header follows the MAY portion of the draft and does not send
back
hints of what are acceptable priorities.

Also, when you say the NCS "owns" GETS and WPS, are you refering to the
systems
or the namespace?  If its both, then I have my doubts about the latter.  I
see
no association of the NCS with WPS or GETS in the draft and it would seem
that
initial "ownership" of those namespaces, as well as the others like ETS,
lies
with the authors of the draft.  (I'll be happy to be corrected on that
assumption).

-ken






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From: "Medhavi Bhatia" <mbhatia@nextone.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
Subject: RE: [Sip] Re: GRUU and things that rewrite contacts
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	Jon'" <jon.peterson@neustar.biz>
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Paul,

See inline.

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Sunday, November 07, 2004 3:16 PM
> To: mbhatia@nextone.com
> Cc: 'Jonathan Rosenberg'; sip@ietf.org; 'Robert Sparks'; 'Peterson, Jon'
> Subject: Re: [Sip] Re: GRUU and things that rewrite contacts
> 
> 
> 
> Medhavi Bhatia wrote:
> > I may have caught up to this discussion too late, but here are some
> > thoughts. I probably need to dig deeper into the GRUU draft too.
> >
> > There seem to be two cases.
> >
> > 1) SBC sits between the UA and the proxy/registrar, where the UA sends
> > requests for the proxy addressed to the SBC.
> 
> Can you clarify what you mean above - perhaps give an example?
> In general, the UA doesn't know much about what is going on. At best, it
> knows how to construct the To-uri and derive the request-uri from it,
> and then may have a loose route to insert in the message. Perhaps the
> loose route transits an SBC, and maybe it acts as a B2BUA. Or do you
> have something else in mind?
> 
[MB] I meant that the SBC was acting as an out-bound proxy to get to the
proxy/registrar. Packets are destined (at the IP level) to the SBC. The
To/request uri's are pointing to the proxy.

>  > In this case, I don't see much
> > of a concern as contact parameters are not overwritten for REGISTER.
> 
> You state this as fact. But this goes to the heart of the problem that
> we have no definition of what an SBC does. I guess yours must not
> overwrite contact parameters in a REGISTER.
> 
[MB] I assume topology hiding (THIG) is the function of SBC we are talking
about. In that case, not passing the parameters would not be the right thing
to do. The THIG only alters IP addresses in the headers involved in routing
SIP messages (like Via/Contact/Route etc).

>  > For the
> > INVITE, since the SBC knows that the Contact is a GRUU,
> 
> It does? How? I guess it can know in some cases, if it saw the register
> that requested the GRUU it might remember the returned gruu and
> recognize its use in future requests, but that would be a lot of work.
> And it won't work for gruu values that it didnt' see being assigned.
> 
> In general you just can't recognize a gruu by inspection.
>
[MB] I assume the supported header in the INVITE will contain the gruu
option tag.  The draft mentions somewhere about how a receiving UA can
detect that the Contact is a GRUU.
 
>  > it is not
> > overwritten. The last part needs SBCs to understand the GRUUs.
> 
> Please explain what kind of understanding you think you require, and how
> it is even theoretically possible to have it.
> 
> > 2) The same as above, except the UA does not know who the Registrar is.
This
> > is a tricky application and I guess not part of the usual suspects. In
this
> > case, the SBC will need to be aware of GRUUs on the REGISTER also.
> 
> I have no idea what you mean here.
> 
[MB] Service providers, especially in broadband users have been deploying
out-bound proxies which forward requests and responses between the client
and the main server similar to the case I describe in (1). However there is
one interesting variation. Consider that the registering UA does not know it
is exchanging messages with an out-bound proxy. The To/req uri as well as
the dest IP address are pointing to a proxy which then uses some other means
to target the request to the UA's real proxy:

(1) Request-uri/To point to the proxy (P) and dest IP points to the SBC. 
(2) Request-uri/To/dest IP all point to the SBC. SBC forwards request to P.

In both cases, SBC is acting as an out-bound proxy in a functional sense.
However in (2), the UA does not know about P. I assume that the GRUU which
is assigned to the UA also cannot point to the proxy and the domain part
must be changed by the SBC.

> > The SBC also may have an option to create a GRUU and map one obtained
from
> > the Registrar to this one.
> 
> I think that was more or less what Jonathan was suggesting it might do
> in his original posting. It is pretty ugly.
> 
[MB] I don't think it is necessary. I may be wrong of course.

> I have the distressing feeling that SBCs are just doing random stuff,
> which has lots of potential unintended consequences.
> 
[MB] Yep. SBCs implement a lot of features. Some of the them were mentioned
earlier on another thread. However there are clearly some things which must
not be done (like re-writing contact parameters) and some things which it
must be able to decipher when it processes packets (like media description
for the message). These functions are provided today only via B2BUAs and
most people designing SBCs or planning to make one don't always know the
requirements clearly. Application Servers based on B2BUAs deal with the same
problem. I know of core proxies manufactured by some major vendors which are
based on B2BUAs.

At minimal we need to see how requirements of intermediate elements and
networks, like THIG and media relays are factored in when designing
features. We need to look at requirements of service providers in greater
detail.


> 	Paul
> 
> > -Medhavi.
> >
> >
> >>-----Original Message-----
> >>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Paul
> >
> > Kyzivat
> >
> >>Sent: Tuesday, September 21, 2004 9:13 AM
> >>To: Jonathan Rosenberg
> >>Cc: sip@ietf.org; Robert Sparks; Peterson, Jon
> >>Subject: Re: [Sip] Re: GRUU and things that rewrite contacts
> >>
> >>This would be a good time for those that build SBCs to take note and
> >>action, and to speak up if they have difficulty with the deployment of
> >>GRUUs.
> >>
> >>	Paul
> >>
> >>Jonathan Rosenberg wrote:
> >>
> >>>Jon,
> >>>
> >>>I agree with you and Paul that we cannot, and should not, try here or
> >>>anywhere else to define an SBC and consider its impacts. I think we can
> >>>and should discuss the impact of contact-rewriting elements, and point
> >>>out, as you say, that the identity draft would detect such a case.
> >>>
> >>> From a practical matter, I will say that the fact that SBCs will
likely
> >>>not work with GRUU is a real concern. It is going to make it hard to
> >>>deploy gruu, and generally I am a really big fan of easy-to-deploy
> >>>solutions. For better or worse, there ARE SBCs deployed in many
existing
> >>>SIP networks, and in those networks, getting gruu deployed will be
> >>>harder. It will require coordination now between not just the client
> >>>vendor and SIP server vendor, but now also the SBC vendor. In my
> >>>experience, the more parts of the machine that all need to be
> >>>simultaneously upgraded for a new feature, the less likely,
> >>>exponentially I think, it is that such an upgrade will occur. If you
> >>>couple this with the fact that, while gruu is definitely a key part of
> >>>our specs, its an infrastructure improvement with no new features per
> >>>se, it becomes a hard sell.
> >>>
> >>>I still believe that the only alternative on the table - using a new
> >>>header field for conveying gruu in register and invite/200 - has an
even
> >>>worse deployment path.
> >>>
> >>>So, absent a better suggestion, and I don't have one, I think our only
> >>>choice is to document the problems per above.
> >>>
> >>>-Jonathan R.
> >>>
> >>>
> >>>
> >>>Peterson, Jon wrote:
> >>>
> >>>
> >>>>I agree with Paul that SBC behavior is probably defined too poorly to
> >>>>refer
> >>>>to it directly in the GRUU draft, and personally I'm very skeptical of
> >>>>taking on that definition as a deliverable for GRUU. At the risk of
> >>>>complicating a discussion about non-standard entities with standards:
> >>>>
> >>>>RFC3261, Table 2:
> >>>>
> >>>>      Header field          where   proxy ACK BYE CAN INV OPT REG
> >>>>      ___________________________________________________________
> >>>>      Contact                 R            o   -   -   m   o   o
> >>>>
> >>>>
> >>>>I don't see an "m" under "proxy" there... but of course we're not
> >>>
> > talking
> >
> >>>>about proxy servers as such. That much said, if Contacts were
> >>>>something that
> >>>>an entity other than a user agent could safely set, there would be an
> >>>>"m" in
> >>>>the table up there. If an intermediary wants to request that it be in
> >>>
> > the
> >
> >>>>path of further SIP transactions in a dialog in a safe way, well,
> >>>
> > there's
> >
> >>>>another header for that. Furthermore, the GRUU acquisition mechanism
> >>>>inherently provides a way for intermediaries to give a Contact header
> >>>>to a
> >>>>user agent in order to strongly bind request routing to an
> >>>>intermediary. So,
> >>>>there are reasonable alternatives to rewriting the header in transit.
> >>>>
> >>>>Furthmore, I would like to point out that sip-identity provides
> >>>
> > integrity
> >
> >>>>over the addr-spec of the Contact header field. If the Contact header
> >>>>field
> >>>>has been modified by an intermediary after integrity has been applied,
> >>>
> > it
> >
> >>>>should be considered an integrity violation by the recipient.
> >>>>
> >>>>This is of course not to say that there are no architectures where
> >>>>SBCs and
> >>>>sip-identity might live together in harmony. The point is that SIP
> >>>>can't be
> >>>>expected to differentiate between friendly and unfriendly
> >>>
> > intermediaries
> >
> >>>>that break the rules, and that we certainly can't enable arbitary
> >>>>entities
> >>>>to modify the Contact header or we open ourselves up to all sorts of
> >>>>nasty
> >>>>attacks. Whatever security gains people think they are getting from
> >>>>rewriting the Contact header at an intermediary, I'm sure they pale in
> >>>>comparison to the threats that the arise if Contact header can be
> >>>>arbitarily
> >>>>changed by anyone. If we want to prevent something from rewriting a
> >>>>header
> >>>>that shouldn't be rewritten, we should use some form of integrity to
> >>>>prevent
> >>>>that.
> >>>>
> >>>>So, I do think it is reasonable for the GRUU doc to discuss the impact
> >>>>of an
> >>>>intermediary rewriting the Contact header in general (without defining
> >>>>SBCs), and point out that integrity mechanisms like sip-identity could
> >>>
> > be
> >
> >>>>used to detect a rewrite of a Contact header. But, on some level, I
> >>>>guess I
> >>>>feel rewriting Contacts at an intermediary is already acknowledged to
> >>>
> > be
> >
> >>>>problematic by the core SIP standard. It'd also be a problem for GRUU
> >>>>if an
> >>>>SBC removed the Call-ID header field, or did any of a number of things
> >>>
> > it
> >
> >>>>shouldn't do.
> >>>>
> >>>>Jon Peterson
> >>>>NeuStar, Inc.
> >>>>
> >>>>
> >>>>
> >>>>>-----Original Message-----
> >>>>>From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >>>>>Sent: Thursday, September 16, 2004 4:32 PM
> >>>>>To: Jonathan Rosenberg
> >>>>>Cc: sip@ietf.org; Robert Sparks
> >>>>>Subject: Re: [Sip] Re: GRUU and things that rewrite contacts
> >>>>>
> >>>>
> >>>>[snip]
> >>>>
> >>>>
> >>>>>I don't want to comment on the above right now because we are clearly
> >>>>>have different assumptions here.
> >>>>>
> >>>>>I don't think we can meaningfully comment on the impact on SBCs when
> >>>>>we have no definition of such a beast. And trying to generalize to
> >>>>>all forms of B2BUA makes things worse - we repeatedly state that
> >>>>>about all you can say in general about a B2BUA is that it is two
> >>>>>connected UAs - you can't say anything about how the two calls are
> >>>>>related.
> >>>>>
> >>>>>If this is an important problem for us to mention at all, (and I'm
> >>>>>not yet convinced it is), then I think the first step is to define an
> >>>>>SBC, along with the kinds of transformations it performs. Since that
> >>>>>has never been written down, when it is written it can deal properly
> >>>>>with GRUUs.
> >>>>>
> >>>>>    Paul
> >>>>>
> >>>>
> >>>>
> >>
> >>_______________________________________________
> >>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 makes a lot of sense to me when we authorize using the TLS peer name.
However, it seem like creating an alias using the name if the TLS peerName
instead of the name in the Via is a better plan. Otherwise we need to
describe what names in the via are legal for a given peerName in the TLS.
The alias flag in the via would still be the hint that an alias should be
created. 




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comments in-line below

At 11:47 PM 11/6/2004 -0500, Janet P Gunn wrote:
>From: "Ken Carlberg"
> > Unfortunately, this  (semi-strict) is NOT the behavior desired for the
>ets
> > and wps namespaces.

Janet - I like your list. It is quite valuable because it gives very good 
points for discussion.  As Ken has stated already, a couple of issues from 
your list (I believe) are already addressed within this current version.

> >
> > The behavior desired for the ets and wps namespaces are
> > 1 RPH is optional (semi-strict and loose)

got it

> > 2 SIP element ignores a namespace it does not recognize (loose)

To this, remember the ID does not say to "fail" the call, just that 
Request. This is one way to inform the calling UAC to know it included a 
namespace or priority value that was not recognizable (meaning not 
ets.0/1/2/3/4 or wps.0/1/2/3/4).

There is no way to inform the UAC of this issue within the SIP Request 
(with the bad namespace.priority-value) without failing the request. 
Meaning, there can be no preferential treatment without the sending element 
being told so. SIP doesn't have a "pass, but this wasn't a good choice" 
provisional response like other protocols.

To the user, they might not notice several SIP Request failures, the person 
is merely waiting for the call to be set up (still under a second).


>I would prefer a strict or semi-strict mode because these provide feedback
>in
>case: a) a misconfigured UA sends the wrong namespace.priority, and/or b)
>initial selection of a server/proxy that does not support the desired
>namespace.  The former is quite possible given an ability (and possibly,
>necessity) of the user to configure their own device to support the given
>namespace.value combination.
>
>[JG] Before posting this, I checked with Dr. Fonash (head of NCS, which
>"owns" GETS and WPS).  He was very clear about wanting "If the SIP element
>does not recognize the namespace (wps or ets) it should pass it on without
>acting on it, rather than rejecting it."

again - this doesn't mean the call attempt has failed and the user needs to 
dial again  - which is one way to read what you wrote. The text in this doc 
clearly needs to not call for failing the call in the ets/wps use-case 
because of a bad namespace or priority-value.

Ironically this is where the A-R-P header would be handy, if it didn't 
reveal to everyone who purposely or inadvertently included a bad or unknown 
RP header.


> > 3 The content of the error message should not reveal (to the end user)
>the
> > valid values for that namespace (strict)
>
>section 4.5 of the drafts states:
>
>    If the UAS understands the resource value, but refuses to honor the
>    request with elevated priority for this particular user, it returns
>    the 403 (Forbidden) response code.  It MAY include the list of
>    resource values that the user is allowed to use in the
>    'Accept-Resource-Priority' response header field.
>
>The "MAY" is the critical term.

I feel this MAY answers your concern - as Ken points out. You may mandate 
vendor's equipment be able to be configured to *not* send the A-R-P in this 
case. That is a useful marketing requirement.

>My understanding would be that this
>decision
>is a local policy issue.  Outside of that I personally don't see a problem
>with
>providing "hints" of accepted values under the context that they can only
>acted
>upon by authorized users.  Informing the user of the acceptable set of
>priorities does not bypass or circumvent the need for proper authentication
>credentials.
>
>[JG]I am not as concerned about this case (valid namespace.valid value -
>just not authorized).

I would think this is of concern - as anyone (literally) can be in this group.

>I am more concerned about the  error message when
>the namespace, or value are not recognized.  But if ets and wps operate
>with the behavior "SIP element ignores a namespace it does not recognize",
>half of that goes away.
>
>[JG]I am still not entirely sure about the desired behavior for the other
>half- (valid namespace.invalid value).  But I suspect that it would be
>better to
>a) Reject the message
>b) NOT return the list of valid values.

I consider this to be the case every time a message is not accurate, but 
want to discuss it with you to tease out any "ships in the night" scenarios 
we're missing


>[JG] I do not think that the full loose mode as defined (if the SIP element
>gets (valid namespace.invalid value) it processes the message as if it had
>no RPH) is the right approach.

nor do I

>I think these messages should be rejected,
>and get an error message.

I agree


>[JG]But I am open to other approaches.

we have time to talk



> > 4 SIP messages that include the RPH with the ets or wps namespaces should
> > include the authorization header (or equivalent mechanism) (semi-strict)

this should be self evident in public networks....

> > 5 Precedence (queueing) is generally the means of providing priority, but
> > exemption from network management controls, for instance call admission
> > within an IP domain, when a "regular" call would not be admitted, is also
> > needed.  Pre-emption is not used.  (semi-strict "plus more")

I do not understand what you have written above, can you rephrase please?


>the "plus more" would seem to be a product of local policy
>
>[JG] Agree.  While it doesn't explicitly state it, the document leads you
>to believe that queueing is the only behavior permitted in semi-strict
>mode.

This is stated a couple of times in the doc, including in the IANA section 
in the table


> > 6 There is no a priori requirement to use, or not use, the "Require"
> > header field.
>
>isn't this addressed in section 4.3.2 below?  or am I missing something?
>
>    If the request includes a 'Require' header field with the
>    'Resource-Priority' option tag, a UAS MUST follow the strict or
>    semi-strict mode rules, otherwise UAS and proxies MUST operate in
>    loose mode.
>
>
>[JG]I guess I wasn't clear.   I simply meant that there wasn't a concern
>about the "require" header itself, independent of the
>strict/semi-strict/loose mode.  Once that is "right", the approach to the
>"require" header will follow

hmmmm

The "Require" header is to mandate that SIP Responses use the header as is 
(copied from request to response without change). If ever there is a case 
needed in which preferential treatment is within the processing of SIP 
elements (such as Proxies), knowing there will always be more than one SIP 
message necessary to complete a transaction - meaning both Request and 
Response will need to be RP marked for preferential treatment for the 
session/transaction to be given priority over other transactions.

To put this simply, if the INVITE is consistently moved to the head of the 
line in all SIP elements between UAs 1 & 2, and the 200 OK is 
slowed/delayed or lost in the congestion of one or more Proxies, the call 
will not be established. In this case, the INVITE, 200 OK and ACK will need 
the RP indication to gain the preferential treatment of that domain; and 
any other messages within that transaction will require it too (such as 
183, PRACK and UPDATE messages if RFC 3312 Preconditions were to be 
required to set up, say RSVP between endpoints).


>-ken


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Date: Mon, 08 Nov 2004 09:12:11 -0500
From: Jonathan Rosenberg <jdrosen@cisco.com>
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To: Adam Roach <adam@nostrum.com>
Subject: Re: [Sip] GRUU and sips
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Cc: Alan Johnston <alan.johnston@mci.com>, Cullen Jennings <fluffy@cisco.com>,
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Adam Roach wrote:

> I'm getting in a bit late on this thread (sorry), but I've been under=20
> the impression for quite some time that sip:foo and sips:foo are=20
> *defined* to be the same resource under all circumstances. Section 19.1=
=20
> of 3261 seems rather clear on this point:
>=20
>>    Any resource described by a SIP URI can be
>>    "upgraded" to a SIPS URI by just changing the scheme, if it is
>>    desired to communicate with that resource securely.
>=20
>=20
>=20
> So, if we just hand out sip: URIs, then full 3261 implementations *know=
*=20
> that they can use it with a sips: scheme and get the same resource.=20

Thanks for pointing this out; I had forgotten this was in there. I think=20
this at least implies that the two URIs, if they exist, point to the=20
same resource. But, it doesnt address questions like whether both have=20
to exist, or more specific questions about sips URI in GRUU.


> Perhaps it is worth reiterating this point in the GRUU document, but it=
=20
> seems like this is a non-problem.

I think we need to say the following:

If a sip and sips URI exist, they should point to the same resource
   Existence of sip doesn=92t imply sips and vice a versa
If contact is sip, gruu is a sip URI, but the server creates the sips
   resource (i.e., it will accept and handle sips if it can do a secure
   connection)
If the contact is sips, the gruu is sips, but the server does not create
   the sip resource
     Allows for sips only resources

I'll be talking about this today in sip

Thanks,
Jonathan R.

>=20
> /a
>=20
>=20
> Paul Kyzivat wrote:
>=20
>> Alan,
>>
>> Did you see my later message on this subject?
>>
>> As far as bid down, that is only a problem if the UAS isn't aware that=
=20
>> it happened. Note that no matter what we say, bad guys can make the=20
>> change, so it isn't the change that matters, but how it is handled=20
>> along the path.
>>
>> In a direct connection, the UAS can see whether the r-uri is sip/sips.=
=20
>> If the r-uri is sip it may conclude that the session isn't secure. For=
=20
>> indirect connections it becomes the obligation of the proxies along=20
>> the path to use a sip r-uri if the connection can't be guaranteed=20
>> secure to them.
>>
>> My concern, beyond understanding how GRUU should work, is that a UAS=20
>> that wants to support both secure and insecure communication should=20
>> only need to make one registration. I have feeling people widely think=
=20
>> things should work this way, but the standard isn't clear if/how this=20
>> can be achieved.
>>
>>     Paul
>>
>> Alan Johnston wrote:
>>
>>> At 10:35 AM 7/27/2004 -0400, Paul Kyzivat wrote:
>>>
>>>
>>>> Cullen Jennings wrote:
>>>>
>>>>> Sounds like a possibility. Can you think of a way where I do not=20
>>>>> have to
>>>>> Register twice to get both a sip and sips URL?
>>>>> Perhaps we should just say that is a proxy hands out a sip GRUU=20
>>>>> sip:foo then
>>>>> sips:foo MUST also be a valid GRUU with the contact moved to sips=20
>>>>> from sip.
>>>>
>>>>
>>>>
>>>>
>>>> That would rule out proxies that support gruu but not TLS.
>>>>
>>>> I think the converse might make more sense - if the proxy hands out=20
>>>> sips:foo then sip:foo MUST also be a valid GRUU.
>>>
>>>
>>>
>>>
>>> Are you sure this is a good idea?  Doesn't this open a bid down=20
>>> attack for the GRUU + grid dialog identifier described by Robert in=20
>>> his dialog reuse draft?
>>>
>>> Here's the example - I have a secure SIP dialog with you.  You are=20
>>> using a secure SIP GRUU + grid dialog identifier.  I pass this GRUU=20
>>> to another party, such as Robert with a Replaces header field in a=20
>>> REFER for the intent to transfer you to Robert.  Robert changes the=20
>>> URI to a regular SIP GRUU and sends the INVITE with Replaces.  If you=
=20
>>> accept this, you have replaced a secure SIP session with an insecure=20
>>> session - not a good idea.  Better for this transfer to fail if=20
>>> Robert tries to downgrade the security on the connection.
>>>
>>> I would say a sips:foo GRUU MUST NOT be valid as sip:foo to avoid=20
>>> this kind of attack.
>>>
>>> Thanks,
>>> Alan Johnston
>>>
>>>
>>>>         Paul
>>>>
>>>>> If I get a sip GRUU, others can still use TLS to contact it. I=20
>>>>> would like to
>>>>> make sure that the following case works:  a UA only does UDP to=20
>>>>> it's proxy,
>>>>> however, it want communications from outside the domain to use TLS=20
>>>>> to the
>>>>> proxy. I think this works fine with what you are proposing.
>>>>> On 7/23/04 12:27 PM, "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>=
=20
>>>>> wrote:
>>>>>
>>>>>> The current GRUU spec says very little about interactions between=20
>>>>>> GRUU
>>>>>> and sips, and I think its sufficiently non-obvious that a=20
>>>>>> discussion is
>>>>>> warranted.
>>>>>>
>>>>>> Specifically, under what conditions would the registrar provide th=
e
>>>>>> client with a gruu that is a sips URI?
>>>>>>
>>>>>> I think the right answer here is that if teh Contact URI is a sips=
=20
>>>>>> URI,
>>>>>> then the registrar MUST return a gruu which is a sips URI. If the
>>>>>> contact URI is not a sips URI, the registrar MUST NOT return a=20
>>>>>> sips URI.
>>>>>>
>>>>>> Other thoughts?
>>>>>>
>>>>>> Thanks,
>>>>>> Jonathan R.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>
>>
>=20
>=20

--=20
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Subject: RE: [Sip] Strict,
	Semi-Strict and Loose mode in RPH - not a  good fit for ets and wps
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James said,

>[JG]I am not as concerned about this case (valid namespace.valid value -
>just not authorized).

I would think this is of concern - as anyone (literally) can be in this
group.

I was thinking that it was of less concern because it is someone who
already knows the valid namespace and value.

But you are right, if it is someone who is randomly or systematically
trying namespace/value combinations, we just told them "you got a hit".
So I take it back about being less concerned.

Janet



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I got no answers on this question last time so I try again.

=20

Is the approach described below so naive that no one even has the energy
point out its weaknesses? Maybe it has already been discussed and deemed
inadequate?

=20

/ Christian

=20

=20

  _____ =20

From: Christian Jansson=20
Sent: den 21 oktober 2004 16:05
To: sip@ietf.org
Subject: [Sip] Join header (draft-ietf-sip-join-03.txt) unnecessary?
[html]

=20

Why do we need a Join header to be able to indicate that we want to join
an ongoing session? Wouldn't it be enough to, for the party that wants
to join, actually use the call-id for the session when sending the
join-INVITE? Suppose A and B are in a session with the dialog identified
by call-id=3D12345;tag=3DA;tag=3DB. If C now wants to join the session =
he
sends an INVITE to A or B with call-id=3D12345, from-tag=3DC, and an =
empty
to-tag. Wouldn't it make more sense that the 2 dialog-ids had some parts
in common, that is the call-id, and something that distinguished them,
that is the to and from tags? It would be much easier to find the
session related messages as they would share the same call-id.

=20

Have I missed some essential part of the Join header, or are we just
adding something that we could already express easily with what we
already have?

=20

=20

/ Christian Jansson, Hotsip


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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I got no answers on this question =
last
time so I try again.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Is the approach described below so =
naive that
no one even has the energy point out its weaknesses? Maybe it has =
already been
discussed and deemed inadequate?<o:p></o:p></span></font></p>

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

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

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

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

<div>

<div class=3DMsoNormal align=3Dcenter =
style=3D'margin-left:36.0pt;text-align:center'><font
size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

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

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

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> <st1:PersonName
w:st=3D"on">Christian Jansson</st1:PersonName> <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> den 21 oktober 2004 =
16:05<br>
<b><span style=3D'font-weight:bold'>To:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] Join =
header
(draft-ietf-sip-join-03.txt) unnecessary? =
[html]</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Why do we need a Join =
header to be
able to indicate that we want to join an ongoing session? Wouldn&#8217;t =
it be
enough to, for the party that wants to join, actually use the call-id =
for the
session when sending the join-INVITE? Suppose A and B are in a session =
with the
dialog identified by call-id=3D12345;tag=3DA;tag=3DB. If C now wants to =
join the
session he sends an INVITE to A or B with call-id=3D12345, from-tag=3DC, =
and an
empty to-tag. Wouldn&#8217;t it make more sense that the 2 dialog-ids =
had some
parts in common, that is the call-id, and something that distinguished =
them,
that is the to and from tags? It would be much easier to find the =
session
related messages as they would share the same =
call-id.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></fo=
nt></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Have I missed some =
essential part of
the Join header, or are we just adding something that we could already =
express
easily with what we already have?<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></fo=
nt></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></fo=
nt></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>/ <st1:PersonName =
w:st=3D"on">Christian
 Jansson</st1:PersonName>, Hotsip<o:p></o:p></span></font></p>

</div>

</body>

</html>

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Use sip-implementors@cs.columbia.edu for questions on current sip
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--===============0027131194==--



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Subject: RE: [Sip] Strict, Semi-Strict and Loose mode in RPH - not a 
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At 09:13 AM 11/8/2004 -0500, Janet P Gunn wrote:
>James said,
>
> >[JG]I am not as concerned about this case (valid namespace.valid value -
> >just not authorized).
>
>I would think this is of concern - as anyone (literally) can be in this
>group.
>
>I was thinking that it was of less concern because it is someone who
>already knows the valid namespace and value.

the RPH document is in the public domain - and we are having a fairly 
public discussion about it (and its ramifications). Curious eyes are 
probably reading this too.

;-)


>But you are right, if it is someone who is randomly or systematically
>trying namespace/value combinations, we just told them "you got a hit".
>So I take it back about being less concerned.
>
>Janet


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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From sip-bounces@ietf.org  Mon Nov  8 10:54:13 2004
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Thanks for the read, Adam. Some responses:

> > Any discrepancies or violations MUST be reported to the user.
> 
> 
> If I implement a proxy, it is not clear how I can satisfy 
> this second MUST.

Agreed, that should be clarified. Furthermore, this is normative behavior
about authorization decisions that will be made given the presence/absence
of identity in a message, and technically, such authorization is outside the
scope of the document.

> A secondary sub-issue arises from the mechanism by which the proxy 
> decides to trust certificates. There is nothing in the current draft 
> that suggests, for example, that trusting self-signed domain certs for 
> the purposes of user authentication might be a Really Bad Idea. (Obvious 
> to security wonks, maybe not so much for your average implementor).

Really, the text should be more specific about this, and it will be. While
we shouldn't go so far as to identify particular root CAs that need to be
supported, we intend for these to be publicly-verifiable certificates.

> The 
> reason this ties into user interaction is that it becomes unclear how a 
> proxy should react if it receives a request; that request has an 
> authenticated identity asserted by a domain; and the root CA for the 
> domain's certificate is not known. With user applications, this is 
> typically handled by saying, "I don't know who blackhelicopers.org is, 
> but they signed this certificate. Click here to examine the cert 
> yourself, and let me know if I should accept this cert as invalid, 
> temporarily valid, or permanently valid." When the check is performed by 
> a proxy, such interaction is really quite difficult. The best a proxy 
> can do is reject the request with a 428 (which seems a bit extreme), or 
> let it through (in which case the proxy's check serves little 
> to no purpose)
> 
> So, I suggest that the draft probably needs a little more text around 
> proxy handling of requests; further, given the all-or-nothing nature of 
> proxy validation of authenticated identity, we might consider 
> discouraging its use in favor of client validation.
> 

Clearly, client validation is a superior option, yes. But, I do think there
are some authorization policies that are best implemented at a domain level,
and it is reasonable for intermediaries to instantiate those policies rather
than receiving user agents. We do need to solve for the case of how that
authorization decision will be executed, but as I've said, authZ should be
outside the scope of this draft for the most part. The only sort of authZ
action that we could reasonably expect an intermediary to perform is to
reject the request in some fashion, as you suggest.

> The second issue is a tiny nit in that the current draft doesn't specify 
> what to do if you receive a request with an Identity-Info header which 
> contains a URI with a scheme other than sips: or https:. Presumably, 
> recipients should 428 such requests? 400?

Yeah, I think I'm going to relax that restriction for other reasons (there
is at least one other URI type, "cid:", that I think would be useful here).
Ultimately, I think that if you can dereference the URI and acquire the
necessary keying material, you shouldn't complain. There should be an error
response that says, "I can't find dereference that URI" (maybe because I
don't support the scheme, maybe because the link is broken, etc). We know we
also want to describe the case where the root CA is unsupported (and
subsumed under that, the possibility that the cert used to generate the
identity digest sig is self-signed). Identifying response codes for these
conditions is needed, I agree.

Jon Peterson
NeuStar, Inc.

> /a
> 

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Subject: Re: [Sip] Join header (draft-ietf-sip-join-03.txt) unnecessary?
	Second attempt.
Date: Mon, 8 Nov 2004 10:30:21 -0500
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Hi,

Sorry for the delayed response.

Based on our experiences with Replaces, Dan Petrie, Alan Johnston,=20
Robert Sparks, myself and other members of the working group where able=20=

to point out significant problems with reusing dialog identifiers for=20
multiple dialogs.  The simplest example is that a Replaces header sent=20=

to a locally mixed conference might match multiple dialogs.  It is then=20=

impossible to replace a single (specific) dialog.  Also Join allows a=20
user agent to redirect a Join request to a conference in a much more=20
natural way.  Hope this helps.  In any case, Join is now RFC3911.

thanks,
-rohan


On Nov 8, 2004, at 9:45, Christian Jansson wrote:

> I got no answers on this question last time so I try again.
>
> =A0
>
> Is the approach described below so naive that no one even has the=20
> energy point out its weaknesses? Maybe it has already been discussed=20=

> and deemed inadequate?
>
> =A0
>
> / Christian
>
> =A0
>
> =A0
>
>
> From: Christian Jansson
> Sent: den 21 oktober 2004 16:05
> To: sip@ietf.org
> Subject: [Sip] Join header (draft-ietf-sip-join-03.txt) unnecessary?=20=

> [html]
>
> =A0
>
> Why do we need a Join header to be able to indicate that we want to=20
> join an ongoing session? Wouldn=92t it be enough to, for the party =
that=20
> wants to join, actually use the call-id for the session when sending=20=

> the join-INVITE? Suppose A and B are in a session with the dialog=20
> identified by call-id=3D12345;tag=3DA;tag=3DB. If C now wants to join =
the=20
> session he sends an INVITE to A or B with call-id=3D12345, from-tag=3DC,=
=20
> and an empty to-tag. Wouldn=92t it make more sense that the 2 =
dialog-ids=20
> had some parts in common, that is the call-id, and something that=20
> distinguished them, that is the to and from tags? It would be much=20
> easier to find the session related messages as they would share the=20
> same call-id.
>
> =A0
>
> Have I missed some essential part of the Join header, or are we just=20=

> adding something that we could already express easily with what we=20
> already have?
>
> =A0
>
> =A0
>
> / Christian Jansson, Hotsip

--Apple-Mail-1--785946357
Content-Type: text/enriched;
	charset=WINDOWS-1252
Content-Transfer-Encoding: quoted-printable

Hi,


Sorry for the delayed response.


Based on our experiences with Replaces, Dan Petrie, Alan Johnston,
Robert Sparks, myself and other members of the working group where
able to point out significant problems with reusing dialog identifiers
for multiple dialogs.  The simplest example is that a Replaces header
sent to a locally mixed conference might match multiple dialogs.  It
is then impossible to replace a single (specific) dialog.  Also Join
allows a user agent to redirect a Join request to a conference in a
much more natural way.  Hope this helps.  In any case, Join is now
RFC3911.


thanks,

-rohan



On Nov 8, 2004, at 9:45, Christian Jansson wrote:


=
<excerpt><fontfamily><param>Arial</param><color><param>0000,0000,8080</par=
am><x-tad-bigger>I
got no answers on this question last time so I try =
again.</x-tad-bigger></color></fontfamily>


=
<fontfamily><param>Arial</param><color><param>0000,0000,8080</param><x-tad=
-bigger>=A0</x-tad-bigger></color></fontfamily>


=
<fontfamily><param>Arial</param><color><param>0000,0000,8080</param><x-tad=
-bigger>Is
the approach described below so naive that no one even has the energy
point out its weaknesses? Maybe it has already been discussed and
deemed inadequate?</x-tad-bigger></color></fontfamily>


=
<fontfamily><param>Arial</param><color><param>0000,0000,8080</param><x-tad=
-bigger>=A0</x-tad-bigger></color></fontfamily>


=
<fontfamily><param>Arial</param><color><param>0000,0000,8080</param><x-tad=
-bigger>/
Christian</x-tad-bigger></color></fontfamily>


=
<fontfamily><param>Arial</param><color><param>0000,0000,8080</param><x-tad=
-bigger>=A0</x-tad-bigger></color></fontfamily>


=
<fontfamily><param>Arial</param><color><param>0000,0000,8080</param><x-tad=
-bigger>=A0</x-tad-bigger></color></fontfamily>



=
<bold><fontfamily><param>Tahoma</param><x-tad-bigger>From:</x-tad-bigger><=
/fontfamily></bold><fontfamily><param>Tahoma</param><x-tad-bigger>
Christian Jansson </x-tad-bigger></fontfamily>

=
<bold><fontfamily><param>Tahoma</param><x-tad-bigger>Sent:</x-tad-bigger><=
/fontfamily></bold><fontfamily><param>Tahoma</param><x-tad-bigger>
den 21 oktober 2004 16:05</x-tad-bigger></fontfamily>

=
<bold><fontfamily><param>Tahoma</param><x-tad-bigger>To:</x-tad-bigger></f=
ontfamily></bold><fontfamily><param>Tahoma</param><x-tad-bigger>
sip@ietf.org</x-tad-bigger></fontfamily>

=
<bold><fontfamily><param>Tahoma</param><x-tad-bigger>Subject:</x-tad-bigge=
r></fontfamily></bold><fontfamily><param>Tahoma</param><x-tad-bigger>
[Sip] Join header (draft-ietf-sip-join-03.txt) unnecessary? =
[html]</x-tad-bigger></fontfamily>


<fontfamily><param>Times New =
Roman</param><bigger><bigger>=A0</bigger></bigger></fontfamily>


<fontfamily><param>Arial</param><x-tad-bigger>Why do we need a Join
header to be able to indicate that we want to join an ongoing session?
Wouldn=92t it be enough to, for the party that wants to join, actually
use the call-id for the session when sending the join-INVITE? Suppose
A and B are in a session with the dialog identified by
call-id=3D12345;tag=3DA;tag=3DB. If C now wants to join the session he =
sends
an INVITE to A or B with call-id=3D12345, from-tag=3DC, and an empty
to-tag. Wouldn=92t it make more sense that the 2 dialog-ids had some
parts in common, that is the call-id, and something that distinguished
them, that is the to and from tags? It would be much easier to find
the session related messages as they would share the same =
call-id.</x-tad-bigger></fontfamily>


=
<fontfamily><param>Arial</param><x-tad-bigger>=A0</x-tad-bigger></fontfami=
ly>


<fontfamily><param>Arial</param><x-tad-bigger>Have I missed some
essential part of the Join header, or are we just adding something
that we could already express easily with what we already =
have?</x-tad-bigger></fontfamily>


=
<fontfamily><param>Arial</param><x-tad-bigger>=A0</x-tad-bigger></fontfami=
ly>


=
<fontfamily><param>Arial</param><x-tad-bigger>=A0</x-tad-bigger></fontfami=
ly>


<fontfamily><param>Arial</param><x-tad-bigger>/ Christian Jansson,
Hotsip</x-tad-bigger></fontfamily>

</excerpt>=

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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
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From sip-bounces@ietf.org  Mon Nov  8 11:06:55 2004
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Subject: RE: [Sip] Strict,
	Semi-Strict and Loose mode in RPH - not a   good fit for ets and wps
To: "James M. Polk" <jmpolk@cisco.com>
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OK, to turn it around. If "curious eyes" already can know the valid
namespace/value combinations, simply by navigating the ietf site, why
bother to hide it in the protocol messages? At least for the "published"
namespaces.


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                      "James M. Polk"                                                                                          
                      <jmpolk                  To:      Janet P Gunn/FED/CSC@CSC                                               
                      @cisco.com>              cc:      a.ephrath@ieee.org, "Ken Carlberg" <carlberg@g11.org.uk>, Darren E     
                                               Pado/FED/CSC@CSC, Dennis Q Berg/FED/CSC@CSC, fonashp@ncs.gov,                   
                      11/08/2004 10:20         KENNETH.R.ERNEY@saic.com, mosleyv@ncs.gov, nyquetek@msn.com, Richard F          
                      AM                       Kaczmarek/FED/SC/CSC@CSC, Saud Negash/FED/CSC@CSC, sip@ietf.org,                
                                               suracif@ncs.gov                                                                 
                                               Subject: RE: [Sip] Strict, Semi-Strict and Loose mode in RPH - not a   good fit 
                                               for ets and wps                                                                 
                                                                                                                               




At 09:13 AM 11/8/2004 -0500, Janet P Gunn wrote:
>James said,
>
> >[JG]I am not as concerned about this case (valid namespace.valid value -
> >just not authorized).
>
>I would think this is of concern - as anyone (literally) can be in this
>group.
>
>I was thinking that it was of less concern because it is someone who
>already knows the valid namespace and value.

the RPH document is in the public domain - and we are having a fairly
public discussion about it (and its ramifications). Curious eyes are
probably reading this too.

;-)


>But you are right, if it is someone who is randomly or systematically
>trying namespace/value combinations, we just told them "you got a hit".
>So I take it back about being less concerned.
>
>Janet


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented






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From sip-bounces@ietf.org  Mon Nov  8 12:08:13 2004
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Subject: RE: [Sip] Join header (draft-ietf-sip-join-03.txt) unnecessary?
	Second attempt.
Date: Mon, 8 Nov 2004 17:33:28 +0100
Message-ID: <B7192C0D8D60754DADA9E22294C57369631284@mailserver.hotsip.com>
Thread-Topic: [Sip] Join header (draft-ietf-sip-join-03.txt) unnecessary?
	Second attempt.
Thread-Index: AcTFp5SjhMfP9/OBSTapWQQwo/5T0AAAOcaw
From: "Christian Jansson" <christian.jansson@hotsip.com>
To: "Rohan Mahy" <rohan@ekabal.com>
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I saw that the draft had made it to RFC so in some way I am too late, =
but I still do not see the advantages with Join.

"The simplest example is that a Replaces header sent to a locally mixed =
conference might match multiple dialogs"
Is that an advantage or disadvantage of Join vs the model I proposed?=20

"It is then impossible to replace a single (specific) dialog"
But wouldn't that be possible with the proposal I wrote also? I think =
so. Do you have any reference to mailing-list discussions where you =
realized that what was in RFC3261 was not enough?

I thought it would be good to reuse the elements that are in 3261. In a =
way dialogs could be seen to be coupled to a call-id in a heretical =
view.

In the case below UA:A has a call containing 2 dialogs with dialog-ids =
(1234:A1:BB) and (1234:A2:CC). If one would like to use Replaces to =
replace one of the dialogs then simply target one of them using the =
dialog id. To me Join seems to invent the wheel again.

     UA:A

    Call-id: 1234
     |          |
 local-tag=3DA1  local-tag=3DA2
 remote-tag=3DBB remote-tag=3DCC

In this model it is very easy to see that the two dialogs are connected, =
as they both share the call-id part of their dialog-id. I think that is =
a very attractive feature.

Now if we use Join instead do the same thing, we would get some state =
above this (conference?) that relates one dialog with another dialog =
with a completely different set of identifiers. It also makes the =
call-id identifier more or less useless except for the case when forking =
gives back multiple answers, which probably won't happen under normal =
operation. Does this state above dialogs and call-ids have a name =
defined anywhere?=20


/ Christian Jansson


________________________________________
From: Rohan Mahy [mailto:rohan@ekabal.com]=20
Sent: Monday, November 08, 2004 4:30 PM
To: Christian Jansson
Cc: Rohan Mahy; sip@ietf.org
Subject: Re: [Sip] Join header (draft-ietf-sip-join-03.txt) unnecessary? =
Second attempt.

Hi,=20

Sorry for the delayed response.=20

Based on our experiences with Replaces, Dan Petrie, Alan Johnston, =
Robert Sparks, myself and other members of the working group where able =
to point out significant problems with reusing dialog identifiers for =
multiple dialogs. The simplest example is that a Replaces header sent to =
a locally mixed conference might match multiple dialogs. It is then =
impossible to replace a single (specific) dialog. Also Join allows a =
user agent to redirect a Join request to a conference in a much more =
natural way. Hope this helps. In any case, Join is now RFC3911.=20

thanks,=20
-rohan=20

On Nov 8, 2004, at 9:45, Christian Jansson wrote:=20

I got no answers on this question last time so I try again.=20

=A0=20

Is the approach described below so naive that no one even has the energy =
point out its weaknesses? Maybe it has already been discussed and deemed =
inadequate?=20

=A0=20

/ Christian=20

=A0=20

=A0=20

From: Christian Jansson=20
Sent: den 21 oktober 2004 16:05=20
To: sip@ietf.org=20
Subject: [Sip] Join header (draft-ietf-sip-join-03.txt) unnecessary? =
[html]=20

=A0=20

Why do we need a Join header to be able to indicate that we want to join =
an ongoing session? Wouldn't it be enough to, for the party that wants =
to join, actually use the call-id for the session when sending the =
join-INVITE? Suppose A and B are in a session with the dialog identified =
by call-id=3D12345;tag=3DA;tag=3DB. If C now wants to join the session =
he sends an INVITE to A or B with call-id=3D12345, from-tag=3DC, and an =
empty to-tag. Wouldn't it make more sense that the 2 dialog-ids had some =
parts in common, that is the call-id, and something that distinguished =
them, that is the to and from tags? It would be much easier to find the =
session related messages as they would share the same call-id.=20

=A0=20

Have I missed some essential part of the Join header, or are we just =
adding something that we could already express easily with what we =
already have?=20

=A0=20

=A0=20

/ Christian Jansson, Hotsip=20


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To: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] mib-08
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hi cullen,

we had a relatively small number of outstanding issues to
still resolve and were trying to get a -09 rev out a couple
of weeks back.  we didn't make that target because while working
the outstanding issues, we also tried to accommodate a request
to enhance user agent information.

i don't, think we should try and include that in the initial
offering of the user agent mib module.  i think jean-francois
is possibly in agreement with that now.  he was going to take
a crack at the issue but didn't have time yet.  thus we haven't
published rev -09 but hope to soon.  i don't have any time to
devote to the new issue of additional user agent information.
so i'd like to leave that for the future.

the delta's between -08 and -09 shouldn't be vast.... relative
to the deltas between -07 and -08 ;)   i think the "bunch of people"
you refer to should be the wglc reviewers.  i don't mind others
also reviewing, but i don't want this to become yet another wglc on
this draft (we've had two in as many years!).  we need to move the
I-D forward so we can get a RFC that can get some implementation
experience.

kevin

Cullen Jennings wrote:
> I tired to read this whole thing again and it made my head want to explode -
> I think getting a bunch of people to read it again would be really good.
> There has been some nice work done on it since the previous rev.
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 



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From sip-bounces@ietf.org  Mon Nov  8 13:06:10 2004
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Subject: RE: [Sip] Strict,
	Semi-Strict and Loose mode in RPH - not a  good fit for ets and wps
To: "James M. Polk" <jmpolk@cisco.com>
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James said
---
> > 2 SIP element ignores a namespace it does not recognize (loose)

To this, remember the ID does not say to "fail" the call, just that
Request. This is one way to inform the calling UAC to know it included a
namespace or priority value that was not recognizable (meaning not
ets.0/1/2/3/4 or wps.0/1/2/3/4).

There is no way to inform the UAC of this issue within the SIP Request
(with the bad namespace.priority-value) without failing the request.
Meaning, there can be no preferential treatment without the sending element

being told so. SIP doesn't have a "pass, but this wasn't a good choice"
provisional response like other protocols.

To the user, they might not notice several SIP Request failures, the person

is merely waiting for the call to be set up (still under a second).
---
OK, please educate me.

If it "fails" that request, but doesn't "fail" the call, does that mean
that the UAC resends the request without the RPH?

If that is what it means, then, for a call which originates in a SIP/IP
domain, but goes to the PSTN, it would reach the PSTN gateway with no
indication that it was entitled to special treatment in the PSTN.

While that is better than failing the call, it still isn't the desired
behavior.

If there is a way that the SIP element can "fail the request but not the
call", AND make sure that the appropriate RPH namespace and value (and
authentication)  get to the PSTN gateway, then it might be OK.

Janet


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From sip-bounces@ietf.org  Mon Nov  8 13:36:14 2004
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Subject: RE: [Sip] mib-08
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From: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
To: "Kevin Lingle" <klingle@cisco.com>, "Cullen Jennings" <fluffy@cisco.com>
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I agree with Kevin. At this point in time we should not change the scope =
of the SIP MIB work, but rather focus on completing this phase of the =
work, in the framework already agreed. This does not mean that more eyes =
to read and comment on the MIB would do harm, quite the contrary - but =
changes should focus on fixing problems, not introducing new items.=20

After this phase is completed, the WG may be open to discussing possible =
extensions which can be introduced in supplementary SIP MIB modules.=20

Regards,

Dan



> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On=20
> Behalf Of Kevin Lingle
> Sent: 08 November, 2004 7:26 PM
> To: Cullen Jennings
> Cc: sip@ietf.org
> Subject: Re: [Sip] mib-08
>=20
>=20
> hi cullen,
>=20
> we had a relatively small number of outstanding issues to
> still resolve and were trying to get a -09 rev out a couple
> of weeks back.  we didn't make that target because while working
> the outstanding issues, we also tried to accommodate a request
> to enhance user agent information.
>=20
> i don't, think we should try and include that in the initial
> offering of the user agent mib module.  i think jean-francois
> is possibly in agreement with that now.  he was going to take
> a crack at the issue but didn't have time yet.  thus we haven't
> published rev -09 but hope to soon.  i don't have any time to
> devote to the new issue of additional user agent information.
> so i'd like to leave that for the future.
>=20
> the delta's between -08 and -09 shouldn't be vast.... relative
> to the deltas between -07 and -08 ;)   i think the "bunch of people"
> you refer to should be the wglc reviewers.  i don't mind others
> also reviewing, but i don't want this to become yet another wglc on
> this draft (we've had two in as many years!).  we need to move the
> I-D forward so we can get a RFC that can get some implementation
> experience.
>=20
> kevin
>=20
> Cullen Jennings wrote:
> > I tired to read this whole thing again and it made my head=20
> want to explode -
> > I think getting a bunch of people to read it again would be=20
> really good.
> > There has been some nice work done on it since the previous rev.
> >=20
> >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From: Aki Niemi <aki.niemi@nokia.com>
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Cc: jon.peterson@neustar.biz
Subject: [Sip] Comments on sip-identity-03
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Hi,

I read the draft and it looks reasonable. One comment though: in section 
6 there is a recommended policy which instructs matching the username 
asserted in the Digest authentication to the From header field.

I think this needs clarification. Someone might read it to mean the 
contents of the username param is being matched, which I'm assuming is 
not the intention. Rather, it should say that the account's URI for 
which the username/passwd is for is matched against the URI in the From.

In addition, the section contains text about aliases and matching those 
usernames, where this passage was quite hard to parse:

    Accordingly, provided
    the authentication service is aware of the relationships between
    these accounts, it might allow a user providing credentials for one
    account to assert a username associated with another account
    controlled by the user name.

I think I got the idea, but rephrasing it would be in order.

Cheers,
Aki

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Subject: RE: [Sip] Strict,
	Semi-Strict and Loose mode in RPH - not a  good fit for ets and wps
To: "James M. Polk" <jmpolk@cisco.com>
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James said:
---


> > 4 SIP messages that include the RPH with the ets or wps namespaces
should
> > include the authorization header (or equivalent mechanism)
(semi-strict)

this should be self evident in public networks....
---

Agree- my point was that this is part of "semi-strict" that IS desirable,
even if we want the treatment to be "loose" in terms of  treatment of an
unrecognized namespace.

Consider this  policy for a specific namespace
 - If the SIP element doesn't recognize the namespace , it ignores the
header (doesn't give any special treatment here, but passes it on if
appropriate- e.g., to a signalling gateway)
-  If the SIP element recognizes the namespace, but there is an invalid
value, the request (and the call) are rejected
-  If the SIP element recognizes the namespace, and the value, but the
authentication header doesn't pass muster, the request (and the call) are
rejected
-  If the SIP element recognizes the namespace, and the value, AND
authentication header is "good", then the request (and the call) get
special treatment here, and the RPH is passed on if appropriate- e.g., to a
signalling gateway.

Is this policy permitted by the current draft?  My reading is that this
policy is not permitted.

Section 4.3.2 says:
"When handling requests with unknown namespaces or priority values,
   elements can operate in one of three modes, "strict", "semi-strict"
   and "loose"."
and
       "If the request includes a 'Require' header field with the
   'Resource-Priority' option tag, a UAS MUST follow the strict or
   semi-strict mode rules, otherwise UAS and proxies MUST operate in
   loose mode."

Since the policy described above does not fit "strict", "semi-strict" or
"loose", it would seem to be forbidden.

Am I missing something?

Janet
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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Join header (draft-ietf-sip-join-03.txt) unnecessary?
	Second attempt.
Date: Mon, 8 Nov 2004 13:52:32 -0500
To: "Christian Jansson" <christian.jansson@hotsip.com>
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On Nov 8, 2004, at 11:33, Christian Jansson wrote:

> I saw that the draft had made it to RFC so in some way I am too late,=20=

> but I still do not see the advantages with Join.

At this point, we have a standard track document that was reviewed by=20
the WG, expresses a consensus approach, technically works, and has a=20
handful of implementations.  I don't think there is much value=20
rehashing this issue.

> "The simplest example is that a Replaces header sent to a locally=20
> mixed conference might match multiple dialogs"
> Is that an advantage or disadvantage of Join vs the model I proposed?
>
> "It is then impossible to replace a single (specific) dialog"
> But wouldn't that be possible with the proposal I wrote also? I think=20=

> so. Do you have any reference to mailing-list discussions where you=20
> realized that what was in RFC3261 was not enough?
>
> I thought it would be good to reuse the elements that are in 3261. In=20=

> a way dialogs could be seen to be coupled to a call-id in a heretical=20=

> view.
>
> In the case below UA:A has a call containing 2 dialogs with dialog-ids=20=

> (1234:A1:BB) and (1234:A2:CC). If one would like to use Replaces to=20
> replace one of the dialogs then simply target one of them using the=20
> dialog id. To me Join seems to invent the wheel again.

You can't use the fact that the call-id is the same to assume that two=20=

calls are part of the same conversation.  For example, when forking an=20=

INVITE for video or text, it is quite reasonable to get multiple 200s=20
(and multiple independent sessions) which stay up.  These are most=20
likely not part of the same conversation.

thanks,
-rohan


>      UA:A
>
>     Call-id: 1234
>      |          |
>  local-tag=3DA1  local-tag=3DA2
>  remote-tag=3DBB remote-tag=3DCC
>
> In this model it is very easy to see that the two dialogs are=20
> connected, as they both share the call-id part of their dialog-id. I=20=

> think that is a very attractive feature.
>
> Now if we use Join instead do the same thing, we would get some state=20=

> above this (conference?) that relates one dialog with another dialog=20=

> with a completely different set of identifiers. It also makes the=20
> call-id identifier more or less useless except for the case when=20
> forking gives back multiple answers, which probably won't happen under=20=

> normal operation. Does this state above dialogs and call-ids have a=20
> name defined anywhere?
>
>
> / Christian Jansson
>
>
> ________________________________________
> From: Rohan Mahy [mailto:rohan@ekabal.com]
> Sent: Monday, November 08, 2004 4:30 PM
> To: Christian Jansson
> Cc: Rohan Mahy; sip@ietf.org
> Subject: Re: [Sip] Join header (draft-ietf-sip-join-03.txt)=20
> unnecessary? Second attempt.
>
> Hi,
>
> Sorry for the delayed response.
>
> Based on our experiences with Replaces, Dan Petrie, Alan Johnston,=20
> Robert Sparks, myself and other members of the working group where=20
> able to point out significant problems with reusing dialog identifiers=20=

> for multiple dialogs. The simplest example is that a Replaces header=20=

> sent to a locally mixed conference might match multiple dialogs. It is=20=

> then impossible to replace a single (specific) dialog. Also Join=20
> allows a user agent to redirect a Join request to a conference in a=20
> much more natural way. Hope this helps. In any case, Join is now=20
> RFC3911.
>
> thanks,
> -rohan
>
> On Nov 8, 2004, at 9:45, Christian Jansson wrote:
>
> I got no answers on this question last time so I try again.
>
> =A0
>
> Is the approach described below so naive that no one even has the=20
> energy point out its weaknesses? Maybe it has already been discussed=20=

> and deemed inadequate?
>
> =A0
>
> / Christian
>
> =A0
>
> =A0
>
> From: Christian Jansson
> Sent: den 21 oktober 2004 16:05
> To: sip@ietf.org
> Subject: [Sip] Join header (draft-ietf-sip-join-03.txt) unnecessary?=20=

> [html]
>
> =A0
>
> Why do we need a Join header to be able to indicate that we want to=20
> join an ongoing session? Wouldn't it be enough to, for the party that=20=

> wants to join, actually use the call-id for the session when sending=20=

> the join-INVITE? Suppose A and B are in a session with the dialog=20
> identified by call-id=3D12345;tag=3DA;tag=3DB. If C now wants to join =
the=20
> session he sends an INVITE to A or B with call-id=3D12345, from-tag=3DC,=
=20
> and an empty to-tag. Wouldn't it make more sense that the 2 dialog-ids=20=

> had some parts in common, that is the call-id, and something that=20
> distinguished them, that is the to and from tags? It would be much=20
> easier to find the session related messages as they would share the=20
> same call-id.
>
> =A0
>
> Have I missed some essential part of the Join header, or are we just=20=

> adding something that we could already express easily with what we=20
> already have?
>
> =A0
>
> =A0
>
> / Christian Jansson, Hotsip


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Subject: RE: [Sip] Strict,
	Semi-Strict and Loose mode in RPH - not a  good fit for ets and wps
To: "James M. Polk" <jmpolk@cisco.com>
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James said:
___

> > 5 Precedence (queueing) is generally the means of providing priority,
but
> > exemption from network management controls, for instance call admission
> > within an IP domain, when a "regular" call would not be admitted, is
also
> > needed.  Pre-emption is not used.  (semi-strict "plus more")

I do not understand what you have written above, can you rephrase please?
___

OK. I'll try again (and I combined two different cases, which didn't help
the clarity).

Section 7 says
      "The ets and wps namespaces operate with preferential treatment but
   without preemption.  These namespaces operate under a policy of
   provide preferential treatment to higher relative priority requests
   instead of processing lower relative priority requests.  Messages
   are not preempted, or deleted, except under extreme loads in which
   all available processing is taken up with higher priority messages.

   An example of this is at a IP/PSTN gateway with all of the PSTN side
   circuits utilized.  In strict mode one of the lower priority
   circuits would be freed using preemption.  In semi-strict mode, the
   SIP request May be granted access to the next available circuit
   based on this header's presence in the authorized message,
   regardless of how many other "regular" requests are received at that
   gateway."

First, is that "May" supposed to be "MAY" or "may"?

But more significantly, the desired behavior at the PSTN gateway is more
complicated than what you describe;

-At an IP/PSTN gateway with all of the PSTN side circuits utilized, not
only would the request with the appropriate RPH be granted "the next
available circuit" but if several requests with the appropriate RPH arrive,
they would be queued for the next several circuits to become available.

- At an IP/PSTN gateway, "restrictive network management controls" may be
active, even if  not all PSTN side circuits are utilized.  (An example of
a "restrictive network management control" is "call gapping" in which only
one call every x seconds is accepted.  "Call gapping" is usually made
active as a response to overall network congestion, a bit like ECN in an IP
network).  If restrictive network management controls are active, a request
with the appropriate RPH would be granted an available circuit, even if
less than x seconds had passed since the last call was admitted.

- At an IP/PSTN gateway, if the request has the appropriate RPH, certain
parameters (CPC and/or Precedence) would be set in the ISUP IAM.

I don't see anything in the RPH ID that prevents this "additional"
functionality, so this comment doesn't necessarily mean anything needs to
change.  It was just that the current text is a little misleading in that
it seems to imply that "access to the next available circuit" is all that
is intended.

Janet

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Subject: RE: [Sip] Strict,
	Semi-Strict and Loose mode in RPH - not a  good fit for ets and wps
To: "James M. Polk" <jmpolk@cisco.com>
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James said:
___


> > 6 There is no a priori requirement to use, or not use, the "Require"
> > header field.
>
>isn't this addressed in section 4.3.2 below?  or am I missing something?
>
>    If the request includes a 'Require' header field with the
>    'Resource-Priority' option tag, a UAS MUST follow the strict or
>    semi-strict mode rules, otherwise UAS and proxies MUST operate in
>    loose mode.
>
>
>[JG]I guess I wasn't clear.   I simply meant that there wasn't a concern
>about the "require" header itself, independent of the
>strict/semi-strict/loose mode.  Once that is "right", the approach to the
>"require" header will follow

hmmmm

The "Require" header is to mandate that SIP Responses use the header as is
(copied from request to response without change). If ever there is a case
needed in which preferential treatment is within the processing of SIP
elements (such as Proxies), knowing there will always be more than one SIP
message necessary to complete a transaction - meaning both Request and
Response will need to be RP marked for preferential treatment for the
session/transaction to be given priority over other transactions.

To put this simply, if the INVITE is consistently moved to the head of the
line in all SIP elements between UAs 1 & 2, and the 200 OK is
slowed/delayed or lost in the congestion of one or more Proxies, the call
will not be established. In this case, the INVITE, 200 OK and ACK will need
the RP indication to gain the preferential treatment of that domain; and
any other messages within that transaction will require it too (such as
183, PRACK and UPDATE messages if RFC 3312 Preconditions were to be
required to set up, say RSVP between endpoints).
___

OK, I was definitely missing the point here.  I was looking at it simply as
the "require" header being the trigger for "strict, or semi-strict" mode
(as opposed to loose mode).

Now that you have explained that it ALSO triggers "setting RPH in the
response, and other associated messages", I would have to say that I DO
want the require header.

In that case , I am unhappy with
      "'Strict' and 'semi-strict' are identified by the
   presence or absence of a 'Require' header field with the 'Resource-
   Priority' option tag.  'Loose' mode does not contain a Require
   header."

Is there some reason we can't have "Loose" and"Require"?

Janet


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To: Janet P Gunn <jgunn6@csc.com>
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Subject: RE: [Sip] Strict, Semi-Strict and Loose mode in RPH - not a 
	good fit for ets and wps
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comments below

At 12:51 PM 11/8/2004 -0500, Janet P Gunn wrote:
>James said
>---
> > > 2 SIP element ignores a namespace it does not recognize (loose)
>
>To this, remember the ID does not say to "fail" the call, just that
>Request. This is one way to inform the calling UAC to know it included a
>namespace or priority value that was not recognizable (meaning not
>ets.0/1/2/3/4 or wps.0/1/2/3/4).
>
>There is no way to inform the UAC of this issue within the SIP Request
>(with the bad namespace.priority-value) without failing the request.
>Meaning, there can be no preferential treatment without the sending element
>
>being told so. SIP doesn't have a "pass, but this wasn't a good choice"
>provisional response like other protocols.
>
>To the user, they might not notice several SIP Request failures, the person
>
>is merely waiting for the call to be set up (still under a second).
>---
>OK, please educate me.
>
>If it "fails" that request, but doesn't "fail" the call, does that mean
>that the UAC resends the request without the RPH?

yep


>If that is what it means, then, for a call which originates in a SIP/IP
>domain, but goes to the PSTN, it would reach the PSTN gateway with no
>indication that it was entitled to special treatment in the PSTN.
>
>While that is better than failing the call, it still isn't the desired
>behavior.

SIP has no mechanism other than failing the request to indicate something 
was wrong. SIP cannot at this time, allow a message to be processed by a 
SIP element (say a proxy) and have that proxy server send the packet 
towards the UAS *and* send some form of a provisional reply to the UAC 
telling it to correct some aspect of itself in future such messages.


>If there is a way that the SIP element can "fail the request but not the
>call", AND make sure that the appropriate RPH namespace and value (and
>authentication)  get to the PSTN gateway, then it might be OK.

A proxy can replace the field in a header, in this case to a more 
appropriate namespace.value - but I do not believe you will want this to 
happen without authentication of the user at the UAC (which would be in the 
form of a 4XX failure - which you are objecting to.... hmmmm)


>Janet
>
>
>----------------------------------------------------------------------------------------
>
>This is a PRIVATE message. If you are not the intended recipient, please
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cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Subject: RE: [Sip] S-S-L mode in RPH for ets and wps (#5)
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comments in-line

I shortened the subject line BTW

At 02:13 PM 11/8/2004 -0500, Janet P Gunn wrote:
>James said:
>___
>
> > > 5 Precedence (queueing) is generally the means of providing priority,
>but
> > > exemption from network management controls, for instance call admission
> > > within an IP domain, when a "regular" call would not be admitted, is
>also
> > > needed.  Pre-emption is not used.  (semi-strict "plus more")
>
>I do not understand what you have written above, can you rephrase please?
>___
>
>OK. I'll try again (and I combined two different cases, which didn't help
>the clarity).
>
>Section 7 says
>       "The ets and wps namespaces operate with preferential treatment but
>    without preemption.  These namespaces operate under a policy of
>    provide preferential treatment to higher relative priority requests
>    instead of processing lower relative priority requests.  Messages
>    are not preempted, or deleted, except under extreme loads in which
>    all available processing is taken up with higher priority messages.
>
>    An example of this is at a IP/PSTN gateway with all of the PSTN side
>    circuits utilized.  In strict mode one of the lower priority
>    circuits would be freed using preemption.  In semi-strict mode, the
>    SIP request May be granted access to the next available circuit
>    based on this header's presence in the authorized message,
>    regardless of how many other "regular" requests are received at that
>    gateway."
>
>First, is that "May" supposed to be "MAY" or "may"?

MAY

I mistyped it

Now, you could contract that this is a MUST within whoever you contract 
with for IP carrier.  This is a big difference that should be pointed out 
here, the difference between market requirements and protocol requirements. 
A doc can say "MAY", but a customer can demand that aspect be a MUST in 
their network. If the doc says MUST (or even SHOULD in most cases), 
equipment vendors really don't have a choice but to code this feature into 
their products, even if 1 or more customers do not use it. This is a 
marketing decision.


>But more significantly, the desired behavior at the PSTN gateway is more
>complicated than what you describe;

I could write a book about exactly how many behaviors are to be.... but I 
may have oversimplified things here.


>-At an IP/PSTN gateway with all of the PSTN side circuits utilized, not
>only would the request with the appropriate RPH be granted "the next
>available circuit" but if several requests with the appropriate RPH arrive,
>they would be queued for the next several circuits to become available.

true, I didn't get into the multiple request scenario - I could add this 
here, but do not want to account for every permutation for each behavior 
every time. All documents would be very long in this philosophy.


>- At an IP/PSTN gateway, "restrictive network management controls" may be
>active, even if  not all PSTN side circuits are utilized.  (An example of
>a "restrictive network management control" is "call gapping" in which only
>one call every x seconds is accepted.  "Call gapping" is usually made
>active as a response to overall network congestion, a bit like ECN in an IP
>network).  If restrictive network management controls are active, a request
>with the appropriate RPH would be granted an available circuit, even if
>less than x seconds had passed since the last call was admitted.

I will restate to "all available circuits are utilized" to account for 
this. but do not want to get into restrictive mgmt control.


>- At an IP/PSTN gateway, if the request has the appropriate RPH, certain
>parameters (CPC and/or Precedence) would be set in the ISUP IAM.

this isn't a SIP issue, this is a translation issue that the GW vendor is 
to deal with.


>I don't see anything in the RPH ID that prevents this "additional"
>functionality, so this comment doesn't necessarily mean anything needs to
>change.  It was just that the current text is a little misleading in that
>it seems to imply that "access to the next available circuit" is all that
>is intended.

I think I cleaned up the text appropriately to your above points


>Janet
>
>----------------------------------------------------------------------------------------
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>----------------------------------------------------------------------------------------


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Subject: RE: [Sip] Privacy statements and History (draft-ietf-sip-history-
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My initial response is awaiting approval by moderator due to the size, =
so
I've snipped and am resending. =20
=20
Mary=20

-----Original Message-----
From: Barnes, Mary [NGC:B601:EXCH]=20
Sent: Monday, November 08, 2004 1:43 PM
To: 'GARCIN Sebastien RD-CORE-ISS'; sip@ietf.org
Subject: RE: [Sip] Privacy statements and History
(draft-ietf-sip-history-info-04.txt)


S=E9bastien,
=20
Thanks for reviewing the draft; my response to your comments is =
embedded
below [MB].
=20
Mary=20


-----Original Message-----
From: GARCIN Sebastien RD-CORE-ISS
[mailto:sebastien.garcin@francetelecom.com]=20
Sent: Monday, November 08, 2004 10:11 AM
To: Barnes, Mary [NGC:B601:EXCH]; sip@ietf.org
Subject: [Sip] Privacy statements and History
(draft-ietf-sip-history-info-04.txt)


Hi mary, all
=20
When reading draft-ietf-sip-history-info-04.txt, I have trouble in
understanding some of the statements which relate to the forwarding =
rules
for history-entries subject to privacy. It is an important requirement =
that
History-entrie(s) with a Privacy=3Dhistory, session, or header are =
indeed
forwarded to entities which belong to the same trust domain. The =
removal of
specific history-entries should only occur if the peer does not belong =
to
the trust domain.
=20
In the current text (section 4.3.3.1.1) :
If a request is  being forwarded to a Request URI associated with a =
domain
for which the proxy is not responsible and there is a Privacy header in =
the
request with a priv-value of "session", "header" or "history", the =
proxy
MUST remove any hi-entry(s) prior to forwarding.=20
=20
The current wording is misleading since it gives the impression (maybe
intentionnal) that it is not possible to forward history-entries with
Privacy statements to domains under the responsability of e.g. another
operator belonging to the same trust domain. =20
=20
[MB]: Current wording is consistent with terminology in RFC 3261 in =
terms of
describing who is able to change the Request URI in a specific request
(based on section 16.5):=20
   " A proxy MUST NOT add additional targets to the target set if the
   Request-URI of the original request does not indicate a resource =
this
   proxy is responsible for.

      A proxy can only change the Request-URI of a request during
      forwarding if it is responsible for that URI. "
Since History-Info (and associated privacy) are only added to the =
request,
when an entity that is allowed to change the Request-URI retargets the
request, it seemed sensible to use consistent wording to explain that.  =
=20
[/MB]
=20
The concept of "trust domain" should be used when discussing the =
forwarding
rules pertaining to information subject to privacy. Furthermore, the
requirement for forwarding history-entries to trusted entities should =
be
stated more clearly in the draft.=20

[MB]: The whole concept of what defines privacy in terms of the proxy's =
use
of the privacy header is outside the scope of History-Info =
functionality and
really a matter of local policy.   I think the functionality that you =
want
is a matter of local implementation and policy in terms of operators
establishing this "trust domain" model to which you refer.  =
History-Info
defines the mechanism to ensure the privacy of the requests, but it =
doesn't
explicitly define how the proxy knows whether it is responsible for =
that
resource.    I don't think this is a matter of standardization.  I =
thought
the use of the term "domain" rather than "resouce" would be helpful, =
but
perhaps changing it to the more general "resource" would resolve this
concern and/or a statement clarifying what I've just described should =
be
added in the draft.=20
[/MB]
=20
=20
Thank you for clarifying this point.
=20
Best regards,
s=E9bastien
=20
=20

------Remainder of non-related part of this thread has been deleted by
Mary------------------


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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1476" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D761145520-08112004>My=20
initial response is awaiting approval by moderator due to the size, so =
I've=20
snipped and am resending.&nbsp; </SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN lang=3Den-us><FONT face=3DArial color=3D#0000ff =
size=3D2>Mary</FONT></SPAN>=20
</DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
Barnes, Mary=20
  [NGC:B601:EXCH] <BR><B>Sent:</B> Monday, November 08, 2004 1:43=20
  PM<BR><B>To:</B> 'GARCIN Sebastien RD-CORE-ISS';=20
  sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Privacy statements and =
History=20
  (draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D163370919-08112004><SPAN=20
  class=3D019105313-08112004><FONT face=3DArial color=3D#800080=20
  size=3D2>S=E9bastien,</FONT></SPAN></SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
  class=3D163370919-08112004></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
  class=3D163370919-08112004>Thanks for reviewing the draft; my =
response to your=20
  comments is embedded below [MB].</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#800080 size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT color=3D#800080><SPAN lang=3Den-us><FONT face=3DArial=20
  size=3D2>Mary</FONT></SPAN> </FONT></DIV>
  <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
    <DIV><FONT color=3D#0000ff></FONT></DIV>
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
    face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
GARCIN=20
    Sebastien RD-CORE-ISS [mailto:sebastien.garcin@francetelecom.com]=20
    <BR><B>Sent:</B> Monday, November 08, 2004 10:11 AM<BR><B>To:</B> =
Barnes,=20
    Mary [NGC:B601:EXCH]; sip@ietf.org<BR><B>Subject:</B> [Sip] Privacy =

    statements and History=20
    (draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
    <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
    class=3D019105313-08112004>Hi mary, all</SPAN></FONT></DIV>
    <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
    class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
    <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
    class=3D019105313-08112004>When=20
    reading&nbsp;draft-ietf-sip-history-info-04.txt, I&nbsp;have =
trouble in=20
    understanding some of the statements which relate to the forwarding =

    rules&nbsp;for history-entries subject to =
privacy.&nbsp;</SPAN></FONT><FONT=20
    face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D019105313-08112004>It is=20
    an&nbsp;important requirement&nbsp;that&nbsp;History-entrie(s) =
with&nbsp;a=20
    Privacy=3Dhistory, session, or&nbsp;header are indeed =
forwarded&nbsp;to=20
    entities which belong&nbsp;to&nbsp;the same&nbsp;trust domain. The =
removal=20
    of specific history-entries should only occur&nbsp;if the peer does =
not=20
    belong to the trust domain.</SPAN></FONT></DIV>
    <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
    class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
    <DIV dir=3Dltr align=3Dleft><FONT face=3DArial><SPAN=20
    class=3D019105313-08112004><FONT size=3D2><FONT color=3D#0000ff>In =
the current=20
    text (section 4.3.3.1.1)&nbsp;:</FONT></FONT></SPAN></FONT></DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><FONT =
face=3DArial=20
    color=3D#0000ff size=3D2></FONT>If a request is&nbsp; being =
forwarded to a 
    Request URI associated with <STRONG>a domain for which&nbsp;the =
proxy is not=20
    responsible</STRONG> and there is a Privacy header in=20
    the&nbsp;request&nbsp;<SPAN class=3D019105313-08112004>w</SPAN>ith =
a=20
    priv-value of "session", "header" or "history", the&nbsp;proxy MUST =
remove=20
    any hi-entry(s) prior to forwarding. </DIV>
    <DIV><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>&nbsp;</DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
    color=3D#0000ff><FONT size=3D2>The&nbsp;current&nbsp;wording is =
misleading since=20
    it gives the impression (maybe intentionnal) that it is not =
possible to=20
    forward history-entries with Privacy statements to domains under =
the=20
    responsability of&nbsp;e.g. another operator belonging to the same =
trust=20
    domain.&nbsp;<SPAN=20
    =
class=3D163370919-08112004>&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DI=
V>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
    color=3D#0000ff><FONT size=3D2><SPAN=20
    =
class=3D163370919-08112004></SPAN></FONT></FONT></FONT></SPAN>&nbsp;</DI=
V>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
size=3D+0><FONT=20
    color=3D#800080 size=3D2><SPAN class=3D163370919-08112004>[MB]: =
Current wording is=20
    consistent with terminology in RFC 3261 in terms of describing who =
is able=20
    to&nbsp;change the Request URI in a specific request (based =
on&nbsp;section=20
    16.5):&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
    color=3D#800080><SPAN class=3D163370919-08112004>&nbsp;&nbsp; " A =
proxy MUST NOT=20
    add additional targets to the target set if the<BR>&nbsp;&nbsp; =
Request-URI=20
    of the original request does not indicate a resource =
this<BR>&nbsp;&nbsp;=20
    proxy is responsible for.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; A =
proxy can=20
    only change the Request-URI of a request=20
    during<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; forwarding if it is =
responsible for=20
    that URI. "</SPAN></FONT></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
size=3D+0><FONT=20
    color=3D#800080 size=3D2><SPAN class=3D163370919-08112004>Since =
History-Info (and=20
    associated privacy)&nbsp;are only added to the request,&nbsp;when=20
    an&nbsp;entity that is allowed to change&nbsp;the Request-URI =
retargets the=20
    request, it seemed sensible to use consistent wording to explain =
that.&nbsp;=20
    &nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
size=3D+0><FONT=20
    color=3D#800080 size=3D2><SPAN=20
    =
class=3D163370919-08112004>[/MB]</SPAN></FONT></FONT></FONT></SPAN></DIV=
>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#800080=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
    color=3D#0000ff><FONT size=3D2>The&nbsp;concept of "trust =
domain"&nbsp;should=20
    be&nbsp;used when discussing the&nbsp;forwarding rules =
pertaining&nbsp;to=20
    information subject to privacy. Furthermore,&nbsp;the requirement =
for=20
    forwarding history-entries to trusted entities should =
be&nbsp;stated more=20
    clearly in the draft.<SPAN=20
    =
class=3D163370919-08112004>&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DI=
V>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
    color=3D#0000ff><FONT size=3D2><SPAN class=3D163370919-08112004>
    <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
    class=3D163370919-08112004>[MB]: The whole concept of what defines =
privacy in=20
    terms of the proxy's use of the privacy header is outside the scope =
of=20
    History-Info functionality and really a matter of local policy. =
&nbsp; I=20
    think the functionality that you want is a matter of local =
implementation=20
    and policy in terms of&nbsp;operators establishing this "trust =
domain"=20
    model&nbsp;to which you refer.&nbsp; History-Info defines the =
mechanism=20
    to&nbsp;ensure the privacy of the requests, but it =
doesn't&nbsp;explicitly=20
    define how the&nbsp;proxy knows whether it is responsible for that=20
    resource.&nbsp;&nbsp;&nbsp;&nbsp;I don't think this is a matter of=20
    standardization.&nbsp; I thought the use of the term "domain" =
rather than=20
    "resouce" would be helpful, but perhaps changing it to the more =
general=20
    "resource" would resolve this concern and/or a statement clarifying =
what=20
    I've just described should be added in the draft. =
</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
    class=3D163370919-08112004></SPAN></FONT><FONT face=3DArial =
color=3D#800080=20
    size=3D2><SPAN=20
    =
class=3D163370919-08112004>[/MB]</SPAN></FONT></DIV>&nbsp;</SPAN></FONT>=
</FONT></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004></SPAN><SPAN=20
    class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>Thank you&nbsp;for&nbsp;clarifying&nbsp;this=20
    point.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>Best regards,</FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>s=E9bastien</FONT></SPAN></DIV>
    <DIV><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>&nbsp;</DIV>
    <DIV><BR><SPAN class=3D761145520-08112004><FONT face=3DTahoma=20
    size=3D2>------Remainder of non-related part of this thread has =
been deleted=20
    by=20
Mary------------------</FONT></SPAN></DIV></BLOCKQUOTE></BLOCKQUOTE></BO=
DY></HTML>

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From sip-bounces@ietf.org  Mon Nov  8 16:42:23 2004
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Date: Mon, 08 Nov 2004 15:03:13 -0600
To: Janet P Gunn <jgunn6@csc.com>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: RE: [Sip] S-S-L mode in RPH for ets and wps (#6)
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comments in-line

I changed the subject line BTW

At 02:38 PM 11/8/2004 -0500, Janet P Gunn wrote:
>James said:
>___
>
>
> > > 6 There is no a priori requirement to use, or not use, the "Require"
> > > header field.
> >
> >isn't this addressed in section 4.3.2 below?  or am I missing something?
> >
> >    If the request includes a 'Require' header field with the
> >    'Resource-Priority' option tag, a UAS MUST follow the strict or
> >    semi-strict mode rules, otherwise UAS and proxies MUST operate in
> >    loose mode.
> >
> >
> >[JG]I guess I wasn't clear.   I simply meant that there wasn't a concern
> >about the "require" header itself, independent of the
> >strict/semi-strict/loose mode.  Once that is "right", the approach to the
> >"require" header will follow
>
>hmmmm
>
>The "Require" header is to mandate that SIP Responses use the header as is
>(copied from request to response without change). If ever there is a case
>needed in which preferential treatment is within the processing of SIP
>elements (such as Proxies), knowing there will always be more than one SIP
>message necessary to complete a transaction - meaning both Request and
>Response will need to be RP marked for preferential treatment for the
>session/transaction to be given priority over other transactions.
>
>To put this simply, if the INVITE is consistently moved to the head of the
>line in all SIP elements between UAs 1 & 2, and the 200 OK is
>slowed/delayed or lost in the congestion of one or more Proxies, the call
>will not be established. In this case, the INVITE, 200 OK and ACK will need
>the RP indication to gain the preferential treatment of that domain; and
>any other messages within that transaction will require it too (such as
>183, PRACK and UPDATE messages if RFC 3312 Preconditions were to be
>required to set up, say RSVP between endpoints).
>___
>
>OK, I was definitely missing the point here.  I was looking at it simply as
>the "require" header being the trigger for "strict, or semi-strict" mode
>(as opposed to loose mode).

no, this Require header will have "Resource-Priority" as a field value 
stating that the RPH is to be expected in the Response.


>Now that you have explained that it ALSO triggers "setting RPH in the
>response, and other associated messages", I would have to say that I DO
>want the require header.

thanks, it really is needed


>In that case , I am unhappy with
>       "'Strict' and 'semi-strict' are identified by the
>    presence or absence of a 'Require' header field with the 'Resource-
>    Priority' option tag.  'Loose' mode does not contain a Require
>    header."
>
>Is there some reason we can't have "Loose" and"Require"?

no technical reason that I can think of right now


>Janet
>
>
>----------------------------------------------------------------------------------------
>
>This is a PRIVATE message. If you are not the intended recipient, please
>delete without copying and kindly advise us by e-mail of the mistake in
>delivery. NOTE: Regardless of content, this e-mail shall not operate to
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>----------------------------------------------------------------------------------------


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


_______________________________________________
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From sip-bounces@ietf.org  Mon Nov  8 16:55:18 2004
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This is a multi-part message in MIME format.

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Hello Cullen,
=20
I think it is no harm if there is a PING message defined in SIP. I
personally think it is overdue anyway. Implementors can then decide if
they are using it or not. Lets leave that decision to the implementors!
=20
But we have to consider who is sending keep-alive traffic. "IMHO" is it
definitely better if the UA does this, because then e.g. usually it is
ok to send CRLF keep alive or STUN.
=20
We register with a header that incates (a) that the UA is able to do the
refreshing and (b) what the proposed refresh time is. The UA can do all
these tricky NAT measurement tricks and then propose the refresh time.
Stupid UA may just propose a predefined value, e.g. 20 seconds. The
registrar (or whatever is in the path) then can acknowlege the refresh
role and leave the refreshing to the UA. Otherwise, it will send the
refresh messages. In that case, it will probably use PING, cause it
needs a response to keep the NAT alive in all cases - even if it is a
"503 Not Implemented".
=20
Christian
=20

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.2523" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana =
size=3D2>Hello=20
Cullen,</FONT></SPAN></DIV>
<DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana size=3D2>I =
think it is no=20
harm if there is a PING message defined in SIP. I personally think it is =
overdue=20
anyway. Implementors can then decide if they are using it or not. Lets =
leave=20
that decision to the implementors!</FONT></SPAN></DIV>
<DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana size=3D2>But =
we have to=20
consider who is sending keep-alive traffic. "IMHO" is it definitely =
better if=20
the UA does this, because then e.g. usually it is ok to send CRLF keep =
alive or=20
STUN.</FONT></SPAN></DIV>
<DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana size=3D2>We =
register with a=20
header that incates (a) that the UA is able to do the refreshing and (b) =
what=20
the proposed refresh time is. The UA can do all these tricky NAT =
measurement=20
tricks and then propose the refresh time. Stupid UA may just propose a=20
predefined value, e.g. 20 seconds. The registrar (or whatever is in the =
path)=20
then can acknowlege the refresh role and leave the refreshing to the UA. =

Otherwise, it will send the refresh messages. In that case, it will =
probably use=20
PING, cause it needs a response to keep the NAT alive in all cases - =
even if it=20
is a "503 Not Implemented".</FONT></SPAN></DIV>
<DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
size=3D2>Christian</FONT></SPAN></DIV>
<DIV><SPAN class=3D251072321-08112004></SPAN>&nbsp;</DIV></BODY></HTML>

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--===============0308069664==--



From sip-bounces@ietf.org  Mon Nov  8 17:21:27 2004
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Subject: [Sip] Proposed, but not IANA Registered (yet), Response codes
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Is there a central repository for listing Response codes that are in IDs 
(meaning not IANA registered)?

I'm looking for anywhere I can go, if I am willing to risk my life to 
suggest a new response code, and know that the number picked is not taken 
in any other current SIP/SIPPING draft document.

This seems to me to be something on the softarmor site, but I do not see 
anything listing this.

Searching in every SIP and SIPPING doc seems quite inefficient, time 
consuming and prone to error (though not life threatening  ;-)

cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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From sip-bounces@ietf.org  Mon Nov  8 17:24:14 2004
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Aki Niemi'" <aki.niemi@nokia.com>, SIP WG <sip@ietf.org>
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Subject: [Sip] RE: Comments on sip-identity-03
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Thanks Aki. You are correct that we are not suggesting that the Digest
username match as a literal the username in the local-part of the URI in the
>From header. I mean, I guess that's possible, but certainly not necessary.
What we intend is that an authentication service might persist some mapping
between a Digest username and one or more potential local-parts which are
authorized to appear in the From header field value for that user. I will
attempt to free to language you cite below from the shackles of confusion.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Aki Niemi [mailto:aki.niemi@nokia.com]
> Sent: Monday, November 08, 2004 1:32 PM
> To: SIP WG
> Cc: jon.peterson@neustar.biz
> Subject: Comments on sip-identity-03
> 
> 
> Hi,
> 
> I read the draft and it looks reasonable. One comment though: in section 
> 6 there is a recommended policy which instructs matching the username 
> asserted in the Digest authentication to the From header field.
> 
> I think this needs clarification. Someone might read it to mean the 
> contents of the username param is being matched, which I'm assuming is 
> not the intention. Rather, it should say that the account's URI for 
> which the username/passwd is for is matched against the URI 
> in the From.
> 
> In addition, the section contains text about aliases and matching those 
> usernames, where this passage was quite hard to parse:
> 
>     Accordingly, provided
>     the authentication service is aware of the relationships between
>     these accounts, it might allow a user providing credentials for one
>     account to assert a username associated with another account
>     controlled by the user name.
> 
> I think I got the idea, but rephrasing it would be in order.
> 
> Cheers,
> Aki
> 

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From sip-bounces@ietf.org  Mon Nov  8 17:49:33 2004
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From: Alan Hawrylyshen <alan@jasomi.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, sip@ietf.org
Subject: [Sip] GRUU Comments and interaction with outbound-connection
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Re: the Edge Proxy, Reg/Prox RR problems....

Johnathan;

It strikes me that the EP can directly route the GRUU since it is 
(except in the 3GPP case) in the same administrative / security / 
policy domain as the registrar and SHOULD be capable of detecting the 
GRUU's applicability when the initial request hits the EP.  An 
appropriately constructed (encoded) GRUU could be decoded and dealt 
with as soon as it reaches the UA's EP, no?

And as I write this, Cullen has just proposed the same thing.

Alan



a l a n a t j a s o m i d o t c o m


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From sip-bounces@ietf.org  Mon Nov  8 18:20:55 2004
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Subject: [Sip] reinvite and refer methods
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what is exactly the difference between the Re-Invite and Refer methods?

thanks
moe smadi

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From: "m. smadi" <smadi@power.eng.mcmaster.ca>
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Subject: [Sip] SIP reinvite
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hello;

my understanding of a SIP reinvite is that it modifies the media 
session.  Is it capable of modifying the address of the send too?  for 
example of A is talking to B, can A send a reinvite to B with C's 
address instead of its own address?

Moe Smadi

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From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] Proposed, but not IANA Registered (yet), Response codes
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James M. Polk wrote:

> Is there a central repository for listing Response codes that are in 
> IDs (meaning not IANA registered)?
>
None that I know of, but if you want to be the editor, I'll happily host 
the web page.

--
Dean


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To: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] Proposed, but not IANA Registered (yet), Response
  codes
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At 07:26 PM 11/8/2004 -0600, Dean Willis wrote:
>James M. Polk wrote:
>
>>Is there a central repository for listing Response codes that are in IDs 
>>(meaning not IANA registered)?
>None that I know of, but if you want to be the editor, I'll happily host 
>the web page.

well.... I did bring it up


>--
>Dean
>


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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As requested by chairs, I think I understand the issues and I think what
Jonathan proposed sounds like it will work.




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From: Avasarala Ranjit-A20990 <ranjit@motorola.com>
To: "m. smadi" <smadi@power.eng.mcmaster.ca>, sip@ietf.org
Subject: RE: [Sip] SIP reinvite
Date: Tue, 9 Nov 2004 10:20:05 +0530 
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Hi
   For this you can use REFER with Refer-To header. So A can specify C's address in Refer-To header. 

Regards
Ranjit



-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of m. smadi
Sent: Tuesday, November 09, 2004 4:46 AM
To: sip@ietf.org
Subject: [Sip] SIP reinvite


hello;

my understanding of a SIP reinvite is that it modifies the media 
session.  Is it capable of modifying the address of the send too?  for 
example of A is talking to B, can A send a reinvite to B with C's 
address instead of its own address?

Moe Smadi

_______________________________________________
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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] Privacy statements and History
	(draft-ietf-sip-history-info-04.txt)
Date: Tue, 9 Nov 2004 10:52:38 +0100
Message-ID: <49E7012A614B024B80A7D175CB9A64EC86C8F3@ftrdmel1.rd.francetelecom.fr>
Thread-Topic: [Sip] Privacy statements and History
	(draft-ietf-sip-history-info-04.txt)
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From: "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>
To: "Mary Barnes" <mary.barnes@nortelnetworks.com>, <sip@ietf.org>
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Mary,
=20
Responses below marked [SG]
=20
Best regards,=20
s=E9bastien=20

________________________________

De : sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] De la part de =
Mary Barnes
Envoy=E9 : lundi 8 novembre 2004 21:57
=C0 : Mary Barnes; GARCIN Sebastien RD-CORE-ISS; 'sip@ietf.org'
Objet : RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)


My initial response is awaiting approval by moderator due to the size, =
so I've snipped and am resending. =20
=20
Mary=20

	-----Original Message-----
	From: Barnes, Mary [NGC:B601:EXCH]=20
	Sent: Monday, November 08, 2004 1:43 PM
	To: 'GARCIN Sebastien RD-CORE-ISS'; sip@ietf.org
	Subject: RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)
=09
=09
	S=E9bastien,
	=20
	Thanks for reviewing the draft; my response to your comments is =
embedded below [MB].
	=20
	Mary=20

	=09
		-----Original Message-----
		From: GARCIN Sebastien RD-CORE-ISS =
[mailto:sebastien.garcin@francetelecom.com]=20
		Sent: Monday, November 08, 2004 10:11 AM
		To: Barnes, Mary [NGC:B601:EXCH]; sip@ietf.org
		Subject: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)
	=09
	=09
		Hi mary, all
		=20
		When reading draft-ietf-sip-history-info-04.txt, I have trouble in =
understanding some of the statements which relate to the forwarding =
rules for history-entries subject to privacy. It is an important =
requirement that History-entrie(s) with a Privacy=3Dhistory, session, or =
header are indeed forwarded to entities which belong to the same trust =
domain. The removal of specific history-entries should only occur if the =
peer does not belong to the trust domain.
		=20
		In the current text (section 4.3.3.1.1) :
		If a request is  being forwarded to a Request URI associated with a =
domain for which the proxy is not responsible and there is a Privacy =
header in the request with a priv-value of "session", "header" or =
"history", the proxy MUST remove any hi-entry(s) prior to forwarding.=20
		=20
		The current wording is misleading since it gives the impression (maybe =
intentionnal) that it is not possible to forward history-entries with =
Privacy statements to domains under the responsability of e.g. another =
operator belonging to the same trust domain. =20
		=20
		[MB]: Current wording is consistent with terminology in RFC 3261 in =
terms of describing who is able to change the Request URI in a specific =
request (based on section 16.5):=20
		   " A proxy MUST NOT add additional targets to the target set if the
		   Request-URI of the original request does not indicate a resource =
this
		   proxy is responsible for.
	=09
		      A proxy can only change the Request-URI of a request during
		      forwarding if it is responsible for that URI. "
		Since History-Info (and associated privacy) are only added to the =
request, when an entity that is allowed to change the Request-URI =
retargets the request, it seemed sensible to use consistent wording to =
explain that.  =20
		[/MB]=20
		 [SG] I have no problem with the statements above. My concern is that =
if there is a hi-entry already embedded in the request with a Privacy =
statement, then, it should be up to local policy to decide whether or =
not a proxy shall pass on those hi-entries to a trusted domain. The =
sentence in section 4.3.3.1.1 precludes this. I would propose to lighten =
the strenght of the sentence as follows:
		=20
		If a request is  being forwarded to a Request URI associated with a =
domain for which the proxy is not responsible and there is a Privacy =
header in the request with a priv-value of "session", "header" or =
"history", the proxy MAY remove any hi-entry(s) prior to forwarding.=20
		=20
		[SG] Another issue is the decision to add hi-entries in a privacy =
context. A proxy changing the target (i.e. the proxy is responsible of =
the resource reflected in the received Request-URI) of a request which =
contained a "privacy=3Dhistory" header MAY add a history-entries =
provided that it knows it can rely on other entities within the trust =
domain to apply the requested privacy. This affect item 4 in the list on =
conditions of section 4.3.3 and 4.3.3.1.=20
		=20
		The concept of "trust domain" should be used when discussing the =
forwarding rules pertaining to information subject to privacy. =
Furthermore, the requirement for forwarding history-entries to trusted =
entities should be stated more clearly in the draft.=20
	=09
		[MB]: The whole concept of what defines privacy in terms of the =
proxy's use of the privacy header is outside the scope of History-Info =
functionality and really a matter of local policy.   I think the =
functionality that you want is a matter of local implementation and =
policy in terms of operators establishing this "trust domain" model to =
which you refer.  History-Info defines the mechanism to ensure the =
privacy of the requests, but it doesn't explicitly define how the proxy =
knows whether it is responsible for that resource.    I don't think this =
is a matter of standardization.  I thought the use of the term "domain" =
rather than "resouce" would be helpful, but perhaps changing it to the =
more general "resource" would resolve this concern and/or a statement =
clarifying what I've just described should be added in the draft.=20
		[/MB]
		[SG]   Changing the term "domain" to "resource" does not solve the =
problem I mentionned above. In order to reflect current operator =
requirements, the draft should not PRECLUDE the fowarding of existing =
history information towards trusted domains.
		=20
		Thank you for clarifying this point.
		=20
		Best regards,
		s=E9bastien
		=20
		=20

		------Remainder of non-related part of this thread has been deleted by =
Mary------------------


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1476" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D436390008-09112004>Mary,</SPAN></FONT></DIV>
<DIV><SPAN class=3D436390008-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2>&nbsp;</FONT></SPAN></DIV>
<DIV><SPAN class=3D436390008-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2>Responses below marked [SG]</FONT></SPAN></DIV>
<DIV><SPAN class=3D436390008-09112004>&nbsp;</SPAN></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D436390008-09112004>Best r</SPAN><SPAN=20
class=3D436390008-09112004>egards,&nbsp;</SPAN></FONT></FONT></FONT></DIV=
>
<DIV><FONT><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D436390008-09112004></SPAN></FONT></FONT></FONT><SPAN=20
class=3D436390008-09112004></SPAN><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
size=3D2>s<SPAN=20
class=3D436390008-09112004>=E9bastien&nbsp;</SPAN></FONT></FONT></FONT><B=
R></DIV>
<DIV class=3DOutlookMessageHeader lang=3Dfr dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>De&nbsp;:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>De la part de</B> Mary=20
Barnes<BR><B>Envoy=E9&nbsp;:</B> lundi 8 novembre 2004 =
21:57<BR><B>=C0&nbsp;:</B>=20
Mary Barnes; GARCIN Sebastien RD-CORE-ISS; =
'sip@ietf.org'<BR><B>Objet&nbsp;:</B>=20
RE: [Sip] Privacy statements and History=20
(draft-ietf-sip-history-info-04.txt)<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D761145520-08112004>My=20
initial response is awaiting approval by moderator due to the size, so =
I've=20
snipped and am resending.&nbsp; </SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN lang=3Den-us><FONT face=3DArial color=3D#0000ff =
size=3D2>Mary</FONT></SPAN>=20
</DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
Barnes, Mary=20
  [NGC:B601:EXCH] <BR><B>Sent:</B> Monday, November 08, 2004 1:43=20
  PM<BR><B>To:</B> 'GARCIN Sebastien RD-CORE-ISS';=20
  sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Privacy statements and =
History=20
  (draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D163370919-08112004><SPAN=20
  class=3D019105313-08112004><FONT face=3DArial color=3D#800080=20
  size=3D2>S=E9bastien,</FONT></SPAN></SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
  class=3D163370919-08112004></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
  class=3D163370919-08112004>Thanks for reviewing the draft; my response =
to your=20
  comments is embedded below [MB].</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#800080 size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT color=3D#800080><SPAN lang=3Den-us><FONT face=3DArial=20
  size=3D2>Mary</FONT></SPAN> </FONT></DIV>
  <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
    <DIV><FONT color=3D#0000ff></FONT></DIV>
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
    face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
GARCIN=20
    Sebastien RD-CORE-ISS [mailto:sebastien.garcin@francetelecom.com]=20
    <BR><B>Sent:</B> Monday, November 08, 2004 10:11 AM<BR><B>To:</B> =
Barnes,=20
    Mary [NGC:B601:EXCH]; sip@ietf.org<BR><B>Subject:</B> [Sip] Privacy=20
    statements and History=20
    (draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
    <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
    class=3D019105313-08112004>Hi mary, all</SPAN></FONT></DIV>
    <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
    class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
    <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
    class=3D019105313-08112004>When=20
    reading&nbsp;draft-ietf-sip-history-info-04.txt, I&nbsp;have trouble =
in=20
    understanding some of the statements which relate to the forwarding=20
    rules&nbsp;for history-entries subject to =
privacy.&nbsp;</SPAN></FONT><FONT=20
    face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D019105313-08112004>It is=20
    an&nbsp;important requirement&nbsp;that&nbsp;History-entrie(s) =
with&nbsp;a=20
    Privacy=3Dhistory, session, or&nbsp;header are indeed =
forwarded&nbsp;to=20
    entities which belong&nbsp;to&nbsp;the same&nbsp;trust domain. The =
removal=20
    of specific history-entries should only occur&nbsp;if the peer does =
not=20
    belong to the trust domain.</SPAN></FONT></DIV>
    <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
    class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
    <DIV dir=3Dltr align=3Dleft><FONT face=3DArial><SPAN=20
    class=3D019105313-08112004><FONT size=3D2><FONT color=3D#0000ff>In =
the current=20
    text (section 4.3.3.1.1)&nbsp;:</FONT></FONT></SPAN></FONT></DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><FONT =
face=3DArial=20
    color=3D#0000ff size=3D2></FONT>If a request is&nbsp; being =
forwarded to a=20
    Request URI associated with <STRONG>a domain for which&nbsp;the =
proxy is not=20
    responsible</STRONG> and there is a Privacy header in=20
    the&nbsp;request&nbsp;<SPAN class=3D019105313-08112004>w</SPAN>ith a =

    priv-value of "session", "header" or "history", the&nbsp;proxy MUST =
remove=20
    any hi-entry(s) prior to forwarding. </DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
    color=3D#0000ff><FONT size=3D2>The&nbsp;current&nbsp;wording is =
misleading since=20
    it gives the impression (maybe intentionnal) that it is not possible =
to=20
    forward history-entries with Privacy statements to domains under the =

    responsability of&nbsp;e.g. another operator belonging to the same =
trust=20
    domain.&nbsp;<SPAN=20
    =
class=3D163370919-08112004>&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV=
>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
    color=3D#0000ff><FONT size=3D2><SPAN=20
    =
class=3D163370919-08112004></SPAN></FONT></FONT></FONT></SPAN>&nbsp;</DIV=
>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
size=3D+0><FONT=20
    color=3D#800080 size=3D2><SPAN class=3D163370919-08112004>[MB]: =
Current wording is=20
    consistent with terminology in RFC 3261 in terms of describing who =
is able=20
    to&nbsp;change the Request URI in a specific request (based =
on&nbsp;section=20
    16.5):&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
    color=3D#800080><SPAN class=3D163370919-08112004>&nbsp;&nbsp; " A =
proxy MUST NOT=20
    add additional targets to the target set if the<BR>&nbsp;&nbsp; =
Request-URI=20
    of the original request does not indicate a resource =
this<BR>&nbsp;&nbsp;=20
    proxy is responsible for.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; A =
proxy can=20
    only change the Request-URI of a request=20
    during<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; forwarding if it is =
responsible for=20
    that URI. "</SPAN></FONT></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
size=3D+0><FONT=20
    color=3D#800080 size=3D2><SPAN class=3D163370919-08112004>Since =
History-Info (and=20
    associated privacy)&nbsp;are only added to the request,&nbsp;when=20
    an&nbsp;entity that is allowed to change&nbsp;the Request-URI =
retargets the=20
    request, it seemed sensible to use consistent wording to explain =
that.&nbsp;=20
    &nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT color=3D#800080><SPAN=20
    class=3D163370919-08112004><FONT face=3DArial><FONT =
size=3D2>[/MB]<SPAN=20
    class=3D436390008-09112004><FONT=20
    =
color=3D#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></SPAN></FONT></SPAN></=
DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT color=3D#800080><SPAN=20
    class=3D163370919-08112004><FONT face=3DArial><FONT size=3D2><SPAN=20
    =
class=3D436390008-09112004>&nbsp;</SPAN></FONT></FONT></SPAN></FONT></SPA=
N><SPAN=20
    class=3D019105313-08112004><FONT face=3DArial color=3D#800080 =
size=3D2><SPAN=20
    class=3D436390008-09112004><STRONG><FONT color=3D#808080>[SG] I have =
no problem=20
    with the statements above.&nbsp;My concern is that if there is=20
    a&nbsp;hi-entry&nbsp;<U>already</U> embedded in the request with a =
Privacy=20
    statement, then, it should be&nbsp;up to local policy to decide =
whether or=20
    not&nbsp;a proxy shall pass on those hi-entries to a trusted domain. =
The=20
    sentence in section 4.3.3.1.1 precludes this.&nbsp;I would propose =
to=20
    lighten the strenght of the sentence as=20
    follows:</FONT></STRONG></SPAN></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#808080=20
    size=3D2><SPAN=20
    =
class=3D436390008-09112004><STRONG></STRONG></SPAN></FONT></SPAN>&nbsp;</=
DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT><SPAN =
class=3D436390008-09112004>If=20
    a request is&nbsp; being forwarded to a Request URI associated with=20
    <STRONG>a domain for which&nbsp;the proxy is not =
responsible</STRONG> and=20
    there is a Privacy header in the&nbsp;request&nbsp;<SPAN=20
    class=3D019105313-08112004>w</SPAN>ith a priv-value of "session", =
"header" or=20
    "history", the&nbsp;proxy&nbsp;MAY remove any hi-entry(s) prior to=20
    forwarding. </SPAN></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#808080=20
    size=3D2><SPAN class=3D436390008-09112004></SPAN></FONT></SPAN><SPAN =

    class=3D019105313-08112004><FONT face=3DArial color=3D#808080 =
size=3D2><SPAN=20
    =
class=3D436390008-09112004><STRONG></STRONG></SPAN></FONT></SPAN>&nbsp;</=
DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#808080=20
    size=3D2><SPAN class=3D436390008-09112004><STRONG>[SG]&nbsp;Another =
issue is the=20
    decision to add hi-entries in a&nbsp;privacy context. A proxy =
changing the=20
    target (i.e. the proxy is responsible of the resource reflected in =
the=20
    received Request-URI) of a request which contained a=20
    "privacy=3Dhistory"&nbsp;header MAY add&nbsp;a =
history-entries&nbsp;provided=20
    that&nbsp;it&nbsp;knows it can rely on other entities within =
the&nbsp;trust=20
    domain to apply the requested privacy. This affect item 4 in the =
list on=20
    conditions of section 4.3.3 and 4.3.3.1.=20
</STRONG></SPAN></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><SPAN=20
    class=3D436390008-09112004>&nbsp;</SPAN></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
    color=3D#0000ff><FONT size=3D2>The&nbsp;concept of "trust =
domain"&nbsp;should=20
    be&nbsp;used when discussing the&nbsp;forwarding rules =
pertaining&nbsp;to=20
    information subject to privacy. Furthermore,&nbsp;the requirement =
for=20
    forwarding history-entries to trusted entities should be&nbsp;stated =
more=20
    clearly in the draft.<SPAN=20
    =
class=3D163370919-08112004>&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV=
>
    <DIV><SPAN class=3D019105313-08112004><FONT color=3D#0000ff><SPAN=20
    class=3D163370919-08112004>
    <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
    class=3D163370919-08112004>[MB]: The whole concept of what defines =
privacy in=20
    terms of the proxy's use of the privacy header is outside the scope =
of=20
    History-Info functionality and really a matter of local policy. =
&nbsp; I=20
    think the functionality that you want is a matter of local =
implementation=20
    and policy in terms of&nbsp;operators establishing this "trust =
domain"=20
    model&nbsp;to which you refer.&nbsp; History-Info defines the =
mechanism=20
    to&nbsp;ensure the privacy of the requests, but it =
doesn't&nbsp;explicitly=20
    define how the&nbsp;proxy knows whether it is responsible for that=20
    resource.&nbsp;&nbsp;&nbsp;&nbsp;I don't think this is a matter of=20
    standardization.&nbsp; I thought the use of the term "domain" rather =
than=20
    "resouce" would be helpful, but perhaps changing it to the more =
general=20
    "resource" would resolve this concern and/or a statement clarifying =
what=20
    I've just described should be added in the draft. =
</SPAN></FONT></DIV>
    <DIV><FONT color=3D#800080><SPAN =
class=3D163370919-08112004></SPAN></FONT><FONT=20
    face=3DArial color=3D#800080 size=3D2><SPAN=20
    class=3D163370919-08112004>[/MB]</SPAN></FONT></DIV><FONT =
face=3DArial><FONT=20
    size=3D2><FONT color=3D#808080><STRONG><SPAN=20
    =
class=3D436390008-09112004>[SG]&nbsp;</SPAN>&nbsp;</STRONG></FONT><SPAN=20
    class=3D436390008-09112004><FONT =
color=3D#808080><STRONG>&nbsp;Changing the=20
    term&nbsp;"domain" to "resource" does not solve the problem I =
mentionned=20
    above. In order to reflect current&nbsp;operator&nbsp;requirements, =
the=20
    draft should not PRECLUDE the fowarding of existing history =
information=20
    towards trusted=20
    =
domains.</STRONG></FONT></SPAN></FONT></FONT></SPAN></FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004></SPAN><SPAN=20
    class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>Thank you&nbsp;for&nbsp;clarifying&nbsp;this=20
    point.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>Best regards,</FONT></SPAN></DIV>
    <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>s=E9bastien</FONT></SPAN></DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
    <DIV><BR><SPAN class=3D761145520-08112004><FONT face=3DTahoma=20
    size=3D2>------Remainder of non-related part of this thread has been =
deleted=20
    by=20
Mary------------------</FONT></SPAN></DIV></BLOCKQUOTE></BLOCKQUOTE></BOD=
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Christian Stredicke'" <Christian.Stredicke@snom.de>, fluffy@cisco.com
Subject: RE: [Sip] PING/PONG
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Hi,
 
I don't have anything PING, but no matter how short and effective (in the number of SIP headers etc) we make it it is still pretty "heavy" on the network.
 
I think we should, as I said yesterday, at least have a look at an option which "combines" CRLF and PING (or whatever method is used), ie a UA can send CRLF every X seconds, and PING every n*X seconds.
 
Of course, using a combined mechanism would mean that it would take longer to detect a NAT failure, in case the NAT binding goes done while the UA sends CRLF. 
 
But, we also have to remember, that eventhough PING would have a response (which CRLC does not have), due to the transcation timeout- and re-transmission timers it would still take a while (until the transcation expires) to figure out that the NAT binding has been closed.
 
Regards,
 
Christer Holmberg
Ericsson Finland
 
 
 

 
 -----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of Christian Stredicke
Sent: 8. marraskuuta 2004 23:31
To: fluffy@cisco.com
Cc: sip@ietf.org
Subject: [Sip] PING/PONG


Hello Cullen,
 
I think it is no harm if there is a PING message defined in SIP. I personally think it is overdue anyway. Implementors can then decide if they are using it or not. Lets leave that decision to the implementors!
 
But we have to consider who is sending keep-alive traffic. "IMHO" is it definitely better if the UA does this, because then e.g. usually it is ok to send CRLF keep alive or STUN.
 
We register with a header that incates (a) that the UA is able to do the refreshing and (b) what the proposed refresh time is. The UA can do all these tricky NAT measurement tricks and then propose the refresh time. Stupid UA may just propose a predefined value, e.g. 20 seconds. The registrar (or whatever is in the path) then can acknowlege the refresh role and leave the refreshing to the UA. Otherwise, it will send the refresh messages. In that case, it will probably use PING, cause it needs a response to keep the NAT alive in all cases - even if it is a "503 Not Implemented".
 
Christian
 


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Content-Type: text/html;
	charset="iso-8859-1"

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
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<META content="MSHTML 6.00.2800.1476" name=GENERATOR></HEAD>
<BODY>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2>Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff size=2>I 
don't have anything PING, but no matter how short and effective (in the number 
of SIP headers etc) we make it it is still pretty "heavy" on the 
network.</FONT></SPAN></DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff size=2>I 
think we should, as I said yesterday, at least have a look at an option which 
"combines" CRLF and PING (or whatever method is used), ie a UA can send CRLF 
every X seconds, and PING every n*X seconds.</FONT></SPAN></DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff size=2>Of 
course, using a combined mechanism would mean that it would take longer to 
detect a NAT failure, in case the NAT binding goes done while the UA sends CRLF. 
</FONT></SPAN></DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff size=2>But, 
we also&nbsp;have to remember, that eventhough PING would have a response (which 
CRLC does not have), due to the transcation timeout- and re-transmission timers 
it would still take a while (until the transcation expires)&nbsp;to figure out 
that the NAT binding has been closed.</FONT></SPAN></DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2>Christer Holmberg</FONT></SPAN></DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2>Ericsson Finland</FONT></SPAN></DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=345572311-09112004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=ltr 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma><FONT 
  size=2><SPAN class=345572311-09112004><FONT face=Arial 
  color=#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></DIV>
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma><FONT 
  size=2><SPAN class=345572311-09112004>&nbsp;</SPAN>-----Original 
  Message-----<BR><B>From:</B> sip-bounces@ietf.org 
  [mailto:sip-bounces@ietf.org]<B>On Behalf Of </B>Christian 
  Stredicke<BR><B>Sent:</B> 8. marraskuuta 2004 23:31<BR><B>To:</B> 
  fluffy@cisco.com<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] 
  PING/PONG<BR><BR></DIV></FONT></FONT>
  <DIV><SPAN class=251072321-08112004><FONT face=Verdana size=2>Hello 
  Cullen,</FONT></SPAN></DIV>
  <DIV><SPAN class=251072321-08112004><FONT face=Verdana 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=251072321-08112004><FONT face=Verdana size=2>I think it is no 
  harm if there is a PING message defined in SIP. I personally think it is 
  overdue anyway. Implementors can then decide if they are using it or not. Lets 
  leave that decision to the implementors!</FONT></SPAN></DIV>
  <DIV><SPAN class=251072321-08112004><FONT face=Verdana 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=251072321-08112004><FONT face=Verdana size=2>But we have to 
  consider who is sending keep-alive traffic. "IMHO" is it definitely better if 
  the UA does this, because then e.g. usually it is ok to send CRLF keep alive 
  or STUN.</FONT></SPAN></DIV>
  <DIV><SPAN class=251072321-08112004><FONT face=Verdana 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=251072321-08112004><FONT face=Verdana size=2>We register with 
  a header that incates (a) that the UA is able to do the refreshing and (b) 
  what the proposed refresh time is. The UA can do all these tricky NAT 
  measurement tricks and then propose the refresh time. Stupid UA may just 
  propose a predefined value, e.g. 20 seconds. The registrar (or whatever is in 
  the path) then can acknowlege the refresh role and leave the refreshing to the 
  UA. Otherwise, it will send the refresh messages. In that case, it will 
  probably use PING, cause it needs a response to keep the NAT alive in all 
  cases - even if it is a "503 Not Implemented".</FONT></SPAN></DIV>
  <DIV><SPAN class=251072321-08112004><FONT face=Verdana 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=251072321-08112004><FONT face=Verdana 
  size=2>Christian</FONT></SPAN></DIV>
  <DIV><SPAN 
class=251072321-08112004></SPAN>&nbsp;</DIV></BLOCKQUOTE></BODY></HTML>

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_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip
--===============1112080699==--



From sip-bounces@ietf.org  Tue Nov  9 07:25:33 2004
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Subject: RE: [Sip] PING/PONG
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Doesn't the OPTIONS message do this job (of keeping a link 'alive')?
=20

--=20
Steve Langstaff.=20

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of =
Christer Holmberg (JO/LMF)
Sent: 09 November 2004 11:37
To: 'Christian Stredicke'; fluffy@cisco.com
Cc: sip@ietf.org
Subject: RE: [Sip] PING/PONG


Hi,
=20
I don't have anything PING, but no matter how short and effective (in =
the number of SIP headers etc) we make it it is still pretty "heavy" on =
the network.
=20
I think we should, as I said yesterday, at least have a look at an =
option which "combines" CRLF and PING (or whatever method is used), ie a =
UA can send CRLF every X seconds, and PING every n*X seconds.
=20
Of course, using a combined mechanism would mean that it would take =
longer to detect a NAT failure, in case the NAT binding goes done while =
the UA sends CRLF.=20
=20
But, we also have to remember, that eventhough PING would have a =
response (which CRLC does not have), due to the transcation timeout- and =
re-transmission timers it would still take a while (until the =
transcation expires) to figure out that the NAT binding has been closed.
=20
Regards,
=20
Christer Holmberg
Ericsson Finland
=20
=20
=20

=20
 -----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of =
Christian Stredicke
Sent: 8. marraskuuta 2004 23:31
To: fluffy@cisco.com
Cc: sip@ietf.org
Subject: [Sip] PING/PONG


Hello Cullen,
=20
I think it is no harm if there is a PING message defined in SIP. I =
personally think it is overdue anyway. Implementors can then decide if =
they are using it or not. Lets leave that decision to the implementors!
=20
But we have to consider who is sending keep-alive traffic. "IMHO" is it =
definitely better if the UA does this, because then e.g. usually it is =
ok to send CRLF keep alive or STUN.
=20
We register with a header that incates (a) that the UA is able to do the =
refreshing and (b) what the proposed refresh time is. The UA can do all =
these tricky NAT measurement tricks and then propose the refresh time. =
Stupid UA may just propose a predefined value, e.g. 20 seconds. The =
registrar (or whatever is in the path) then can acknowlege the refresh =
role and leave the refreshing to the UA. Otherwise, it will send the =
refresh messages. In that case, it will probably use PING, cause it =
needs a response to keep the NAT alive in all cases - even if it is a =
"503 Not Implemented".
=20
Christian
=20


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charset=3Diso-8859-1">


<META content=3D"MSHTML 6.00.2800.1476" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D122221412-09112004>Doesn't the OPTIONS message do this job (of =
keeping a=20
link 'alive')?</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<P><FONT face=3DArial size=3D2>--</FONT> <BR><FONT face=3DArial =
size=3D2>Steve=20
Langstaff.</FONT> </P>
<DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
size=3D2>-----Original Message-----<BR><B>From:</B> sip-bounces@ietf.org =

[mailto:sip-bounces@ietf.org]<B>On Behalf Of </B>Christer Holmberg=20
(JO/LMF)<BR><B>Sent:</B> 09 November 2004 11:37<BR><B>To:</B> 'Christian =

Stredicke'; fluffy@cisco.com<BR><B>Cc:</B> =
sip@ietf.org<BR><B>Subject:</B> RE:=20
[Sip] PING/PONG<BR><BR></FONT></DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2>Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =
size=3D2>I=20
don't have anything PING, but no matter how short and effective (in the =
number=20
of SIP headers etc) we make it it is still pretty "heavy" on the=20
network.</FONT></SPAN></DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =
size=3D2>I=20
think we should, as I said yesterday, at least have a look at an option =
which=20
"combines" CRLF and PING (or whatever method is used), ie a UA can send =
CRLF=20
every X seconds, and PING every n*X seconds.</FONT></SPAN></DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =
size=3D2>Of=20
course, using a combined mechanism would mean that it would take longer =
to=20
detect a NAT failure, in case the NAT binding goes done while the UA =
sends CRLF.=20
</FONT></SPAN></DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =
size=3D2>But,=20
we also&nbsp;have to remember, that eventhough PING would have a =
response (which=20
CRLC does not have), due to the transcation timeout- and re-transmission =
timers=20
it would still take a while (until the transcation expires)&nbsp;to =
figure out=20
that the NAT binding has been closed.</FONT></SPAN></DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2>Christer Holmberg</FONT></SPAN></DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2>Ericsson Finland</FONT></SPAN></DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D345572311-09112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma><FONT=20
  size=3D2><SPAN class=3D345572311-09112004><FONT face=3DArial=20
  color=3D#0000ff></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma><FONT=20
  size=3D2><SPAN class=3D345572311-09112004>&nbsp;</SPAN>-----Original=20
  Message-----<BR><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org]<B>On Behalf Of </B>Christian=20
  Stredicke<BR><B>Sent:</B> 8. marraskuuta 2004 23:31<BR><B>To:</B>=20
  fluffy@cisco.com<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> [Sip]=20
  PING/PONG<BR><BR></DIV></FONT></FONT>
  <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana =
size=3D2>Hello=20
  Cullen,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana size=3D2>I =
think it is no=20
  harm if there is a PING message defined in SIP. I personally think it =
is=20
  overdue anyway. Implementors can then decide if they are using it or =
not. Lets=20
  leave that decision to the implementors!</FONT></SPAN></DIV>
  <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana =
size=3D2>But we have to=20
  consider who is sending keep-alive traffic. "IMHO" is it definitely =
better if=20
  the UA does this, because then e.g. usually it is ok to send CRLF keep =
alive=20
  or STUN.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana size=3D2>We =
register with=20
  a header that incates (a) that the UA is able to do the refreshing and =
(b)=20
  what the proposed refresh time is. The UA can do all these tricky NAT=20
  measurement tricks and then propose the refresh time. Stupid UA may =
just=20
  propose a predefined value, e.g. 20 seconds. The registrar (or =
whatever is in=20
  the path) then can acknowlege the refresh role and leave the =
refreshing to the=20
  UA. Otherwise, it will send the refresh messages. In that case, it =
will=20
  probably use PING, cause it needs a response to keep the NAT alive in =
all=20
  cases - even if it is a "503 Not Implemented".</FONT></SPAN></DIV>
  <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
  size=3D2>Christian</FONT></SPAN></DIV>
  <DIV><SPAN=20
class=3D251072321-08112004></SPAN>&nbsp;</DIV></BLOCKQUOTE></BODY></HTML>=


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References: <F8EFC4B4A8C016428BC1F589296D4FBF07B0876E@esealnt630.al.sw.ericsson.se>
Subject: Re: [Sip] PING/PONG
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----- Original Message ----- 
From: "Christer Holmberg (JO/LMF)"
> But, we also have to remember, that eventhough PING would
> have a response (which CRLC does not have), due to the
> transcation timeout- and re-transmission timers it would
> still take a while (until the transcation expires) to
> figure out that the NAT binding has been closed.
>

With UDP, the PING will create a new NAT binding and the server can update
its mapping for the UA.

With TCP, if you want to be able to detect that the NAT binding is gone on
the order of a transaction timeout, you would need to send a PING every 32
seconds or so. Given that NAT bindings for TCP live longer than that, I
don't see what the CRLF buys you.

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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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org  Tue Nov  9 08:02:03 2004
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To: "'Bob Penfield'" <bpenfield@acmepacket.com>,
        "'Christian Stredicke'"
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Subject: RE: [Sip] PING/PONG
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Hi,

>With UDP, the PING will create a new NAT binding and the 
>server can update its mapping for the UA.

So will CRLF, won't it? The difference is that the UA will not be informed that the message has reached the destination (since there is no response), but the question is if we really would need that information for every keep-nat-binding message that we send (ie we use CRLF most of the time, and every now and then we use a SIP request to make sure that everything really works).

>With TCP, if you want to be able to detect that the NAT binding is gone on
>the order of a transaction timeout, you would need to send a 
>PING every 32 seconds or so. Given that NAT bindings for TCP live longer 
>than that, I don't see what the CRLF buys you.

My point was that using PING you will not be able to detect problems immediately, since you will have to wait for the transaction timeout before you can determine that something is wrong. 

What I do NOT want to do is to send SIP requests too often, and therefor I was wondering if it would be possible to also use CRLF :)

Regards,

Christer Holmberg
Ericsson Finland



> 
> 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 sip-bounces@ietf.org  Tue Nov  9 08:24:21 2004
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From: "Christian Stredicke" <Christian.Stredicke@snom.de>
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All I am saying is lets define the tool "PING", if and how implementors
are using it is up to them.
=20
The other question is who sends the keep-alive. I would be good if the
UA does that job, but it is not backward compatible with the specs that
I know. There we need to specify something.
=20
CS


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Christer Holmberg (JO/LMF)
	Sent: Tuesday, November 09, 2004 7:16 AM
	To: Christian Stredicke; fluffy@cisco.com
	Cc: sip@ietf.org
	Subject: RE: [Sip] PING/PONG
=09
=09
	Hi,
	=20
	I don't have anything PING, but no matter how short and
effective (in the number of SIP headers etc) we make it it is still
pretty "heavy" on the network.
	=20
	I think we should, as I said yesterday, at least have a look at
an option which "combines" CRLF and PING (or whatever method is used),
ie a UA can send CRLF every X seconds, and PING every n*X seconds.
	=20
	Of course, using a combined mechanism would mean that it would
take longer to detect a NAT failure, in case the NAT binding goes done
while the UA sends CRLF.=20
	=20
	But, we also have to remember, that eventhough PING would have a
response (which CRLC does not have), due to the transcation timeout- and
re-transmission timers it would still take a while (until the
transcation expires) to figure out that the NAT binding has been closed.
	=20
	Regards,
	=20
	Christer Holmberg
	Ericsson Finland
	=20
	=20
	=20

		=20
		 -----Original Message-----
		From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org]On Behalf Of Christian Stredicke
		Sent: 8. marraskuuta 2004 23:31
		To: fluffy@cisco.com
		Cc: sip@ietf.org
		Subject: [Sip] PING/PONG
	=09
	=09
		Hello Cullen,
		=20
		I think it is no harm if there is a PING message defined
in SIP. I personally think it is overdue anyway. Implementors can then
decide if they are using it or not. Lets leave that decision to the
implementors!
		=20
		But we have to consider who is sending keep-alive
traffic. "IMHO" is it definitely better if the UA does this, because
then e.g. usually it is ok to send CRLF keep alive or STUN.
		=20
		We register with a header that incates (a) that the UA
is able to do the refreshing and (b) what the proposed refresh time is.
The UA can do all these tricky NAT measurement tricks and then propose
the refresh time. Stupid UA may just propose a predefined value, e.g. 20
seconds. The registrar (or whatever is in the path) then can acknowlege
the refresh role and leave the refreshing to the UA. Otherwise, it will
send the refresh messages. In that case, it will probably use PING,
cause it needs a response to keep the NAT alive in all cases - even if
it is a "503 Not Implemented".
		=20
		Christian
		=20


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D190085612-09112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>All I am saying is lets define the tool "PING", =

<STRONG>if</STRONG> and <STRONG>how</STRONG> implementors are using it =
is up to=20
them.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D190085612-09112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D190085612-09112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>The other question is <STRONG>who</STRONG> =
sends the=20
keep-alive. I would be good if the UA does that job, but it is not =
backward=20
compatible with the specs that I know. There we need to specify=20
something.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D190085612-09112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D190085612-09112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>CS</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Christer Holmberg=20
  (JO/LMF)<BR><B>Sent:</B> Tuesday, November 09, 2004 7:16 =
AM<BR><B>To:</B>=20
  Christian Stredicke; fluffy@cisco.com<BR><B>Cc:</B>=20
  sip@ietf.org<BR><B>Subject:</B> RE: [Sip] =
PING/PONG<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Hi,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff size=3D2>I=20
  don't have anything PING, but no matter how short and effective (in =
the number=20
  of SIP headers etc) we make it it is still pretty "heavy" on the=20
  network.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff size=3D2>I=20
  think we should, as I said yesterday, at least have a look at an =
option which=20
  "combines" CRLF and PING (or whatever method is used), ie a UA can =
send CRLF=20
  every X seconds, and PING every n*X seconds.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff size=3D2>Of=20
  course, using a combined mechanism would mean that it would take =
longer to=20
  detect a NAT failure, in case the NAT binding goes done while the UA =
sends=20
  CRLF. </FONT></SPAN></DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff size=3D2>But,=20
  we also&nbsp;have to remember, that eventhough PING would have a =
response=20
  (which CRLC does not have), due to the transcation timeout- and=20
  re-transmission timers it would still take a while (until the =
transcation=20
  expires)&nbsp;to figure out that the NAT binding has been=20
  closed.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Regards,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Christer Holmberg</FONT></SPAN></DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Ericsson Finland</FONT></SPAN></DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D345572311-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma><FONT=20
    size=3D2><SPAN class=3D345572311-09112004><FONT face=3DArial=20
    color=3D#0000ff></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma><FONT=20
    size=3D2><SPAN class=3D345572311-09112004>&nbsp;</SPAN>-----Original =

    Message-----<BR><B>From:</B> sip-bounces@ietf.org=20
    [mailto:sip-bounces@ietf.org]<B>On Behalf Of </B>Christian=20
    Stredicke<BR><B>Sent:</B> 8. marraskuuta 2004 23:31<BR><B>To:</B>=20
    fluffy@cisco.com<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] =

    PING/PONG<BR><BR></DIV></FONT></FONT>
    <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana =
size=3D2>Hello=20
    Cullen,</FONT></SPAN></DIV>
    <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana =
size=3D2>I think it is=20
    no harm if there is a PING message defined in SIP. I personally =
think it is=20
    overdue anyway. Implementors can then decide if they are using it or =
not.=20
    Lets leave that decision to the implementors!</FONT></SPAN></DIV>
    <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana =
size=3D2>But we have to=20
    consider who is sending keep-alive traffic. "IMHO" is it definitely =
better=20
    if the UA does this, because then e.g. usually it is ok to send CRLF =
keep=20
    alive or STUN.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana =
size=3D2>We register=20
    with a header that incates (a) that the UA is able to do the =
refreshing and=20
    (b) what the proposed refresh time is. The UA can do all these =
tricky NAT=20
    measurement tricks and then propose the refresh time. Stupid UA may =
just=20
    propose a predefined value, e.g. 20 seconds. The registrar (or =
whatever is=20
    in the path) then can acknowlege the refresh role and leave the =
refreshing=20
    to the UA. Otherwise, it will send the refresh messages. In that =
case, it=20
    will probably use PING, cause it needs a response to keep the NAT =
alive in=20
    all cases - even if it is a "503 Not =
Implemented".</FONT></SPAN></DIV>
    <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D251072321-08112004><FONT face=3DVerdana=20
    size=3D2>Christian</FONT></SPAN></DIV>
    <DIV><SPAN=20
class=3D251072321-08112004></SPAN>&nbsp;</DIV></BLOCKQUOTE></BLOCKQUOTE><=
/BODY></HTML>

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--===============0208285681==--



From sip-bounces@ietf.org  Tue Nov  9 08:36:07 2004
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Subject: RE: [Sip] GRUU Comments and interaction with outbound-connection
Date: Tue, 9 Nov 2004 15:24:29 +0200
Message-ID: <5816828233DEFA41807A6CFDFDF2343C3A8C39@esebe056.ntc.nokia.com>
Thread-Topic: [Sip] GRUU Comments and interaction with outbound-connection
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That's exactly what I was thinking. The Proxy/registrar can add the =
actual contact of the recipient as the first entry in the target set. =
That gets placed in the request-URI. Proxy routes to first route header =
(EP) in this case. EP routes to end point.

/Hisham

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On=20
> Behalf Of ext
> Alan Hawrylyshen
> Sent: 09.November.2004 00:18
> To: Jonathan Rosenberg
> Cc: Cullen Jennings; sip@ietf.org
> Subject: [Sip] GRUU Comments and interaction with outbound-connection
>=20
>=20
> Re: the Edge Proxy, Reg/Prox RR problems....
>=20
> Johnathan;
>=20
> It strikes me that the EP can directly route the GRUU since it is=20
> (except in the 3GPP case) in the same administrative / security /=20
> policy domain as the registrar and SHOULD be capable of detecting the=20
> GRUU's applicability when the initial request hits the EP.  An=20
> appropriately constructed (encoded) GRUU could be decoded and dealt=20
> with as soon as it reaches the UA's EP, no?
>=20
> And as I write this, Cullen has just proposed the same thing.
>=20
> Alan
>=20
>=20
>=20
> a l a n a t j a s o m i d o t c o m
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From sip-bounces@ietf.org  Tue Nov  9 08:47:00 2004
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Subject: Re: [Sip] PING/PONG
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----- Original Message ----- 
From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>

>
> Hi,
>
> >With UDP, the PING will create a new NAT binding and the
> >server can update its mapping for the UA.
>
> So will CRLF, won't it? The difference is that the UA will not be informed
> that the message has reached the destination (since there is no response),
> but the question is if we really would need that information for every
> keep-nat-binding message that we send (ie we use CRLF most of the time,
> and every now and then we use a SIP request to make sure that everything
> really works).

The CRLF does not contain enough information for the server to update its
mapping for the UA. There could be many UAs behind the same NAT, so the
server has no way to know which one it is.

The CRLF will keep an existing binding open in the NAT, but it does not
cover the case of a NAT reset. So the it depends on how long you are willing
to live without a valid binding in the server for the UA, and that would be
how often you would need to send a PING. You would only send CRLF too if
that was longer than the binding lifetime in the NAT.

>
> >With TCP, if you want to be able to detect that the NAT binding is gone
on
> >the order of a transaction timeout, you would need to send a
> >PING every 32 seconds or so. Given that NAT bindings for TCP live longer
> >than that, I don't see what the CRLF buys you.
>
> My point was that using PING you will not be able to detect problems
immediately, since you will have to wait for the transaction timeout before
you can determine that something is wrong.
>
> What I do NOT want to do is to send SIP requests too often, and therefor I
was wondering if it would be possible to also use CRLF :)
>
> Regards,
>
> Christer Holmberg
> Ericsson Finland
>
>
>
> >
> > 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 sip-bounces@ietf.org  Tue Nov  9 09:13:21 2004
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Subject: RE: [Sip] PING/PONG
Date: Tue, 9 Nov 2004 13:45:49 -0000
Message-ID: <45730E094814E44488F789C1CDED27AE03118566@gbnewp0758m.eu.ubiquity.net>
Thread-Topic: [Sip] PING/PONG
Thread-Index: AcTGVeD72vlivuzXRNmVFA8//spI5wABZoOgAAGeqgA=
From: "Chris Boulton" <cboulton@ubiquity.net>
To: "Christian Stredicke" <Christian.Stredicke@snom.de>,
        "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>,
        <fluffy@cisco.com>
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=20

All I am saying is lets define the tool "PING", if and how implementors
are using it is up to them.

=20

[Chris Boulton] Sorry if I am missing something BUT was does this buy
you (That say OPTIONs doesn't)?  IMHO wither we should use existing
methods to achieve this OR go with the STUN option for both transport
protocols - to keep it consistent.=20

=20

The other question is who sends the keep-alive. I would be good if the
UA does that job, but it is not backward compatible with the specs that
I know. There we need to specify something.

=20

[Chris Boulton] Whatever we decide we shouldn't restrict it to
'one-way'.  In different deployments and scenarios it should be possible
for either entity to keep a connection alive.

=20

CS

=09=20

=09
  _____=20=20


	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Christer Holmberg (JO/LMF)
	Sent: Tuesday, November 09, 2004 7:16 AM
	To: Christian Stredicke; fluffy@cisco.com
	Cc: sip@ietf.org
	Subject: RE: [Sip] PING/PONG

	Hi,

=09=20

	I don't have anything PING, but no matter how short and
effective (in the number of SIP headers etc) we make it it is still
pretty "heavy" on the network.

=09=20

	I think we should, as I said yesterday, at least have a look at
an option which "combines" CRLF and PING (or whatever method is used),
ie a UA can send CRLF every X seconds, and PING every n*X seconds.

=09=20

	Of course, using a combined mechanism would mean that it would
take longer to detect a NAT failure, in case the NAT binding goes done
while the UA sends CRLF.=20

=09=20

	But, we also have to remember, that eventhough PING would have a
response (which CRLC does not have), due to the transcation timeout- and
re-transmission timers it would still take a while (until the
transcation expires) to figure out that the NAT binding has been closed.

=09=20

	Regards,

=09=20

	Christer Holmberg

	Ericsson Finland

=09=20

=09=20

=09=20

=09=09=20

		 -----Original Message-----
		From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org]On Behalf Of Christian Stredicke
		Sent: 8. marraskuuta 2004 23:31
		To: fluffy@cisco.com
		Cc: sip@ietf.org
		Subject: [Sip] PING/PONG

		Hello Cullen,

=09=09=20

		I think it is no harm if there is a PING message defined
in SIP. I personally think it is overdue anyway. Implementors can then
decide if they are using it or not. Lets leave that decision to the
implementors!

=09=09=20

		But we have to consider who is sending keep-alive
traffic. "IMHO" is it definitely better if the UA does this, because
then e.g. usually it is ok to send CRLF keep alive or STUN.

=09=09=20

		We register with a header that incates (a) that the UA
is able to do the refreshing and (b) what the proposed refresh time is.
The UA can do all these tricky NAT measurement tricks and then propose
the refresh time. Stupid UA may just propose a predefined value, e.g. 20
seconds. The registrar (or whatever is in the path) then can acknowlege
the refresh role and leave the refreshing to the UA. Otherwise, it will
send the refresh messages. In that case, it will probably use PING,
cause it needs a response to keep the NAT alive in all cases - even if
it is a "503 Not Implemented".

=09=09=20

		Christian

=09=09=20



This email and any attached files are confidential and copyright protected.=
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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DVerdana><span style=3D'font-size:10.0pt;font-family:Verdana;color:bl=
ue'>All
I am saying is lets define the tool &quot;</span></font><font size=3D2
 color=3Dblue face=3DVerdana><span style=3D'font-size:10.0pt;font-family:Ve=
rdana;
 color:blue'>PING</span></font><font size=3D2 color=3Dblue face=3DVerdana><=
span
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>&quot;, <strong><=
b><font
face=3DVerdana><span style=3D'font-family:Verdana'>if</span></font></b></st=
rong>
and <strong><b><font face=3DVerdana><span style=3D'font-family:Verdana'>how=
</span></font></b></strong>
implementors are using it is up to them.</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3 color=3Dna=
vy
face=3D"Times New Roman"><span style=3D'font-size:12.0pt;color:navy'>&nbsp;=
</span></font></p>

<p class=3DMsoNormal><b><i><font size=3D2 color=3Dnavy face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy;font-weight:bold;
font-style:italic'>[</span></font></i></b><b><i><font size=3D2 color=3Dnavy
 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:navy;
 font-weight:bold;font-style:italic'>Chris Boulton</span></font></i></b><b>=
<i><font
size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size:10.0pt;font-fam=
ily:Arial;
color:navy;font-weight:bold;font-style:italic'>] Sorry if I am missing
something BUT was does this buy you (That say OPTIONs doesn&#8217;t)?&nbsp;
IMHO wither we should use existing methods to achieve this OR go with the S=
TUN
option for both transport protocols &#8211; to keep it consistent. </span><=
/font></i></b></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DVerdana><span style=3D'font-size:10.0pt;font-family:Verdana;color:bl=
ue'>The
other question is <strong><b><font face=3DVerdana><span style=3D'font-famil=
y:Verdana'>who</span></font></b></strong>
sends the keep-alive. I would be good if the UA does that job, but it is not
backward compatible with the specs that I know. There we need to specify
something.</span></font></p>

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

<p class=3DMsoNormal><b><i><font size=3D2 color=3Dnavy face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy;font-weight:bold;
font-style:italic'>[</span></font></i></b><b><i><font size=3D2 color=3Dnavy
 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:navy;
 font-weight:bold;font-style:italic'>Chris Boulton</span></font></i></b><b>=
<i><font
size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size:10.0pt;font-fam=
ily:Arial;
color:navy;font-weight:bold;font-style:italic'>] Whatever we decide we shou=
ldn&#8217;t
restrict it to &#8216;one-way&#8217;.&nbsp; In different deployments and
scenarios it should be possible for either entity to keep a connection aliv=
e.</span></font></i></b></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DVerdana><span style=3D'font-size:10.0pt;font-family:Verdana;color:bl=
ue'>CS</span></font></p>

<blockquote style=3D'border:none;border-left:solid blue 1.5pt;padding:0cm 0=
cm 0cm 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt'>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

<div class=3DMsoNormal align=3Dcenter style=3D'margin-left:36.0pt;text-alig=
n:center'><font
size=3D3 face=3D"Times New Roman"><span lang=3DDE style=3D'font-size:12.0pt=
'>

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

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

<p class=3DMsoNormal style=3D'margin-right:0cm;margin-bottom:12.0pt;margin-=
left:
36.0pt'><b><font size=3D2 face=3DTahoma><span lang=3DDE style=3D'font-size:=
10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span lang=3DDE style=3D'font-size:10.0pt;font-family:Tahoma'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b><span style=3D'font-w=
eight:
bold'>On Behalf Of </span></b>Christer Holmberg (JO/LMF)<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, November 09, =
2004
7:16 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Christian Stredicke;
fluffy@cisco.com<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [Sip] PING/PONG=
</span></font></p>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:blue'>=
Hi,</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:blue'>=
I don't
have anything </span></font><font size=3D2 color=3Dblue face=3DArial><span
 style=3D'font-size:10.0pt;font-family:Arial;color:blue'>PING</span></font>=
<font
size=3D2 color=3Dblue face=3DArial><span style=3D'font-size:10.0pt;font-fam=
ily:Arial;
color:blue'>, but no matter how short and effective (in the number of SIP
headers etc) we make it it is still pretty &quot;heavy&quot; on the network=
.</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:blue'>=
I think
we should, as I said yesterday, at least have a look at an option which
&quot;combines&quot; CRLF and </span></font><font size=3D2 color=3Dblue fac=
e=3DArial><span
 style=3D'font-size:10.0pt;font-family:Arial;color:blue'>PING</span></font>=
<font
size=3D2 color=3Dblue face=3DArial><span style=3D'font-size:10.0pt;font-fam=
ily:Arial;
color:blue'> (or whatever method is used), ie a UA can send CRLF every X
seconds, and </span></font><font size=3D2 color=3Dblue face=3DArial><span
 style=3D'font-size:10.0pt;font-family:Arial;color:blue'>PING</span></font>=
<font
size=3D2 color=3Dblue face=3DArial><span style=3D'font-size:10.0pt;font-fam=
ily:Arial;
color:blue'> every n*X seconds.</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:blue'>=
Of
course, using a combined mechanism would mean that it would take longer to
detect a NAT failure, in case the NAT binding goes done while the UA sends
CRLF. </span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:blue'>=
But, we
also&nbsp;have to remember, that eventhough </span></font><font size=3D2
 color=3Dblue face=3DArial><span style=3D'font-size:10.0pt;font-family:Aria=
l;
 color:blue'>PING</span></font><font size=3D2 color=3Dblue face=3DArial><sp=
an
style=3D'font-size:10.0pt;font-family:Arial;color:blue'> would have a respo=
nse
(which CRLC does not have), due to the transcation timeout- and re-transmis=
sion
timers it would still take a while (until the transcation expires)&nbsp;to
figure out that the NAT binding has been closed.</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:blue'>=
Regards,</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:blue'>=
Christer
Holmberg</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:blue'>=
Ericsson
</span></font><font size=3D2 color=3Dblue face=3DArial><span style=3D'font-=
size:10.0pt;
  font-family:Arial;color:blue'>Finland</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<blockquote style=3D'border:none;border-left:solid blue 1.5pt;padding:0cm 0=
cm 0cm 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt'>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

<p class=3DMsoNormal style=3D'margin-right:0cm;margin-bottom:12.0pt;margin-=
left:
36.0pt'><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-f=
amily:
Tahoma'>&nbsp;-----Original Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org]<b><span style=3D'font-weight:bold'>On Behalf =
Of </span></b>Christian
Stredicke<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 8. marraskuuta 2004 </=
span></font><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;fo=
nt-family:Tahoma'>23:31</span></font><font
size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>=
<br>
<b><span style=3D'font-weight:bold'>To:</span></b> fluffy@cisco.com<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] PING/PONG</sp=
an></font></p>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 face=3DVer=
dana><span
style=3D'font-size:10.0pt;font-family:Verdana'>Hello Cullen,</span></font><=
/p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 face=3DVer=
dana><span
style=3D'font-size:10.0pt;font-family:Verdana'>I think it is no harm if the=
re is
a </span></font><font size=3D2 face=3DVerdana><span style=3D'font-size:10.0=
pt;
 font-family:Verdana'>PING</span></font><font size=3D2 face=3DVerdana><span
style=3D'font-size:10.0pt;font-family:Verdana'> message defined in SIP. I
personally think it is overdue anyway. Implementors can then decide if they=
 are
using it or not. Lets leave that decision to the implementors!</span></font=
></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 face=3DVer=
dana><span
style=3D'font-size:10.0pt;font-family:Verdana'>But we have to consider who =
is
sending keep-alive traffic. &quot;IMHO&quot; is it definitely better if the=
 UA
does this, because then e.g. usually it is ok to send CRLF keep alive or ST=
UN.</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 face=3DVer=
dana><span
style=3D'font-size:10.0pt;font-family:Verdana'>We register with a header th=
at
incates (a) that the UA is able to do the refreshing and (b) what the propo=
sed
refresh time is. The UA can do all these tricky NAT measurement tricks and =
then
propose the refresh time. Stupid UA may just propose a predefined value, e.=
g.
20 seconds. The registrar (or whatever is in the path) then can acknowlege =
the
refresh role and leave the refreshing to the UA. Otherwise, it will send the
refresh messages. In that case, it will probably use </span></font><font
 size=3D2 face=3DVerdana><span style=3D'font-size:10.0pt;font-family:Verdan=
a'>PING</span></font><font
size=3D2 face=3DVerdana><span style=3D'font-size:10.0pt;font-family:Verdana=
'>, cause
it needs a response to keep the NAT alive in all cases - even if it is a
&quot;503 Not Implemented&quot;.</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 face=3DVer=
dana><span
style=3D'font-size:10.0pt;font-family:Verdana'>Christian</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

</blockquote>

</blockquote>

</div>

</body>

</html>
<br><br>
<FONT style=3D"BACKGROUND-COLOR: #ffffff">
<DIV>
<DIV><FONT face=3DVerdana color=3D#808080 size=3D1>This email and any attac=
hed files are confidential and copyright protected.&nbsp; If you are not th=
e addressee, any dissemination, distribution or copying of this communicati=
on is strictly prohibited.&nbsp; Unless otherwise expressly agreed in writi=
ng, nothing stated in this communication shall be legally binding.</FONT></=
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--===============1020041952==--



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Subject: RE: [Sip] PING/PONG
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From: "Christian Stredicke" <Christian.Stredicke@snom.de>
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________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Chris Boulton
Sent: Tuesday, November 09, 2004 9:31 AM
To: Christian Stredicke; Christer Holmberg (JO/LMF); fluffy@cisco.com
Cc: sip@ietf.org
Subject: RE: [Sip] PING/PONG



	=20

	All I am saying is lets define the tool "PING", if and how
implementors are using it is up to them.

	=20

	[Chris Boulton] Sorry if I am missing something BUT was does
this buy you (That say OPTIONs doesn't)?  IMHO wither we should use
existing methods to achieve this OR go with the STUN option for both
transport protocols - to keep it consistent. =20

	=20

	(CS) OPTIONS has the problem that it is relatively expensive
(size and CPU) and might result in responses that exceed the UDP
fragmentation limit (there are many DSL routers that treat UDP
fragmentation as security problem including the one that I have at
home). Therefore I think it would not hurt to have a specific message
that is designed only for the keep-alive job.

	=20

	The other question is who sends the keep-alive. I would be good
if the UA does that job, but it is not backward compatible with the
specs that I know. There we need to specify something.

	=20

	[Chris Boulton] Whatever we decide we shouldn't restrict it to
'one-way'.  In different deployments and scenarios it should be possible
for either entity to keep a connection alive.=20

	=20

	(CS) Sure. But the role has to be negotiated. If possible, the
UA should do that job. If it is not able to do this (e.g. because it
does not support refreshing) the fall back solution will always be that
the network side has to do the job.=20

	=20

	CS

		=20

	=09
________________________________


		From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]
On Behalf Of Christer Holmberg (JO/LMF)
		Sent: Tuesday, November 09, 2004 7:16 AM
		To: Christian Stredicke; fluffy@cisco.com
		Cc: sip@ietf.org
		Subject: RE: [Sip] PING/PONG

		Hi,

		=20

		I don't have anything PING, but no matter how short and
effective (in the number of SIP headers etc) we make it it is still
pretty "heavy" on the network.

		=20

		I think we should, as I said yesterday, at least have a
look at an option which "combines" CRLF and PING (or whatever method is
used), ie a UA can send CRLF every X seconds, and PING every n*X
seconds.

		=20

		Of course, using a combined mechanism would mean that it
would take longer to detect a NAT failure, in case the NAT binding goes
done while the UA sends CRLF.=20

		=20

		But, we also have to remember, that eventhough PING
would have a response (which CRLC does not have), due to the transcation
timeout- and re-transmission timers it would still take a while (until
the transcation expires) to figure out that the NAT binding has been
closed.

		=20

		Regards,

		=20

		Christer Holmberg

		Ericsson Finland

		=20

		=20

		=20

			=20

			 -----Original Message-----
			From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org]On Behalf Of Christian Stredicke
			Sent: 8. marraskuuta 2004 23:31
			To: fluffy@cisco.com
			Cc: sip@ietf.org
			Subject: [Sip] PING/PONG

			Hello Cullen,

			=20

			I think it is no harm if there is a PING message
defined in SIP. I personally think it is overdue anyway. Implementors
can then decide if they are using it or not. Lets leave that decision to
the implementors!

			=20

			But we have to consider who is sending
keep-alive traffic. "IMHO" is it definitely better if the UA does this,
because then e.g. usually it is ok to send CRLF keep alive or STUN.

			=20

			We register with a header that incates (a) that
the UA is able to do the refreshing and (b) what the proposed refresh
time is. The UA can do all these tricky NAT measurement tricks and then
propose the refresh time. Stupid UA may just propose a predefined value,
e.g. 20 seconds. The registrar (or whatever is in the path) then can
acknowlege the refresh role and leave the refreshing to the UA.
Otherwise, it will send the refresh messages. In that case, it will
probably use PING, cause it needs a response to keep the NAT alive in
all cases - even if it is a "503 Not Implemented".

			=20

			Christian

			=20



=09
	This email and any attached files are confidential and copyright
protected.  If you are not the addressee, any dissemination,
distribution or copying of this communication is strictly prohibited.
Unless otherwise expressly agreed in writing, nothing stated in this
communication shall be legally binding.


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<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Chris =
Boulton<BR><B>Sent:</B>=20
Tuesday, November 09, 2004 9:31 AM<BR><B>To:</B> Christian Stredicke; =
Christer=20
Holmberg (JO/LMF); fluffy@cisco.com<BR><B>Cc:</B>=20
sip@ietf.org<BR><B>Subject:</B> RE: [Sip] PING/PONG<BR></FONT><BR></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DSection1>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
  size=3D3><SPAN style=3D"FONT-SIZE: 12pt"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DVerdana =
color=3Dblue=20
  size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Verdana">All I=20
  am saying is lets define the tool "</SPAN></FONT><FONT face=3DVerdana =
color=3Dblue=20
  size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Verdana">PING</SPAN></FONT><FONT=20
  face=3DVerdana color=3Dblue size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Verdana">", =
<STRONG><B><FONT=20
  face=3DVerdana><SPAN style=3D"FONT-FAMILY: =
Verdana">if</SPAN></FONT></B></STRONG>=20
  and <STRONG><B><FONT face=3DVerdana><SPAN=20
  style=3D"FONT-FAMILY: Verdana">how</SPAN></FONT></B></STRONG> =
implementors are=20
  using it is up to them.</SPAN></FONT></P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
  color=3Dnavy size=3D3><SPAN=20
  style=3D"FONT-SIZE: 12pt; COLOR: navy"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial">[</SPAN></FONT><FONT=20
  face=3DArial color=3Dnavy><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial">Chris=20
  Boulton</SPAN></FONT><FONT face=3DArial color=3Dnavy><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial">]=20
  Sorry if I am missing something BUT was does this buy you (That say =
OPTIONs=20
  doesn&#8217;t)?&nbsp; IMHO wither we should use existing methods to =
achieve this OR=20
  go with the STUN option for both transport protocols &#8211; to keep =
it=20
  consistent.&nbsp;<SPAN class=3D546190215-09112004><FONT face=3DVerdana =

  color=3D#0000ff>&nbsp;</FONT></SPAN></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial"><SPAN=20
  class=3D546190215-09112004></SPAN></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial"><SPAN=20
  class=3D546190215-09112004><FONT face=3DVerdana color=3D#0000ff>(CS) =
OPTIONS has the=20
  problem that it is relatively expensive (size and CPU) and might =
result in=20
  responses that exceed the UDP fragmentation limit (there are many DSL =
routers=20
  that treat UDP fragmentation as security problem including the one =
that I have=20
  at home). Therefore I think it would not hurt to have a specific =
message that=20
  is designed only for the keep-alive =
job.</FONT></SPAN></SPAN></FONT></P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DVerdana =
color=3D#0000ff=20
  size=3D2><SPAN style=3D"FONT-SIZE: 12pt"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DVerdana =
color=3Dblue=20
  size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Verdana">The=20
  other question is <STRONG><B><FONT face=3DVerdana><SPAN=20
  style=3D"FONT-FAMILY: Verdana">who</SPAN></FONT></B></STRONG> sends =
the=20
  keep-alive. I would be good if the UA does that job, but it is not =
backward=20
  compatible with the specs that I know. There we need to specify=20
  something.</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial">[</SPAN></FONT><FONT=20
  face=3DArial color=3Dnavy><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial">Chris=20
  Boulton</SPAN></FONT><FONT face=3DArial color=3Dnavy><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial">]=20
  Whatever we decide we shouldn&#8217;t restrict it to =
&#8216;one-way&#8217;.&nbsp; In different=20
  deployments and scenarios it should be possible for either entity to =
keep a=20
  connection alive.<SPAN class=3D546190215-09112004><FONT face=3DVerdana =

  color=3D#0000ff>&nbsp;</FONT></SPAN></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial"><SPAN=20
  class=3D546190215-09112004></SPAN></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial"><SPAN=20
  class=3D546190215-09112004><FONT face=3DVerdana color=3D#0000ff>(CS) =
Sure. But the=20
  role has to be negotiated. If possible, the UA should do that job. If =
it is=20
  not able to do this (e.g. because it does not support refreshing) the =
fall=20
  back solution will always be that the network side has to do the job.=20
  </FONT></SPAN></SPAN></FONT></P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
  size=3D3><SPAN style=3D"FONT-SIZE: 12pt"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DVerdana =
color=3Dblue=20
  size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Verdana">CS</SPAN></FONT></P>
  <BLOCKQUOTE=20
  style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0cm; BORDER-TOP: =
medium none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0cm; MARGIN: 5pt 0cm 5pt =
3.75pt; BORDER-LEFT: blue 1.5pt solid; PADDING-TOP: 0cm; BORDER-BOTTOM: =
medium none">
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
    size=3D3><SPAN style=3D"FONT-SIZE: 12pt"></SPAN></FONT>&nbsp;</P>
    <DIV class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt; TEXT-ALIGN: =
center"=20
    align=3Dcenter><FONT face=3D"Times New Roman" size=3D3><SPAN =
lang=3DDE=20
    style=3D"FONT-SIZE: 12pt">
    <HR align=3Dcenter width=3D"100%" SIZE=3D2>
    </SPAN></FONT></DIV>
    <P class=3DMsoNormal=20
    style=3D"MARGIN-BOTTOM: 12pt; MARGIN-LEFT: 36pt; MARGIN-RIGHT: =
0cm"><B><FONT=20
    face=3DTahoma size=3D2><SPAN lang=3DDE=20
    style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: =
Tahoma">From:</SPAN></FONT></B><FONT=20
    face=3DTahoma size=3D2><SPAN lang=3DDE=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"> sip-bounces@ietf.org =

    [mailto:sip-bounces@ietf.org] <B><SPAN style=3D"FONT-WEIGHT: =
bold">On Behalf=20
    Of </SPAN></B>Christer Holmberg (JO/LMF)<BR><B><SPAN=20
    style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Tuesday, November 09, =
2004 7:16=20
    AM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Christian =
Stredicke;=20
    fluffy@cisco.com<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">Cc:</SPAN></B>=20
    sip@ietf.org<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">Subject:</SPAN></B> RE:=20
    [Sip] PING/PONG</SPAN></FONT></P>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DArial =
color=3Dblue=20
    size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">Hi,</SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
    size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DArial =
color=3Dblue=20
    size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">I=20
    don't have anything </SPAN></FONT><FONT face=3DArial color=3Dblue =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">PING</SPAN></FONT><FONT=20
    face=3DArial color=3Dblue size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">, but no =
matter how=20
    short and effective (in the number of SIP headers etc) we make it it =
is=20
    still pretty "heavy" on the network.</SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
    size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DArial =
color=3Dblue=20
    size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">I=20
    think we should, as I said yesterday, at least have a look at an =
option=20
    which "combines" CRLF and </SPAN></FONT><FONT face=3DArial =
color=3Dblue=20
    size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">PING</SPAN></FONT><FONT=20
    face=3DArial color=3Dblue size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial"> (or =
whatever=20
    method is used), ie a UA can send CRLF every X seconds, and=20
    </SPAN></FONT><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">PING</SPAN></FONT><FONT=20
    face=3DArial color=3Dblue size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial"> every =
n*X=20
    seconds.</SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
    size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DArial =
color=3Dblue=20
    size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">Of=20
    course, using a combined mechanism would mean that it would take =
longer to=20
    detect a NAT failure, in case the NAT binding goes done while the UA =
sends=20
    CRLF. </SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
    size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DArial =
color=3Dblue=20
    size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">But,=20
    we also&nbsp;have to remember, that eventhough </SPAN></FONT><FONT=20
    face=3DArial color=3Dblue size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">PING</SPAN></FONT><FONT=20
    face=3DArial color=3Dblue size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial"> would =
have a=20
    response (which CRLC does not have), due to the transcation timeout- =
and=20
    re-transmission timers it would still take a while (until the =
transcation=20
    expires)&nbsp;to figure out that the NAT binding has been=20
    closed.</SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
    size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DArial =
color=3Dblue=20
    size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">Regards,</SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
    size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DArial =
color=3Dblue=20
    size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Christer=20
    Holmberg</SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3DArial =
color=3Dblue=20
    size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Ericsson=20
    </SPAN></FONT><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">Finland</SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
    size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
    size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT face=3D"Times =
New Roman"=20
    size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
    <BLOCKQUOTE=20
    style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0cm; BORDER-TOP: =
medium none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0cm; MARGIN: 5pt 0cm 5pt =
3.75pt; BORDER-LEFT: blue 1.5pt solid; PADDING-TOP: 0cm; BORDER-BOTTOM: =
medium none">
      <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT =
face=3D"Times New Roman"=20
      size=3D3><SPAN style=3D"FONT-SIZE: 12pt"></SPAN></FONT>&nbsp;</P>
      <P class=3DMsoNormal=20
      style=3D"MARGIN-BOTTOM: 12pt; MARGIN-LEFT: 36pt; MARGIN-RIGHT: =
0cm"><FONT=20
      face=3DTahoma size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">&nbsp;-----Original =

      Message-----<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">From:</SPAN></B>=20
      sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]<B><SPAN=20
      style=3D"FONT-WEIGHT: bold">On Behalf Of </SPAN></B>Christian=20
      Stredicke<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> =
8.=20
      marraskuuta 2004 </SPAN></FONT><FONT face=3DTahoma size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Tahoma">23:31</SPAN></FONT><FONT=20
      face=3DTahoma size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"><BR><B><SPAN=20
      style=3D"FONT-WEIGHT: bold">To:</SPAN></B> =
fluffy@cisco.com<BR><B><SPAN=20
      style=3D"FONT-WEIGHT: bold">Cc:</SPAN></B> =
sip@ietf.org<BR><B><SPAN=20
      style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> [Sip]=20
      PING/PONG</SPAN></FONT></P>
      <DIV>
      <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT =
face=3DVerdana=20
      size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Verdana">Hello=20
      Cullen,</SPAN></FONT></P></DIV>
      <DIV>
      <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT =
face=3D"Times New Roman"=20
      size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
      <DIV>
      <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT =
face=3DVerdana=20
      size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Verdana">I =
think it is=20
      no harm if there is a </SPAN></FONT><FONT face=3DVerdana =
size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Verdana">PING</SPAN></FONT><FONT=20
      face=3DVerdana size=3D2><SPAN style=3D"FONT-SIZE: 10pt; =
FONT-FAMILY: Verdana">=20
      message defined in SIP. I personally think it is overdue anyway.=20
      Implementors can then decide if they are using it or not. Lets =
leave that=20
      decision to the implementors!</SPAN></FONT></P></DIV>
      <DIV>
      <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT =
face=3D"Times New Roman"=20
      size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
      <DIV>
      <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT =
face=3DVerdana=20
      size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Verdana">But =
we have to=20
      consider who is sending keep-alive traffic. "IMHO" is it =
definitely better=20
      if the UA does this, because then e.g. usually it is ok to send =
CRLF keep=20
      alive or STUN.</SPAN></FONT></P></DIV>
      <DIV>
      <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT =
face=3D"Times New Roman"=20
      size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
      <DIV>
      <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT =
face=3DVerdana=20
      size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Verdana">We =
register=20
      with a header that incates (a) that the UA is able to do the =
refreshing=20
      and (b) what the proposed refresh time is. The UA can do all these =
tricky=20
      NAT measurement tricks and then propose the refresh time. Stupid =
UA may=20
      just propose a predefined value, e.g. 20 seconds. The registrar =
(or=20
      whatever is in the path) then can acknowlege the refresh role and =
leave=20
      the refreshing to the UA. Otherwise, it will send the refresh =
messages. In=20
      that case, it will probably use </SPAN></FONT><FONT face=3DVerdana =

      size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Verdana">PING</SPAN></FONT><FONT=20
      face=3DVerdana size=3D2><SPAN style=3D"FONT-SIZE: 10pt; =
FONT-FAMILY: Verdana">,=20
      cause it needs a response to keep the NAT alive in all cases - =
even if it=20
      is a "503 Not Implemented".</SPAN></FONT></P></DIV>
      <DIV>
      <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT =
face=3D"Times New Roman"=20
      size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
      <DIV>
      <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT =
face=3DVerdana=20
      size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Verdana">Christian</SPAN></FONT></P></DIV>
      <DIV>
      <P class=3DMsoNormal style=3D"MARGIN-LEFT: 36pt"><FONT =
face=3D"Times New Roman"=20
      size=3D3><SPAN=20
    style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV></BLOCKQUOTE></BLOCKQUOTE></DIV><BR><=
BR><FONT=20
  style=3D"BACKGROUND-COLOR: #ffffff">
  <DIV>
  <DIV><FONT face=3DVerdana color=3D#808080 size=3D1>This email and any =
attached files=20
  are confidential and copyright protected.&nbsp; If you are not the =
addressee,=20
  any dissemination, distribution or copying of this communication is =
strictly=20
  prohibited.&nbsp; Unless otherwise expressly agreed in writing, =
nothing stated=20
  in this communication shall be legally=20
binding.</FONT></DIV></DIV></BLOCKQUOTE></FONT></BODY></HTML>

------_=_NextPart_001_01C4C66E.3FE9FFFE--


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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
--===============1083694966==--



From sip-bounces@ietf.org  Tue Nov  9 11:46:46 2004
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Subject: RE: [Sip] PING/PONG [bcc][faked-from][heur]
Date: Tue, 9 Nov 2004 17:25:51 +0100
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From: "Christian Jansson" <christian.jansson@hotsip.com>
To: "Christian Stredicke" <Christian.Stredicke@snom.de>,
        "Chris Boulton" <cboulton@ubiquity.net>,
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>>[Chris Boulton] Whatever we decide we shouldn't restrict it to =
'one-way'.=A0 In different deployments and scenarios it should be =
possible for either >>entity to keep a connection alive.=A0
>=A0
>(CS) Sure. But the role has to be negotiated. If possible, the UA =
should do that job. If it is not able to do this (e.g. because it does =
not support >refreshing) the fall back solution will always be that the =
network side has to do the job.=20
=A0
Does it really have to be negotiated? If we all think that refreshes =
should be handled by the UA, why not decide that that is the only =
standardized way to do it. Given that things have worked without this =
negotiation for quite some time now, things will not get any worse just =
because we write down a standardized way for a UA to do keep-alives. If =
some UAs did not take the keep-alive issue in consideration before this =
discussion begun, it could only mean two things:
1) The UA did not work in the real world, and I guess they would have =
noticed that.=20
2) They used STUN/CRLF, invented something else, or perhaps they sent =
OPTIONS/PING/??? from the server, which they can continue to do even =
though it is not the documented way of doing it. Either they update =
their UA according to "standard" keep-alives or they keep the =
proprietary solution, and I don't think the standard for keep-alives =
should try to compensate for all those solutions.=20

/ Christian Jansson


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From sip-bounces@ietf.org  Tue Nov  9 11:50:04 2004
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Subject: RE: [Sip] mib-08
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KL: "we need to move the I-D forward so we can get a RFC that can get =
some implementation experience."

I thought that the I-D phase was the time to get the implementation =
experience. Is it your intention that the RFC will be "Experimental" or =
"Informational" before getting this implementation experience, and then =
become a standard afterwards?=20

--
Steve Langstaff.


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Kevin Lingle
Sent: 08 November 2004 17:26
To: Cullen Jennings
Cc: sip@ietf.org
Subject: Re: [Sip] mib-08
[snip]

we need to move the
I-D forward so we can get a RFC that can get some implementation
experience.

kevin

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From sip-bounces@ietf.org  Tue Nov  9 13:38:21 2004
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Subject: RE: [Sip] mib-08
Date: Tue, 9 Nov 2004 20:28:52 +0200
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Thread-Topic: [Sip] mib-08
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From: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
To: "Steve Langstaff" <steve.langstaff@citel.com>,
        "Kevin Lingle" <klingle@cisco.com>
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Steve,

Actually implementation experience and interoperability is required only =
when time comes to advance a standards track Proposed Standard to Draft =
Standard phase. Although 'pre-standard' implementations are not =
discouraged in I-D phase, one cannot ensure interoperable versions =
before the standard, because the OID allocation for the root of the MIB =
module under mib-2 is being performed by IANA only very late, prior to =
the RFC publication.

Whether the document will be a standards track, or not, is an IESG =
decision based on the WG recommendation. =20

Regards,

Dan



> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On=20
> Behalf Of Steve Langstaff
> Sent: 09 November, 2004 6:37 PM
> To: Kevin Lingle
> Cc: sip@ietf.org; Cullen Jennings
> Subject: RE: [Sip] mib-08
>=20
>=20
> KL: "we need to move the I-D forward so we can get a RFC that=20
> can get some implementation experience."
>=20
> I thought that the I-D phase was the time to get the=20
> implementation experience. Is it your intention that the RFC=20
> will be "Experimental" or "Informational" before getting this=20
> implementation experience, and then become a standard afterwards?=20
>=20
> --
> Steve Langstaff.
>=20
>=20
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
> Kevin Lingle
> Sent: 08 November 2004 17:26
> To: Cullen Jennings
> Cc: sip@ietf.org
> Subject: Re: [Sip] mib-08
> [snip]
>=20
> we need to move the
> I-D forward so we can get a RFC that can get some implementation
> experience.
>=20
> kevin
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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From sip-bounces@ietf.org  Tue Nov  9 16:07:00 2004
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Subject: RE: [Sip] PING/PONG [bcc][faked-from][heur]
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thread-index: AcTGVeD72vlivuzXRNmVFA8//spI5wABZoOgAAGeqgAAAsmPsAACDlggAAoKK4A=
From: "Christian Stredicke" <Christian.Stredicke@snom.de>
To: "Christian Jansson" <christian.jansson@hotsip.com>,
        "Chris Boulton" <cboulton@ubiquity.net>,
        "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>,
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Well, the negotiation who refreshes is more or less the question if you =
have a legacy device or not.

I totally agree, if there is no legacy device, the UA should refresh.

But currently, there are only legacy devices or proprietary =
implementations.-) Lets agree on a standard, that's why were are here.

My proposal: A new header "Keep-Alive" in the REGISTER request:

(A) Propose refresh every 15 seconds, offer methods stun, register, crlf

REGISTER sip:snom.com SIP/2.0
Via: SIP/2.0/UDP 10.67.87.210:4064;branch=3Dz9hG4bK-g1gkta5je1yq;rport
From: "Christian Stredicke" <sip:cs@snom.com>;tag=3Dl3rj6dvhfl
To: "Christian Stredicke" <sip:cs@snom.com>
Call-ID: e52d914160a1-8yd3fpzu59l7@snom220
CSeq: 1 REGISTER
Max-Forwards: 70
Contact: <sip:cs@10.67.87.210:4064;line=3Do58ete25>;q=3D1.0
Keep-Alive: 15;method=3D"stun,crlf,register"
Expires: 3600
Content-Length: 0

(B) Other party acknowledges and selects stun refreshes:

SIP/2.0 200 Ok
Via: SIP/2.0/UDP =
10.67.87.210:4064;branch=3Dz9hG4bK-hypaxu2640zy;rport=3D4064;received=3D1=
30.129.97.54
From: "Christian Stredicke" <sip:cs@snom.com>;tag=3Dl3rj6dvhfl
To: "Christian Stredicke" <sip:cs@snom.com>;tag=3D0zikcb2fzo
Call-ID: e52d914160a1-8yd3fpzu59l7@snom220
CSeq: 1 REGISTER
Contact: <sip:cs@10.67.87.210:4064;line=3Do58ete25>;expires=3D3600
Keep-Alive: 15;method=3D"stun"
Content-Length: 0


CS=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Christian Jansson
> Sent: Tuesday, November 09, 2004 12:01 PM
> To: Christian Stredicke; Chris Boulton; Christer Holmberg=20
> (JO/LMF); fluffy@cisco.com
> Cc: sip@ietf.org
> Subject: RE: [Sip] PING/PONG [bcc][faked-from][heur]
>=20
>=20
>=20
> >>[Chris Boulton] Whatever we decide we shouldn't restrict it to=20
> >>'one-way'.=A0 In different deployments and scenarios it=20
> should be possible for either >>entity to keep a connection alive.
> >=A0
> >(CS) Sure. But the role has to be negotiated. If possible,=20
> the UA should do that job. If it is not able to do this (e.g.=20
> because it does not support >refreshing) the fall back=20
> solution will always be that the network side has to do the job.=20
> =A0
> Does it really have to be negotiated? If we all think that=20
> refreshes should be handled by the UA, why not decide that=20
> that is the only standardized way to do it. Given that things=20
> have worked without this negotiation for quite some time now,=20
> things will not get any worse just because we write down a=20
> standardized way for a UA to do keep-alives. If some UAs did=20
> not take the keep-alive issue in consideration before this=20
> discussion begun, it could only mean two things:
> 1) The UA did not work in the real world, and I guess they=20
> would have noticed that.=20
> 2) They used STUN/CRLF, invented something else, or perhaps=20
> they sent OPTIONS/PING/??? from the server, which they can=20
> continue to do even though it is not the documented way of=20
> doing it. Either they update their UA according to "standard"=20
> keep-alives or they keep the proprietary solution, and I=20
> don't think the standard for keep-alives should try to=20
> compensate for all those solutions.=20
>=20
> / Christian Jansson
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20
>=20
>=20

_______________________________________________
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From sip-bounces@ietf.org  Tue Nov  9 17:02:35 2004
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From: Alan Hawrylyshen <alan@jasomi.com>
Subject: Re: [Sip] GRUU Comments and interaction with outbound-connection
Date: Tue, 9 Nov 2004 16:52:20 -0500
To: hisham.khartabil@nokia.com
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On Nov 9, 2004, at 08:24, hisham.khartabil@nokia.com wrote:

> That's exactly what I was thinking. The Proxy/registrar can add the 
> actual contact of the recipient as the first entry in the target set. 
> That gets placed in the request-URI. Proxy routes to first route 
> header (EP) in this case. EP routes to end point.
>
> /Hisham

Johnathan pointed out (verbally to myself when I mentioned this) that 
there are problems with the EP routing the request instead of sending 
it back to the 'registrar /  proxy' that affect separation of services 
and that having the EP route to the endpoint directly breaks one of the 
reasons that you would want to separate the EP role and registrar role 
in the first place.  I have to admit that due to room noise and having 
not written it down at that moment; I no longer recall specifics, but 
it strikes me as plausible and I would request that Johnathan add some 
text to this thread to help raise awareness of what breaks when the EP 
forwards the request directly to the end UA instead of spiralling.

Thanks
Alan

a l a n a t j a s o m i d o t c o m


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From: "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>
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Hi mary, all
=20
When reading draft-ietf-sip-history-info-04.txt, I have trouble in =
understanding some of the statements which relate to the forwarding =
rules for history-entries subject to privacy. It is an important =
requirement that History-entrie(s) with a Privacy=3Dhistory, session, or =
header are indeed forwarded to entities which belong to the same trust =
domain. The removal of specific history-entries should only occur if the =
peer does not belong to the trust domain.
=20
In the current text (section 4.3.3.1.1) :
If a request is  being forwarded to a Request URI associated with a =
domain for which the proxy is not responsible and there is a Privacy =
header in the request with a priv-value of "session", "header" or =
"history", the proxy MUST remove any hi-entry(s) prior to forwarding.=20
=20
The current wording is misleading since it gives the impression (maybe =
intentionnal) that it is not possible to forward history-entries with =
Privacy statements to domains under the responsability of e.g. another =
operator belonging to the same trust domain.=20
=20
The concept of "trust domain" should be used when discussing the =
forwarding rules pertaining to information subject to privacy. =
Furthermore, the requirement for forwarding history-entries to trusted =
entities should be stated more clearly in the draft.
=20
Thank you for clarifying this point.
=20
Best regards,
s=E9bastien
=20
=20

________________________________

De : sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] De la part de =
Mary Barnes
Envoy=E9 : mercredi 3 novembre 2004 17:29
=C0 : 'Takuya Sawada'
Cc : 'sip@ietf.org'
Objet : RE: FW: [Sip] I-D ACTION:draft-ietf-sip-history-info-04.txt



Hi Takuya,=20

I apologize if I missed this specific point in your previous postings. =
You are correct, that index of 2 should have been 1.1 and this  actually =
affects the whole series and also affects the examples in  4.5.1 and =
4.5.2.  All the other entries are correct relative to the index of 2, =
it's just the 2 should have been a 1.1, unless of course that proxy had =
a good reason for starting at 2, which I don't explain so shouldn't use =
in the example. =20

It looks like I introduced that error in the -01 version when I was =
updating the example to include the index for all entries (as the index =
was originally optional). I traced the change back to changes I made =
during the middle of the day, so I can't even blame late nite editting.  =
 I will definitely make a note of that and update that in the -05 =
version, which will hopefully be the one to go to the IESG.

I'm also copying the SIP list, so that folks are aware of this as they =
review the document.=20

Thanks for your careful review,=20
Mary=20


-----Original Message-----=20
From: Takuya Sawada [mailto:tu-sawada@kddi.com]=20
Sent: Tuesday, November 02, 2004 4:29 AM=20
To: Barnes, Mary [NGC:B601:EXCH]=20
Subject: Re: FW: [Sip] I-D ACTION:draft-ietf-sip-history-info-04.txt=20


Hi Mary,=20

Each example flow in section 4.5 begins with the following messages,=20

   UA1        Proxy1  Proxy2     UA2      UA3      UA4      UA5=20
               =20
   |            |         |        |        |        |        |=20
   |--INVITE -->|         |        |        |        |        |=20
   |            |-INVITE->|        |        |        |        |=20
                 Supported: Histinfo=20
                 History-Info: <sip:Bob@P1.example.com>;index=3D1,=20
                               <sip:Bob@P2.example.com>;index=3D2=20

I think this should be=20

   UA1        Proxy1  Proxy2     UA2      UA3      UA4      UA5=20
               =20
   |            |         |        |        |        |        |=20
   |--INVITE -->|         |        |        |        |        |=20
   |            |-INVITE->|        |        |        |        |=20
                 Supported: Histinfo=20
                 History-Info: <sip:Bob@P1.example.com>;index=3D1,=20
                               <sip:Bob@P2.example.com>;index=3D1.1=20

Note that the index of the second hi-entry is 1.1.=20
I made the same comment to the list before, but no response to it.=20

In Appendix C, it shows,=20

   UA1          Proxy        ACDGRP1 Svr   ACDGRP2 Svr UA2-ACDGRP2       =
      =20
               =20
   |              |              |             |          |=20
   |--INVITE F1-->|              |             |          |=20
    Supported:Histinfo=20
   |              |              |             |          |=20
   |              |--INVITE F2-->|             |          |=20
                    Supported:Histinfo=20
                    History-Info: <sip:Gold@example.com>; index=3D1 =20
                    History-Info: <sip:ACDGRP1@example.com>; index=3D1.1 =


I can not find what is the difference between the two.=20
Are you saying that the former is "Retargeting within a  Proxy" and the=20
latter is "Basic Forwarding"?=20
Am I missing something?=20

Thanks.=20

Regards,=20
Takuya=20

>=20
>=20
> Hi all,=20
>=20
> Since this version should be ready for WGLC, a very detailed list of =
the=20
> changes is provided in the document, annotated as to the source, so I =
won't=20
> repeat those details here.   The majority of the changes were the =
issues=20
> discussed at IETF-60, along with the agreements there to change the =
text to=20
> non-normative in section 4 (Protocol structure) and to add some detail =
per=20
> Rohan's comment on the necessary processing should TLS not be =
available.  In=20
> addition, I received some proposed changes on a marked up hardcopy at=20
> IETF-61 from Eric Burger. =20
>=20
> The only items not discussed at IETF-60 were 2 items that came up on =
the=20
> list (one posted by John Elwell on August 18th on handling of privacy =
in=20
> responses) the other on Oct. 15th around the appropriate character =
format=20
> for the escaped headers in the URI. And, as always, there are various =
minor=20
> editorial changes here and there while I was "in the area".  So, there =

> should be no surprises with any of the changes, although, it is =
possible=20
> that there are still errors and nits that can be identified during =
WGLC.  =20
>=20
> Thanks,=20
> Mary=20
>=20
>=20
> -----Original Message-----=20
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of=20
> Internet-Drafts@ietf.org=20
> Sent: Monday, October 25, 2004 3:07 PM=20
> To: i-d-announce@ietf.org=20
> Cc: sip@ietf.org=20
> Subject: [Sip] I-D ACTION:draft-ietf-sip-history-info-04.txt=20
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts=20
> directories.=20
> This draft is a work item of the Session Initiation Protocol Working =
Group=20
> of the IETF.=20
>=20
>       Title           : An Extension to the Session Initiation =
Protocol=20
> for Request History Information=20
>       Author(s)       : M. Barnes=20
>       Filename        : draft-ietf-sip-history-info-04.txt=20
>       Pages           : 47=20
>       Date            : 2004-10-25=20
>      =20
> This draft defines a standard mechanism for capturing the history=20
>    information associated with a SIP request.  This capability enables =

>    many enhanced services by providing the information as to how and =
why=20
>    a call arrives at a specific application or user.  This draft =
defines=20
>    a new optional SIP header, History-Info, for capturing the history=20
>    information in requests. A new option tag, Histinfo, to be included =

>    in the Supported header, is defined to allow UAs to indicate =
whether=20
>    the History-Info should be returned in responses to a request which =

>    has captured the history information. A new priv-value, history, is =

>    added to the Privacy header to allow for privacy handling of the=20
>    History-Info header.=20
>=20
> A URL for this Internet-Draft is:=20
> http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-04.txt =

>=20
> To remove yourself from the I-D Announcement list, send a message to=20
> i-d-announce-request@ietf.org with the word unsubscribe in the body of =
the=20
> message. =20
> You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce =

> to change your subscription settings.=20
>=20
>=20
> Internet-Drafts are also available by anonymous FTP. Login with the =
username=20
> "anonymous" and a password of your e-mail address. After logging in,=20
> type "cd internet-drafts" and then=20
>       "get draft-ietf-sip-history-info-04.txt".=20
>=20
> A list of Internet-Drafts directories can be found in=20
> http://www.ietf.org/shadow.html=20
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt=20
>=20
>=20
> Internet-Drafts can also be obtained by e-mail.=20
>=20
> Send a message to:=20
>       mailserv@ietf.org.=20
> In the body type:=20
>       "FILE /internet-drafts/draft-ietf-sip-history-info-04.txt".=20
>      =20
> NOTE: The mail server at ietf.org can return the document in=20
>       MIME-encoded form by using the "mpack" utility.  To use this=20
>       feature, insert the command "ENCODING mime" before the "FILE"=20
>       command.  To decode the response(s), you will need "munpack" or=20
>       a MIME-compliant mail reader.  Different MIME-compliant mail =
readers=20
>       exhibit different behavior, especially when dealing with=20
>       "multipart" MIME messages (i.e. documents which have been split=20
>       up into multiple messages), so check your local documentation on =

>       how to manipulate these messages.=20
>              =20
>              =20
> Below is the data which will enable a MIME compliant mail reader=20
> implementation to automatically retrieve the ASCII version of the=20
> Internet-Draft.=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________=20
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip=20
> This list is for NEW development of the core SIP Protocol=20
> Use sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip=20
>=20


--------=20
Takuya Sawada=20
KDDI Corporation (KDDI)=20
Garden Air Tower, 3-10-10, Iidabashi,=20
Chiyoda-ku, Tokyo 102-8460, Japan=20
Tel: +81-3-6678-2997=20
Fax: +81-3-6678-0286=20
tu-sawada@kddi.com=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>RE: FW: [Sip] I-D =
ACTION:draft-ietf-sip-history-info-04.txt</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1476" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D019105313-08112004>Hi mary, all</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D019105313-08112004>When =
reading&nbsp;draft-ietf-sip-history-info-04.txt,=20
I&nbsp;have trouble in understanding some of the statements which relate =
to the=20
forwarding rules&nbsp;for history-entries subject to=20
privacy.&nbsp;</SPAN></FONT><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D019105313-08112004>It is an&nbsp;important=20
requirement&nbsp;that&nbsp;History-entrie(s) with&nbsp;a =
Privacy=3Dhistory,=20
session, or&nbsp;header are indeed forwarded&nbsp;to entities which=20
belong&nbsp;to&nbsp;the same&nbsp;trust domain. The removal of specific=20
history-entries should only occur&nbsp;if the peer does not belong to =
the trust=20
domain.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><SPAN =
class=3D019105313-08112004><FONT=20
size=3D2><FONT color=3D#0000ff>In the current text (section=20
4.3.3.1.1)&nbsp;:</FONT></FONT></SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><FONT =
face=3DArial color=3D#0000ff=20
size=3D2></FONT>If a request is&nbsp; being forwarded to a Request URI =
associated=20
with <STRONG>a domain for which&nbsp;the proxy is not =
responsible</STRONG> and=20
there is a Privacy header in the&nbsp;request&nbsp;<SPAN=20
class=3D019105313-08112004>w</SPAN>ith a priv-value of "session", =
"header" or=20
"history", the&nbsp;proxy MUST remove any hi-entry(s) prior to =
forwarding.=20
</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff =

size=3D2>The&nbsp;current&nbsp;wording is misleading since it gives the =
impression=20
(maybe intentionnal) that it is not possible to forward history-entries =
with=20
Privacy statements to domains under the responsability of&nbsp;e.g. =
another=20
operator belonging to the same trust domain. </FONT></SPAN></DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff =

size=3D2>The&nbsp;concept of "trust domain"&nbsp;should be&nbsp;used =
when=20
discussing the&nbsp;forwarding rules pertaining&nbsp;to information =
subject to=20
privacy. Furthermore,&nbsp;the requirement for forwarding =
history-entries to=20
trusted entities should be&nbsp;stated more clearly in the=20
draft.</FONT></SPAN></DIV>
<DIV><SPAN class=3D019105313-08112004></SPAN><SPAN =
class=3D019105313-08112004><FONT=20
face=3DArial color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff =
size=3D2>Thank=20
you&nbsp;for&nbsp;clarifying&nbsp;this point.</FONT></SPAN></DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff =
size=3D2>Best=20
regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff =

size=3D2>s=E9bastien</FONT></SPAN></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><BR></DIV>
<DIV class=3DOutlookMessageHeader lang=3Dfr dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>De&nbsp;:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>De la part de</B> Mary=20
Barnes<BR><B>Envoy=E9&nbsp;:</B> mercredi 3 novembre 2004 =
17:29<BR><B>=C0&nbsp;:</B>=20
'Takuya Sawada'<BR><B>Cc&nbsp;:</B> =
'sip@ietf.org'<BR><B>Objet&nbsp;:</B> RE:=20
FW: [Sip] I-D =
ACTION:draft-ietf-sip-history-info-04.txt<BR></FONT><BR></DIV>
<DIV></DIV>
<P><FONT size=3D2>Hi Takuya,</FONT> </P>
<P><FONT size=3D2>I apologize if I missed this specific point in your =
previous=20
postings. You are correct, that index of 2 should have been 1.1 and =
this&nbsp;=20
actually affects the whole series and also affects the examples in&nbsp; =
4.5.1=20
and 4.5.2.&nbsp; All the other entries are correct relative to the index =
of 2,=20
it's just the 2 should have been a 1.1, unless of course that proxy had =
a good=20
reason for starting at 2, which I don't explain so shouldn't use in the=20
example.&nbsp; </FONT></P>
<P><FONT size=3D2>It looks like I introduced that error in the -01 =
version when I=20
was updating the example to include the index for all entries (as the =
index was=20
originally optional). I traced the change back to changes I made during =
the=20
middle of the day, so I can't even blame late nite editting.&nbsp;&nbsp; =
I will=20
definitely make a note of that and update that in the -05 version, which =
will=20
hopefully be the one to go to the IESG.</FONT></P>
<P><FONT size=3D2>I'm also copying the SIP list, so that folks are aware =
of this=20
as they review the document. </FONT></P>
<P><FONT size=3D2>Thanks for your careful review,</FONT> <BR><FONT=20
size=3D2>Mary</FONT> </P><BR>
<P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From: Takuya=20
Sawada [<A =
href=3D"mailto:tu-sawada@kddi.com">mailto:tu-sawada@kddi.com</A>]=20
</FONT><BR><FONT size=3D2>Sent: Tuesday, November 02, 2004 4:29 =
AM</FONT>=20
<BR><FONT size=3D2>To: Barnes, Mary [NGC:B601:EXCH]</FONT> <BR><FONT=20
size=3D2>Subject: Re: FW: [Sip] I-D=20
ACTION:draft-ietf-sip-history-info-04.txt</FONT> </P><BR>
<P><FONT size=3D2>Hi Mary,</FONT> </P>
<P><FONT size=3D2>Each example flow in section 4.5 begins with the =
following=20
messages,</FONT> </P>
<P><FONT size=3D2>&nbsp;&nbsp; =
UA1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Proxy1&nbsp; Proxy2&nbsp;&nbsp;&nbsp;&nbsp; =
UA2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UA3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA4&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA5=20
</FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;=20
</FONT><BR><FONT size=3D2>&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp; |--INVITE=20
--&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|-INVITE-&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Supported: Histinfo </FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
History-Info: &lt;sip:Bob@P1.example.com&gt;;index=3D1,</FONT> <BR><FONT =

size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;sip:Bob@P2.example.com&gt;;index=3D2 </FONT></P>
<P><FONT size=3D2>I think this should be</FONT> </P>
<P><FONT size=3D2>&nbsp;&nbsp; =
UA1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Proxy1&nbsp; Proxy2&nbsp;&nbsp;&nbsp;&nbsp; =
UA2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UA3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA4&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA5=20
</FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;=20
</FONT><BR><FONT size=3D2>&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp; |--INVITE=20
--&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|-INVITE-&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Supported: Histinfo </FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
History-Info: &lt;sip:Bob@P1.example.com&gt;;index=3D1,</FONT> <BR><FONT =

size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;sip:Bob@P2.example.com&gt;;index=3D1.1</FONT> </P>
<P><FONT size=3D2>Note that the index of the second hi-entry is =
1.1.</FONT>=20
<BR><FONT size=3D2>I made the same comment to the list before, but no =
response to=20
it.</FONT> </P>
<P><FONT size=3D2>In Appendix C, it shows,</FONT> </P>
<P><FONT size=3D2>&nbsp;&nbsp;=20
UA1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Proxy&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ACDGRP1 Svr&nbsp;&nbsp; =
ACDGRP2=20
Svr=20
UA2-ACDGRP2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;=20
</FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;=20
</FONT><BR><FONT size=3D2>&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
</FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp; |--INVITE=20
F1--&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
</FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;&nbsp; Supported:Histinfo </FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
</FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;=20
|--INVITE=20
F2--&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
</FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Supported:Histinfo </FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
History-Info: &lt;sip:Gold@example.com&gt;; index=3D1&nbsp; =
</FONT><BR><FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
History-Info: &lt;sip:ACDGRP1@example.com&gt;; index=3D1.1 </FONT></P>
<P><FONT size=3D2>I can not find what is the difference between the =
two.</FONT>=20
<BR><FONT size=3D2>Are you saying that the former is "Retargeting within =
a&nbsp;=20
Proxy" and the </FONT><BR><FONT size=3D2>latter is "Basic =
Forwarding"?</FONT>=20
<BR><FONT size=3D2>Am I missing something?</FONT> </P>
<P><FONT size=3D2>Thanks.</FONT> </P>
<P><FONT size=3D2>Regards,</FONT> <BR><FONT size=3D2>Takuya</FONT> </P>
<P><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; </FONT><BR><FONT =
size=3D2>&gt;=20
Hi all,</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
Since this=20
version should be ready for WGLC, a very detailed list of the</FONT> =
<BR><FONT=20
size=3D2>&gt; changes is provided in the document, annotated as to the =
source, so=20
I won't</FONT> <BR><FONT size=3D2>&gt; repeat those details =
here.&nbsp;&nbsp; The=20
majority of the changes were the issues</FONT> <BR><FONT size=3D2>&gt; =
discussed=20
at IETF-60, along with the agreements there to change the text to</FONT> =

<BR><FONT size=3D2>&gt; non-normative in section 4 (Protocol structure) =
and to add=20
some detail per</FONT> <BR><FONT size=3D2>&gt; Rohan's comment on the =
necessary=20
processing should TLS not be available.&nbsp; In</FONT> <BR><FONT =
size=3D2>&gt;=20
addition, I received some proposed changes on a marked up hardcopy =
at</FONT>=20
<BR><FONT size=3D2>&gt; IETF-61 from Eric Burger.&nbsp; </FONT><BR><FONT =

size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; The only items not =
discussed at IETF-60=20
were 2 items that came up on the</FONT> <BR><FONT size=3D2>&gt; list =
(one posted=20
by John Elwell on August 18th on handling of privacy in</FONT> <BR><FONT =

size=3D2>&gt; responses) the other on Oct. 15th around the appropriate =
character=20
format</FONT> <BR><FONT size=3D2>&gt; for the escaped headers in the =
URI. And, as=20
always, there are various minor</FONT> <BR><FONT size=3D2>&gt; editorial =
changes=20
here and there while I was "in the area".&nbsp; So, there</FONT> =
<BR><FONT=20
size=3D2>&gt; should be no surprises with any of the changes, although, =
it is=20
possible</FONT> <BR><FONT size=3D2>&gt; that there are still errors and =
nits that=20
can be identified during WGLC.&nbsp;&nbsp; </FONT><BR><FONT =
size=3D2>&gt;=20
</FONT><BR><FONT size=3D2>&gt; Thanks,</FONT> <BR><FONT size=3D2>&gt; =
Mary</FONT>=20
<BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; </FONT><BR><FONT =
size=3D2>&gt;=20
-----Original Message-----</FONT> <BR><FONT size=3D2>&gt; From:=20
sip-bounces@ietf.org [<A=20
href=3D"mailto:sip-bounces@ietf.org">mailto:sip-bounces@ietf.org</A>] On =
Behalf=20
Of</FONT> <BR><FONT size=3D2>&gt; Internet-Drafts@ietf.org</FONT> =
<BR><FONT=20
size=3D2>&gt; Sent: Monday, October 25, 2004 3:07 PM</FONT> <BR><FONT =
size=3D2>&gt;=20
To: i-d-announce@ietf.org</FONT> <BR><FONT size=3D2>&gt; Cc: =
sip@ietf.org</FONT>=20
<BR><FONT size=3D2>&gt; Subject: [Sip] I-D=20
ACTION:draft-ietf-sip-history-info-04.txt</FONT> <BR><FONT size=3D2>&gt; =

</FONT><BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; A New =
Internet-Draft=20
is available from the on-line Internet-Drafts</FONT> <BR><FONT =
size=3D2>&gt;=20
directories.</FONT> <BR><FONT size=3D2>&gt; This draft is a work item of =
the=20
Session Initiation Protocol Working Group</FONT> <BR><FONT size=3D2>&gt; =
of the=20
IETF.</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : An Extension to the Session =

Initiation Protocol</FONT> <BR><FONT size=3D2>&gt; for Request History=20
Information</FONT> <BR><FONT size=3D2>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : M. Barnes</FONT> =
<BR><FONT=20
size=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :=20
draft-ietf-sip-history-info-04.txt</FONT> <BR><FONT size=3D2>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 47</FONT> <BR><FONT =
size=3D2>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2004-10-25</FONT> <BR><FONT =

size=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT><BR><FONT =
size=3D2>&gt; This=20
draft defines a standard mechanism for capturing the history =
</FONT><BR><FONT=20
size=3D2>&gt;&nbsp;&nbsp;&nbsp; information associated with a SIP =
request.&nbsp;=20
This capability enables </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; =
many=20
enhanced services by providing the information as to how and why=20
</FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; a call arrives at a =
specific=20
application or user.&nbsp; This draft defines </FONT><BR><FONT=20
size=3D2>&gt;&nbsp;&nbsp;&nbsp; a new optional SIP header, History-Info, =
for=20
capturing the history </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; =
information=20
in requests. A new option tag, Histinfo, to be included </FONT><BR><FONT =

size=3D2>&gt;&nbsp;&nbsp;&nbsp; in the Supported header, is defined to =
allow UAs=20
to indicate whether </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; the =

History-Info should be returned in responses to a request which =
</FONT><BR><FONT=20
size=3D2>&gt;&nbsp;&nbsp;&nbsp; has captured the history information. A =
new=20
priv-value, history, is </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; =
added to=20
the Privacy header to allow for privacy handling of the </FONT><BR><FONT =

size=3D2>&gt;&nbsp;&nbsp;&nbsp; History-Info header.</FONT> <BR><FONT =
size=3D2>&gt;=20
</FONT><BR><FONT size=3D2>&gt; A URL for this Internet-Draft is:</FONT> =
<BR><FONT=20
size=3D2>&gt; <A=20
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-0=
4.txt"=20
target=3D_blank>http://www.ietf.org/internet-drafts/draft-ietf-sip-histor=
y-info-04.txt</A></FONT>=20
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In the body type:</FONT> <BR><FONT size=3D2>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
"FILE /internet-drafts/draft-ietf-sip-history-info-04.txt".</FONT> =
<BR><FONT=20
size=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT><BR><FONT =
size=3D2>&gt; NOTE:=20
The mail server at ietf.org can return the document in</FONT> <BR><FONT=20
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mime"=20
before the "FILE"</FONT> <BR><FONT size=3D2>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
command.&nbsp; To decode the response(s), you will need "munpack" =
or</FONT>=20
<BR><FONT size=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a MIME-compliant =
mail=20
reader.&nbsp; Different MIME-compliant mail readers</FONT> <BR><FONT =
size=3D2>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit different behavior, especially =
when=20
dealing with</FONT> <BR><FONT size=3D2>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
"multipart" MIME messages (i.e. documents which have been split</FONT> =
<BR><FONT=20
size=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; up into multiple messages), =
so check=20
your local documentation on</FONT> <BR><FONT size=3D2>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; how to manipulate these messages.</FONT>=20
<BR><FONT size=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT><BR><FONT =
size=3D2>&gt;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</FONT><BR><FONT size=3D2>&gt; Below is the data which will enable a =
MIME=20
compliant mail reader</FONT> <BR><FONT size=3D2>&gt; implementation to=20
automatically retrieve the ASCII version of the</FONT> <BR><FONT =
size=3D2>&gt;=20
Internet-Draft.</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT =
size=3D2>&gt;=20
</FONT><BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
</FONT><BR><FONT=20
size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt;=20
_______________________________________________</FONT> <BR><FONT =
size=3D2>&gt; Sip=20
mailing list&nbsp; <A =
href=3D"https://www1.ietf.org/mailman/listinfo/sip"=20
target=3D_blank>https://www1.ietf.org/mailman/listinfo/sip</A></FONT> =
<BR><FONT=20
size=3D2>&gt; This list is for NEW development of the core SIP =
Protocol</FONT>=20
<BR><FONT size=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on=20
current sip</FONT> <BR><FONT size=3D2>&gt; Use sipping@ietf.org for new=20
developments on the application of sip</FONT> <BR><FONT size=3D2>&gt;=20
</FONT></P><BR>
<P><FONT size=3D2>--------</FONT> <BR><FONT size=3D2>Takuya =
Sawada</FONT> <BR><FONT=20
size=3D2>KDDI Corporation (KDDI)</FONT> <BR><FONT size=3D2>Garden Air =
Tower,=20
3-10-10, Iidabashi, </FONT><BR><FONT size=3D2>Chiyoda-ku, Tokyo =
102-8460,=20
Japan</FONT> <BR><FONT size=3D2>Tel: +81-3-6678-2997</FONT> <BR><FONT =
size=3D2>Fax:=20
+81-3-6678-0286</FONT> <BR><FONT size=3D2>tu-sawada@kddi.com</FONT>=20
</P></BODY></HTML>

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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
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From sip-bounces@ietf.org  Tue Nov  9 17:25:14 2004
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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'GARCIN Sebastien RD-CORE-ISS'" <sebastien.garcin@francetelecom.com>,
        sip@ietf.org
Subject: RE: [Sip] Privacy statements and History (draft-ietf-sip-history-
	info-04.txt)
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S=E9bastien,
=20
Thanks for reviewing the draft; my response to your comments is =
embedded
below [MB].
=20
Mary=20


-----Original Message-----
From: GARCIN Sebastien RD-CORE-ISS
[mailto:sebastien.garcin@francetelecom.com]=20
Sent: Monday, November 08, 2004 10:11 AM
To: Barnes, Mary [NGC:B601:EXCH]; sip@ietf.org
Subject: [Sip] Privacy statements and History
(draft-ietf-sip-history-info-04.txt)


Hi mary, all
=20
When reading draft-ietf-sip-history-info-04.txt, I have trouble in
understanding some of the statements which relate to the forwarding =
rules
for history-entries subject to privacy. It is an important requirement =
that
History-entrie(s) with a Privacy=3Dhistory, session, or header are =
indeed
forwarded to entities which belong to the same trust domain. The =
removal of
specific history-entries should only occur if the peer does not belong =
to
the trust domain.
=20
In the current text (section 4.3.3.1.1) :
If a request is  being forwarded to a Request URI associated with a =
domain
for which the proxy is not responsible and there is a Privacy header in =
the
request with a priv-value of "session", "header" or "history", the =
proxy
MUST remove any hi-entry(s) prior to forwarding.=20
=20
The current wording is misleading since it gives the impression (maybe
intentionnal) that it is not possible to forward history-entries with
Privacy statements to domains under the responsability of e.g. another
operator belonging to the same trust domain. =20
=20
[MB]: Current wording is consistent with terminology in RFC 3261 in =
terms of
describing who is able to change the Request URI in a specific request
(based on section 16.5):=20
   " A proxy MUST NOT add additional targets to the target set if the
   Request-URI of the original request does not indicate a resource =
this
   proxy is responsible for.

      A proxy can only change the Request-URI of a request during
      forwarding if it is responsible for that URI. "
Since History-Info (and associated privacy) are only added to the =
request,
when an entity that is allowed to change the Request-URI retargets the
request, it seemed sensible to use consistent wording to explain that.  =
=20
[/MB]
=20
The concept of "trust domain" should be used when discussing the =
forwarding
rules pertaining to information subject to privacy. Furthermore, the
requirement for forwarding history-entries to trusted entities should =
be
stated more clearly in the draft.=20

[MB]: The whole concept of what defines privacy in terms of the proxy's =
use
of the privacy header is outside the scope of History-Info =
functionality and
really a matter of local policy.   I think the functionality that you =
want
is a matter of local implementation and policy in terms of operators
establishing this "trust domain" model to which you refer.  =
History-Info
defines the mechanism to ensure the privacy of the requests, but it =
doesn't
explicitly define how the proxy knows whether it is responsible for =
that
resource.    I don't think this is a matter of standardization.  I =
thought
the use of the term "domain" rather than "resouce" would be helpful, =
but
perhaps changing it to the more general "resource" would resolve this
concern and/or a statement clarifying what I've just described should =
be
added in the draft.=20
[/MB]
=20
=20
Thank you for clarifying this point.
=20
Best regards,
s=E9bastien
=20
=20

  _____ =20

De : sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] De la part de =
Mary
Barnes
Envoy=E9 : mercredi 3 novembre 2004 17:29
=C0 : 'Takuya Sawada'
Cc : 'sip@ietf.org'
Objet : RE: FW: [Sip] I-D ACTION:draft-ietf-sip-history-info-04.txt



Hi Takuya,=20

I apologize if I missed this specific point in your previous postings. =
You
are correct, that index of 2 should have been 1.1 and this  actually =
affects
the whole series and also affects the examples in  4.5.1 and 4.5.2.  =
All the
other entries are correct relative to the index of 2, it's just the 2 =
should
have been a 1.1, unless of course that proxy had a good reason for =
starting
at 2, which I don't explain so shouldn't use in the example. =20

It looks like I introduced that error in the -01 version when I was =
updating
the example to include the index for all entries (as the index was
originally optional). I traced the change back to changes I made during =
the
middle of the day, so I can't even blame late nite editting.   I will
definitely make a note of that and update that in the -05 version, =
which
will hopefully be the one to go to the IESG.

I'm also copying the SIP list, so that folks are aware of this as they
review the document.=20

Thanks for your careful review,=20
Mary=20


-----Original Message-----=20
From: Takuya Sawada [mailto:tu-sawada@kddi.com =
<mailto:tu-sawada@kddi.com> ]

Sent: Tuesday, November 02, 2004 4:29 AM=20
To: Barnes, Mary [NGC:B601:EXCH]=20
Subject: Re: FW: [Sip] I-D ACTION:draft-ietf-sip-history-info-04.txt=20


Hi Mary,=20

Each example flow in section 4.5 begins with the following messages,=20

   UA1        Proxy1  Proxy2     UA2      UA3      UA4      UA5=20
               =20
   |            |         |        |        |        |        |=20
   |--INVITE -->|         |        |        |        |        |=20
   |            |-INVITE->|        |        |        |        |=20
                 Supported: Histinfo=20
                 History-Info: <sip:Bob@P1.example.com>;index=3D1,=20
                               <sip:Bob@P2.example.com>;index=3D2=20

I think this should be=20

   UA1        Proxy1  Proxy2     UA2      UA3      UA4      UA5=20
               =20
   |            |         |        |        |        |        |=20
   |--INVITE -->|         |        |        |        |        |=20
   |            |-INVITE->|        |        |        |        |=20
                 Supported: Histinfo=20
                 History-Info: <sip:Bob@P1.example.com>;index=3D1,=20
                               <sip:Bob@P2.example.com>;index=3D1.1=20

Note that the index of the second hi-entry is 1.1.=20
I made the same comment to the list before, but no response to it.=20

In Appendix C, it shows,=20

   UA1          Proxy        ACDGRP1 Svr   ACDGRP2 Svr UA2-ACDGRP2

               =20
   |              |              |             |          |=20
   |--INVITE F1-->|              |             |          |=20
    Supported:Histinfo=20
   |              |              |             |          |=20
   |              |--INVITE F2-->|             |          |=20
                    Supported:Histinfo=20
                    History-Info: <sip:Gold@example.com>; index=3D1 =20
                    History-Info: <sip:ACDGRP1@example.com>; =
index=3D1.1=20

I can not find what is the difference between the two.=20
Are you saying that the former is "Retargeting within a  Proxy" and the =

latter is "Basic Forwarding"?=20
Am I missing something?=20

Thanks.=20

Regards,=20
Takuya=20

>=20
>=20
> Hi all,=20
>=20
> Since this version should be ready for WGLC, a very detailed list of =
the=20
> changes is provided in the document, annotated as to the source, so I
won't=20
> repeat those details here.   The majority of the changes were the =
issues=20
> discussed at IETF-60, along with the agreements there to change the =
text
to=20
> non-normative in section 4 (Protocol structure) and to add some =
detail per

> Rohan's comment on the necessary processing should TLS not be =
available.
In=20
> addition, I received some proposed changes on a marked up hardcopy at =

> IETF-61 from Eric Burger. =20
>=20
> The only items not discussed at IETF-60 were 2 items that came up on =
the=20
> list (one posted by John Elwell on August 18th on handling of privacy =
in=20
> responses) the other on Oct. 15th around the appropriate character =
format=20
> for the escaped headers in the URI. And, as always, there are various
minor=20
> editorial changes here and there while I was "in the area".  So, =
there=20
> should be no surprises with any of the changes, although, it is =
possible=20
> that there are still errors and nits that can be identified during =
WGLC.

>=20
> Thanks,=20
> Mary=20
>=20
>=20
> -----Original Message-----=20
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org
<mailto:sip-bounces@ietf.org> ] On Behalf Of=20
> Internet-Drafts@ietf.org=20
> Sent: Monday, October 25, 2004 3:07 PM=20
> To: i-d-announce@ietf.org=20
> Cc: sip@ietf.org=20
> Subject: [Sip] I-D ACTION:draft-ietf-sip-history-info-04.txt=20
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts=20
> directories.=20
> This draft is a work item of the Session Initiation Protocol Working =
Group

> of the IETF.=20
>=20
>       Title           : An Extension to the Session Initiation =
Protocol=20
> for Request History Information=20
>       Author(s)       : M. Barnes=20
>       Filename        : draft-ietf-sip-history-info-04.txt=20
>       Pages           : 47=20
>       Date            : 2004-10-25=20
>      =20
> This draft defines a standard mechanism for capturing the history=20
>    information associated with a SIP request.  This capability =
enables=20
>    many enhanced services by providing the information as to how and =
why=20
>    a call arrives at a specific application or user.  This draft =
defines=20
>    a new optional SIP header, History-Info, for capturing the history =

>    information in requests. A new option tag, Histinfo, to be =
included=20
>    in the Supported header, is defined to allow UAs to indicate =
whether=20
>    the History-Info should be returned in responses to a request =
which=20
>    has captured the history information. A new priv-value, history, =
is=20
>    added to the Privacy header to allow for privacy handling of the=20
>    History-Info header.=20
>=20
> A URL for this Internet-Draft is:=20
> =
http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-04.txt
<http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-04.txt>=
 =20
>=20
> To remove yourself from the I-D Announcement list, send a message to=20
> i-d-announce-request@ietf.org with the word unsubscribe in the body =
of the

> message. =20
> You can also visit =
https://www1.ietf.org/mailman/listinfo/I-D-announce
<https://www1.ietf.org/mailman/listinfo/I-D-announce> =20
> to change your subscription settings.=20
>=20
>=20
> Internet-Drafts are also available by anonymous FTP. Login with the
username=20
> "anonymous" and a password of your e-mail address. After logging in,=20
> type "cd internet-drafts" and then=20
>       "get draft-ietf-sip-history-info-04.txt".=20
>=20
> A list of Internet-Drafts directories can be found in=20
> http://www.ietf.org/shadow.html <http://www.ietf.org/shadow.html> =20
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
<ftp://ftp.ietf.org/ietf/1shadow-sites.txt> =20
>=20
>=20
> Internet-Drafts can also be obtained by e-mail.=20
>=20
> Send a message to:=20
>       mailserv@ietf.org.=20
> In the body type:=20
>       "FILE /internet-drafts/draft-ietf-sip-history-info-04.txt".=20
>      =20
> NOTE: The mail server at ietf.org can return the document in=20
>       MIME-encoded form by using the "mpack" utility.  To use this=20
>       feature, insert the command "ENCODING mime" before the "FILE"=20
>       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=20
>       "multipart" MIME messages (i.e. documents which have been split =

>       up into multiple messages), so check your local documentation =
on=20
>       how to manipulate these messages.=20
>              =20
>              =20
> Below is the data which will enable a MIME compliant mail reader=20
> implementation to automatically retrieve the ASCII version of the=20
> Internet-Draft.=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________=20
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
<https://www1.ietf.org/mailman/listinfo/sip> =20
> This list is for NEW development of the core SIP Protocol=20
> Use sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip=20
>=20


--------=20
Takuya Sawada=20
KDDI Corporation (KDDI)=20
Garden Air Tower, 3-10-10, Iidabashi,=20
Chiyoda-ku, Tokyo 102-8460, Japan=20
Tel: +81-3-6678-2997=20
Fax: +81-3-6678-0286=20
tu-sawada@kddi.com=20


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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
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<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D163370919-08112004><SPAN =

class=3D019105313-08112004><FONT face=3DArial color=3D#800080=20
size=3D2>S=E9bastien,</FONT></SPAN></SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
class=3D163370919-08112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN =
class=3D163370919-08112004>Thanks=20
for reviewing the draft; my response to your comments is embedded below =

[MB].</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#800080 size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#800080><SPAN lang=3Den-us><FONT face=3DArial=20
size=3D2>Mary</FONT></SPAN> </FONT></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV><FONT color=3D#0000ff></FONT></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
GARCIN Sebastien=20
  RD-CORE-ISS [mailto:sebastien.garcin@francetelecom.com] =
<BR><B>Sent:</B>=20
  Monday, November 08, 2004 10:11 AM<BR><B>To:</B> Barnes, Mary =
[NGC:B601:EXCH];=20
  sip@ietf.org<BR><B>Subject:</B> [Sip] Privacy statements and History=20
  (draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D019105313-08112004>Hi mary, all</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D019105313-08112004>When =
reading&nbsp;draft-ietf-sip-history-info-04.txt,=20
  I&nbsp;have trouble in understanding some of the statements which =
relate to=20
  the forwarding rules&nbsp;for history-entries subject to=20
  privacy.&nbsp;</SPAN></FONT><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D019105313-08112004>It is an&nbsp;important=20
  requirement&nbsp;that&nbsp;History-entrie(s) with&nbsp;a =
Privacy=3Dhistory,=20
  session, or&nbsp;header are indeed forwarded&nbsp;to entities which=20
  belong&nbsp;to&nbsp;the same&nbsp;trust domain. The removal of =
specific=20
  history-entries should only occur&nbsp;if the peer does not belong to =
the=20
  trust domain.</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial><SPAN =
class=3D019105313-08112004><FONT=20
  size=3D2><FONT color=3D#0000ff>In the current text (section=20
  4.3.3.1.1)&nbsp;:</FONT></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT>If a request is&nbsp; being forwarded =
to a Request=20
  URI associated with <STRONG>a domain for which&nbsp;the proxy is not=20
  responsible</STRONG> and there is a Privacy header in=20
  the&nbsp;request&nbsp;<SPAN class=3D019105313-08112004>w</SPAN>ith a =
priv-value=20
  of "session", "header" or "history", the&nbsp;proxy MUST remove any=20
  hi-entry(s) prior to forwarding. </DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
  size=3D2>The&nbsp;current&nbsp;wording is misleading since it gives =
the=20
  impression (maybe intentionnal) that it is not possible to forward=20
  history-entries with Privacy statements to domains under the =
responsability=20
  of&nbsp;e.g. another operator belonging to the same trust =
domain.&nbsp;<SPAN=20
  =
class=3D163370919-08112004>&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DI=
V>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
  size=3D2><SPAN=20
  =
class=3D163370919-08112004></SPAN></FONT></FONT></FONT></SPAN>&nbsp;</DI=
V>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT><FONT =
color=3D#800080=20
  size=3D2><SPAN class=3D163370919-08112004>[MB]: Current wording is =
consistent with=20
  terminology in RFC 3261 in terms of describing who is able =
to&nbsp;change the=20
  Request URI in a specific request (based on&nbsp;section=20
  16.5):&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
color=3D#800080><SPAN=20
  class=3D163370919-08112004>&nbsp;&nbsp; " A proxy MUST NOT add =
additional=20
  targets to the target set if the<BR>&nbsp;&nbsp; Request-URI of the =
original=20
  request does not indicate a resource this<BR>&nbsp;&nbsp; proxy is =
responsible=20
  for.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; A proxy can only change =
the=20
  Request-URI of a request during<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
forwarding=20
  if it is responsible for that URI. =
"</SPAN></FONT></FONT></SPAN></DIV>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT><FONT =
color=3D#800080=20
  size=3D2><SPAN class=3D163370919-08112004>Since History-Info (and =
associated=20
  privacy)&nbsp;are only added to the request,&nbsp;when an&nbsp;entity =
that is=20
  allowed to change&nbsp;the Request-URI retargets the request, it =
seemed=20
  sensible to use consistent wording to explain that.&nbsp;=20
  &nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT><FONT =
color=3D#800080=20
  size=3D2><SPAN=20
  =
class=3D163370919-08112004>[/MB]</SPAN></FONT></FONT></FONT></SPAN></DIV=
>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#800080=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
  size=3D2>The&nbsp;concept of "trust domain"&nbsp;should be&nbsp;used =
when=20
  discussing the&nbsp;forwarding rules pertaining&nbsp;to information =
subject to=20
  privacy. Furthermore,&nbsp;the requirement for forwarding =
history-entries to=20
  trusted entities should be&nbsp;stated more clearly in the =
draft.<SPAN=20
  =
class=3D163370919-08112004>&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DI=
V>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
  size=3D2><SPAN class=3D163370919-08112004>
  <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
  class=3D163370919-08112004>[MB]: The whole concept of what defines =
privacy in=20
  terms of the proxy's use of the privacy header is outside the scope =
of=20
  History-Info functionality and really a matter of local policy. =
&nbsp; I think=20
  the functionality that you want is a matter of local implementation =
and policy=20
  in terms of&nbsp;operators establishing this "trust domain" =
model&nbsp;to=20
  which you refer.&nbsp; History-Info defines the mechanism =
to&nbsp;ensure the=20
  privacy of the requests, but it doesn't&nbsp;explicitly define how=20
  the&nbsp;proxy knows whether it is responsible for that=20
  resource.&nbsp;&nbsp;&nbsp;&nbsp;I don't think this is a matter of=20
  standardization.&nbsp; I thought the use of the term "domain" rather =
than=20
  "resouce" would be helpful, but perhaps changing it to the more =
general=20
  "resource" would resolve this concern and/or a statement clarifying =
what I've=20
  just described should be added in the draft. </SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
  class=3D163370919-08112004></SPAN></FONT><FONT face=3DArial =
color=3D#800080=20
  size=3D2><SPAN=20
  =
class=3D163370919-08112004>[/MB]</SPAN></FONT></DIV>&nbsp;</SPAN></FONT>=
</FONT></FONT></SPAN></DIV>
  <DIV><SPAN class=3D019105313-08112004></SPAN><SPAN=20
  class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Thank you&nbsp;for&nbsp;clarifying&nbsp;this =
point.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff size=3D2>Best=20
  regards,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>s=E9bastien</FONT></SPAN></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><BR></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Dfr dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>De&nbsp;:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>De la part de</B> Mary=20
  Barnes<BR><B>Envoy=E9&nbsp;:</B> mercredi 3 novembre 2004=20
  17:29<BR><B>=C0&nbsp;:</B> 'Takuya Sawada'<BR><B>Cc&nbsp;:</B>=20
  'sip@ietf.org'<BR><B>Objet&nbsp;:</B> RE: FW: [Sip] I-D=20
  ACTION:draft-ietf-sip-history-info-04.txt<BR></FONT><BR></DIV>
  <DIV></DIV>
  <P><FONT size=3D2>Hi Takuya,</FONT> </P>
  <P><FONT size=3D2>I apologize if I missed this specific point in your =
previous=20
  postings. You are correct, that index of 2 should have been 1.1 and =
this&nbsp;=20
  actually affects the whole series and also affects the examples =
in&nbsp; 4.5.1=20
  and 4.5.2.&nbsp; All the other entries are correct relative to the =
index of 2,=20
  it's just the 2 should have been a 1.1, unless of course that proxy =
had a good=20
  reason for starting at 2, which I don't explain so shouldn't use in =
the=20
  example.&nbsp; </FONT></P>
  <P><FONT size=3D2>It looks like I introduced that error in the -01 =
version when=20
  I was updating the example to include the index for all entries (as =
the index=20
  was originally optional). I traced the change back to changes I made =
during=20
  the middle of the day, so I can't even blame late nite =
editting.&nbsp;&nbsp; I=20
  will definitely make a note of that and update that in the -05 =
version, which=20
  will hopefully be the one to go to the IESG.</FONT></P>
  <P><FONT size=3D2>I'm also copying the SIP list, so that folks are =
aware of this=20
  as they review the document. </FONT></P>
  <P><FONT size=3D2>Thanks for your careful review,</FONT> <BR><FONT=20
  size=3D2>Mary</FONT> </P><BR>
  <P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From:=20
  Takuya Sawada [<A=20
  href=3D"mailto:tu-sawada@kddi.com">mailto:tu-sawada@kddi.com</A>]=20
  </FONT><BR><FONT size=3D2>Sent: Tuesday, November 02, 2004 4:29 =
AM</FONT>=20
  <BR><FONT size=3D2>To: Barnes, Mary [NGC:B601:EXCH]</FONT> <BR><FONT=20
  size=3D2>Subject: Re: FW: [Sip] I-D=20
  ACTION:draft-ietf-sip-history-info-04.txt</FONT> </P><BR>
  <P><FONT size=3D2>Hi Mary,</FONT> </P>
  <P><FONT size=3D2>Each example flow in section 4.5 begins with the =
following=20
  messages,</FONT> </P>
  <P><FONT size=3D2>&nbsp;&nbsp; =
UA1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Proxy1&nbsp; Proxy2&nbsp;&nbsp;&nbsp;&nbsp; =
UA2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  UA3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA4&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
UA5=20
  </FONT><BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </FONT><BR><FONT size=3D2>&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
  size=3D2>&nbsp;&nbsp; |--INVITE=20
  --&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
  size=3D2>&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |-INVITE-&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
  size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Supported: Histinfo </FONT><BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  History-Info: &lt;sip:Bob@P1.example.com&gt;;index=3D1,</FONT> =
<BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  &lt;sip:Bob@P2.example.com&gt;;index=3D2 </FONT></P>
  <P><FONT size=3D2>I think this should be</FONT> </P>
  <P><FONT size=3D2>&nbsp;&nbsp; =
UA1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Proxy1&nbsp; Proxy2&nbsp;&nbsp;&nbsp;&nbsp; =
UA2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  UA3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA4&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
UA5=20
  </FONT><BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </FONT><BR><FONT size=3D2>&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
  size=3D2>&nbsp;&nbsp; |--INVITE=20
  --&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
  size=3D2>&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |-INVITE-&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | </FONT><BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Supported: Histinfo </FONT><BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  History-Info: &lt;sip:Bob@P1.example.com&gt;;index=3D1,</FONT> =
<BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  &lt;sip:Bob@P2.example.com&gt;;index=3D1.1</FONT> </P>
  <P><FONT size=3D2>Note that the index of the second hi-entry is =
1.1.</FONT>=20
  <BR><FONT size=3D2>I made the same comment to the list before, but no =
response=20
  to it.</FONT> </P>
  <P><FONT size=3D2>In Appendix C, it shows,</FONT> </P>
  <P><FONT size=3D2>&nbsp;&nbsp;=20
  UA1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Proxy&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ACDGRP1 =
Svr&nbsp;&nbsp;=20
  ACDGRP2 Svr=20
  =
UA2-ACDGRP2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
  </FONT><BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </FONT><BR><FONT size=3D2>&nbsp;&nbsp;=20
  =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;=20
  =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;=20
  =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
</FONT><BR><FONT=20
  size=3D2>&nbsp;&nbsp; |--INVITE=20
  =
F1--&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;=20
  =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
</FONT><BR><FONT=20
  size=3D2>&nbsp;&nbsp;&nbsp; Supported:Histinfo </FONT><BR><FONT=20
  size=3D2>&nbsp;&nbsp;=20
  =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;=20
  =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
</FONT><BR><FONT=20
  size=3D2>&nbsp;&nbsp;=20
  =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;=20
  |--INVITE=20
  =
F2--&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;=20
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
</FONT><BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Supported:Histinfo </FONT><BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  History-Info: &lt;sip:Gold@example.com&gt;; index=3D1&nbsp; =
</FONT><BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  History-Info: &lt;sip:ACDGRP1@example.com&gt;; index=3D1.1 =
</FONT></P>
  <P><FONT size=3D2>I can not find what is the difference between the =
two.</FONT>=20
  <BR><FONT size=3D2>Are you saying that the former is "Retargeting =
within a&nbsp;=20
  Proxy" and the </FONT><BR><FONT size=3D2>latter is "Basic =
Forwarding"?</FONT>=20
  <BR><FONT size=3D2>Am I missing something?</FONT> </P>
  <P><FONT size=3D2>Thanks.</FONT> </P>
  <P><FONT size=3D2>Regards,</FONT> <BR><FONT size=3D2>Takuya</FONT> =
</P>
  <P><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
</FONT><BR><FONT size=3D2>&gt;=20
  Hi all,</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
Since this=20
  version should be ready for WGLC, a very detailed list of the</FONT> =
<BR><FONT=20
  size=3D2>&gt; changes is provided in the document, annotated as to =
the source,=20
  so I won't</FONT> <BR><FONT size=3D2>&gt; repeat those details =
here.&nbsp;&nbsp;=20
  The majority of the changes were the issues</FONT> <BR><FONT =
size=3D2>&gt;=20
  discussed at IETF-60, along with the agreements there to change the =
text=20
  to</FONT> <BR><FONT size=3D2>&gt; non-normative in section 4 =
(Protocol=20
  structure) and to add some detail per</FONT> <BR><FONT size=3D2>&gt; =
Rohan's=20
  comment on the necessary processing should TLS not be =
available.&nbsp;=20
  In</FONT> <BR><FONT size=3D2>&gt; addition, I received some proposed =
changes on=20
  a marked up hardcopy at</FONT> <BR><FONT size=3D2>&gt; IETF-61 from =
Eric=20
  Burger.&nbsp; </FONT><BR><FONT size=3D2>&gt; </FONT><BR><FONT =
size=3D2>&gt; The=20
  only items not discussed at IETF-60 were 2 items that came up on =
the</FONT>=20
  <BR><FONT size=3D2>&gt; list (one posted by John Elwell on August =
18th on=20
  handling of privacy in</FONT> <BR><FONT size=3D2>&gt; responses) the =
other on=20
  Oct. 15th around the appropriate character format</FONT> <BR><FONT =
size=3D2>&gt;=20
  for the escaped headers in the URI. And, as always, there are various =

  minor</FONT> <BR><FONT size=3D2>&gt; editorial changes here and there =
while I=20
  was "in the area".&nbsp; So, there</FONT> <BR><FONT size=3D2>&gt; =
should be no=20
  surprises with any of the changes, although, it is possible</FONT> =
<BR><FONT=20
  size=3D2>&gt; that there are still errors and nits that can be =
identified during=20
  WGLC.&nbsp;&nbsp; </FONT><BR><FONT size=3D2>&gt; </FONT><BR><FONT =
size=3D2>&gt;=20
  Thanks,</FONT> <BR><FONT size=3D2>&gt; Mary</FONT> <BR><FONT =
size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
-----Original=20
  Message-----</FONT> <BR><FONT size=3D2>&gt; From: =
sip-bounces@ietf.org [<A=20
  href=3D"mailto:sip-bounces@ietf.org">mailto:sip-bounces@ietf.org</A>] =
On Behalf=20
  Of</FONT> <BR><FONT size=3D2>&gt; Internet-Drafts@ietf.org</FONT> =
<BR><FONT=20
  size=3D2>&gt; Sent: Monday, October 25, 2004 3:07 PM</FONT> <BR><FONT =

  size=3D2>&gt; To: i-d-announce@ietf.org</FONT> <BR><FONT =
size=3D2>&gt; Cc:=20
  sip@ietf.org</FONT> <BR><FONT size=3D2>&gt; Subject: [Sip] I-D=20
  ACTION:draft-ietf-sip-history-info-04.txt</FONT> <BR><FONT =
size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; A New =
Internet-Draft=20
  is available from the on-line Internet-Drafts</FONT> <BR><FONT =
size=3D2>&gt;=20
  directories.</FONT> <BR><FONT size=3D2>&gt; This draft is a work item =
of the=20
  Session Initiation Protocol Working Group</FONT> <BR><FONT =
size=3D2>&gt; of the=20
  IETF.</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : An Extension to the =
Session=20
  Initiation Protocol</FONT> <BR><FONT size=3D2>&gt; for Request =
History=20
  Information</FONT> <BR><FONT size=3D2>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : M. Barnes</FONT> =
<BR><FONT=20
  size=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :=20
  draft-ietf-sip-history-info-04.txt</FONT> <BR><FONT size=3D2>&gt;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 47</FONT> <BR><FONT =
size=3D2>&gt;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2004-10-25</FONT> =
<BR><FONT=20
  size=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT><BR><FONT =
size=3D2>&gt; This=20
  draft defines a standard mechanism for capturing the history =
</FONT><BR><FONT=20
  size=3D2>&gt;&nbsp;&nbsp;&nbsp; information associated with a SIP =
request.&nbsp;=20
  This capability enables </FONT><BR><FONT =
size=3D2>&gt;&nbsp;&nbsp;&nbsp; many=20
  enhanced services by providing the information as to how and why=20
  </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; a call arrives at a =
specific=20
  application or user.&nbsp; This draft defines </FONT><BR><FONT=20
  size=3D2>&gt;&nbsp;&nbsp;&nbsp; a new optional SIP header, =
History-Info, for=20
  capturing the history </FONT><BR><FONT =
size=3D2>&gt;&nbsp;&nbsp;&nbsp;=20
  information in requests. A new option tag, Histinfo, to be included=20
  </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; in the Supported =
header, is=20
  defined to allow UAs to indicate whether </FONT><BR><FONT=20
  size=3D2>&gt;&nbsp;&nbsp;&nbsp; the History-Info should be returned =
in responses=20
  to a request which </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; =
has captured=20
  the history information. A new priv-value, history, is =
</FONT><BR><FONT=20
  size=3D2>&gt;&nbsp;&nbsp;&nbsp; added to the Privacy header to allow =
for privacy=20
  handling of the </FONT><BR><FONT size=3D2>&gt;&nbsp;&nbsp;&nbsp; =
History-Info=20
  header.</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
A URL for=20
  this Internet-Draft is:</FONT> <BR><FONT size=3D2>&gt; <A=20
  =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-=
04.txt"=20
  =
target=3D_blank>http://www.ietf.org/internet-drafts/draft-ietf-sip-histo=
ry-info-04.txt</A></FONT>=20
  <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; To remove =
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  I-D Announcement list, send a message to </FONT><BR><FONT =
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  </FONT><BR><FONT size=3D2>&gt; A list of Internet-Drafts directories =
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  or <A href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt"=20
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size=3D2>&gt;=20
  In the body type:</FONT> <BR><FONT size=3D2>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  "FILE /internet-drafts/draft-ietf-sip-history-info-04.txt".</FONT> =
<BR><FONT=20
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size=3D2>&gt; NOTE:=20
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mime"=20
  before the "FILE"</FONT> <BR><FONT size=3D2>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
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or</FONT>=20
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MIME-compliant mail=20
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  size=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit different =
behavior,=20
  especially when dealing with</FONT> <BR><FONT size=3D2>&gt;=20
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documents which=20
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size=3D2>&gt;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </FONT><BR><FONT size=3D2>&gt; Below is the data which will enable a =
MIME=20
  compliant mail reader</FONT> <BR><FONT size=3D2>&gt; implementation =
to=20
  automatically retrieve the ASCII version of the</FONT> <BR><FONT =
size=3D2>&gt;=20
  Internet-Draft.</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT =
size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
</FONT><BR><FONT=20
  size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt;=20
  _______________________________________________</FONT> <BR><FONT =
size=3D2>&gt;=20
  Sip mailing list&nbsp; <A =
href=3D"https://www1.ietf.org/mailman/listinfo/sip"=20
  target=3D_blank>https://www1.ietf.org/mailman/listinfo/sip</A></FONT> =
<BR><FONT=20
  size=3D2>&gt; This list is for NEW development of the core SIP =
Protocol</FONT>=20
  <BR><FONT size=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on=20
  current sip</FONT> <BR><FONT size=3D2>&gt; Use sipping@ietf.org for =
new=20
  developments on the application of sip</FONT> <BR><FONT size=3D2>&gt; =

  </FONT></P><BR>
  <P><FONT size=3D2>--------</FONT> <BR><FONT size=3D2>Takuya =
Sawada</FONT>=20
  <BR><FONT size=3D2>KDDI Corporation (KDDI)</FONT> <BR><FONT =
size=3D2>Garden Air=20
  Tower, 3-10-10, Iidabashi, </FONT><BR><FONT size=3D2>Chiyoda-ku, =
Tokyo 102-8460,=20
  Japan</FONT> <BR><FONT size=3D2>Tel: +81-3-6678-2997</FONT> <BR><FONT =

  size=3D2>Fax: +81-3-6678-0286</FONT> <BR><FONT =
size=3D2>tu-sawada@kddi.com</FONT>=20
  </P></BLOCKQUOTE></BODY></HTML>

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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
--===============1242122896==--



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(CS) OPTIONS has the problem that it is relatively expensive (size and =
CPU) and might result in responses that exceed the UDP fragmentation =
limit (there are many DSL routers that treat UDP fragmentation as =
security problem including the one that I have at home). Therefore I =
think it would not hurt to have a specific message that is designed only =
for the keep-alive job.
=20
...but if you want your keep alive message to be routed over the same =
path and transports as your original INVITE (and hence keep them alive), =
don't you need most of the "expensive" content of (say) the OPTIONS =
message - e.g. the vias.

--=20
Steve Langstaff.=20


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italic; FONT-FAMILY: Arial"><SPAN=20
class=3D546190215-09112004><FONT face=3DVerdana color=3D#0000ff>(CS) =
OPTIONS has the=20
problem that it is relatively expensive (size and CPU) and might result =
in=20
responses that exceed the UDP fragmentation limit (there are many DSL =
routers=20
that treat UDP fragmentation as security problem including the one that =
I have=20
at home). Therefore I think it would not hurt to have a specific message =
that is=20
designed only for the keep-alive job.</FONT></SPAN></SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D563593009-10112004><FONT face=3DArial color=3D#0000ff =
size=3D2>...but=20
if you want your keep alive message to be routed over the same path and=20
transports as your original INVITE (and hence keep them alive), don't =
you need=20
most of the "expensive" content of (say) the OPTIONS&nbsp;message - e.g. =
the=20
vias.</FONT></SPAN></DIV>
<P><FONT face=3DArial size=3D2>--</FONT> <BR><FONT face=3DArial =
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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
--===============1075131135==--



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From: "Christian Stredicke" <Christian.Stredicke@snom.de>
To: "Steve Langstaff" <steve.langstaff@citel.com>,
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True. For that purpose, use e.g. INFO messages with no body. These
messages keep the dialog "alive".=20
=20
However, the goal of the PING/PONG keep-alive messages is to keep the
NAT-binding alive. So that you are able to receive the first INVITE!
=20
CS


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Steve Langstaff
	Sent: Wednesday, November 10, 2004 5:16 AM
	To: Christian Stredicke; Chris Boulton; Christer Holmberg
(JO/LMF); fluffy@cisco.com
	Cc: sip@ietf.org
	Subject: RE: [Sip] PING/PONG
=09
=09
	(CS) OPTIONS has the problem that it is relatively expensive
(size and CPU) and might result in responses that exceed the UDP
fragmentation limit (there are many DSL routers that treat UDP
fragmentation as security problem including the one that I have at
home). Therefore I think it would not hurt to have a specific message
that is designed only for the keep-alive job.
	=20
	...but if you want your keep alive message to be routed over the
same path and transports as your original INVITE (and hence keep them
alive), don't you need most of the "expensive" content of (say) the
OPTIONS message - e.g. the vias.

	--=20
	Steve Langstaff.=20


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D548353310-10112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>True. For that purpose, use e.g. INFO messages =
with no=20
body. These messages keep the dialog "alive". </FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D548353310-10112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D548353310-10112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>However, the goal of the PING/PONG keep-alive =
messages is=20
to keep the NAT-binding alive. So that you are able to receive the first =

INVITE!</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D548353310-10112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D548353310-10112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>CS</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Steve=20
  Langstaff<BR><B>Sent:</B> Wednesday, November 10, 2004 5:16 =
AM<BR><B>To:</B>=20
  Christian Stredicke; Chris Boulton; Christer Holmberg (JO/LMF);=20
  fluffy@cisco.com<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> RE: =
[Sip]=20
  PING/PONG<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial"><SPAN=20
  class=3D546190215-09112004><FONT face=3DVerdana color=3D#0000ff>(CS) =
OPTIONS has the=20
  problem that it is relatively expensive (size and CPU) and might =
result in=20
  responses that exceed the UDP fragmentation limit (there are many DSL =
routers=20
  that treat UDP fragmentation as security problem including the one =
that I have=20
  at home). Therefore I think it would not hurt to have a specific =
message that=20
  is designed only for the keep-alive =
job.</FONT></SPAN></SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D563593009-10112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>...but if you want your keep alive message to be routed over =
the same=20
  path and transports as your original INVITE (and hence keep them =
alive), don't=20
  you need most of the "expensive" content of (say) the =
OPTIONS&nbsp;message -=20
  e.g. the vias.</FONT></SPAN></DIV>
  <P><FONT face=3DArial size=3D2>--</FONT> <BR><FONT face=3DArial =
size=3D2>Steve=20
  Langstaff.</FONT> <FONT=20
style=3D"BACKGROUND-COLOR: =
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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
--===============0775204233==--



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Subject: RE: [Sip] PING/PONG
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From: "Chris Boulton" <cboulton@ubiquity.net>
To: "Christian Stredicke" <Christian.Stredicke@snom.de>,
        "Steve Langstaff" <steve.langstaff@citel.com>,
        "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>,
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Yes - this has nothing to do with dialog's.  The requirement is to keep
bindings alive regardless.

=20

Chris.

=20

=20

-----Original Message-----
From: Christian Stredicke [mailto:Christian.Stredicke@snom.de]=20
Sent: 10 November 2004 10:36
To: Steve Langstaff; Chris Boulton; Christer Holmberg (JO/LMF);
fluffy@cisco.com
Cc: sip@ietf.org
Subject: RE: [Sip] PING/PONG

=20

True. For that purpose, use e.g. INFO messages with no body. These
messages keep the dialog "alive".=20

=20

However, the goal of the PING/PONG keep-alive messages is to keep the
NAT-binding alive. So that you are able to receive the first INVITE!

=20

CS

=09=20

=09
  _____=20=20


	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Steve Langstaff
	Sent: Wednesday, November 10, 2004 5:16 AM
	To: Christian Stredicke; Chris Boulton; Christer Holmberg
(JO/LMF); fluffy@cisco.com
	Cc: sip@ietf.org
	Subject: RE: [Sip] PING/PONG

	(CS) OPTIONS has the problem that it is relatively expensive
(size and CPU) and might result in responses that exceed the UDP
fragmentation limit (there are many DSL routers that treat UDP
fragmentation as security problem including the one that I have at
home). Therefore I think it would not hurt to have a specific message
that is designed only for the keep-alive job.

=09=20

	...but if you want your keep alive message to be routed over the
same path and transports as your original INVITE (and hence keep them
alive), don't you need most of the "expensive" content of (say) the
OPTIONS message - e.g. the vias.

	--=20
	Steve Langstaff.=20



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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
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10.0pt;font-family:Arial;color:navy'>Yes &#8211; this has nothing to do wit=
h dialog&#8217;s.&nbsp;
The requirement is to keep bindings alive regardless.</span></font></p>

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

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 face=3DTah=
oma><span
style=3D'font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> Christian Stredicke
[mailto:Christian.Stredicke@snom.de] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 10 November 2004 10:36=
<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Steve Langstaff; Chris
Boulton; Christer Holmberg (JO/LMF); fluffy@cisco.com<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [Sip] PING/PONG=
</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DVerdana><span style=3D'font-size:10.0pt;font-family:Verdana;color:bl=
ue'>True.
For that purpose, use e.g. INFO messages with no body. These messages keep =
the
dialog &quot;alive&quot;. </span></font></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DVerdana><span style=3D'font-size:10.0pt;font-family:Verdana;color:bl=
ue'>However,
the goal of the PING/PONG keep-alive messages is to keep the NAT-binding al=
ive.
So that you are able to receive the first INVITE!</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DVerdana><span style=3D'font-size:10.0pt;font-family:Verdana;color:bl=
ue'>CS</span></font></p>

<blockquote style=3D'border:none;border-left:solid blue 1.5pt;padding:0cm 0=
cm 0cm 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt'>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

<div class=3DMsoNormal align=3Dcenter style=3D'margin-left:36.0pt;text-alig=
n:center'><font
size=3D3 face=3D"Times New Roman"><span lang=3DDE style=3D'font-size:12.0pt=
'>

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

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

<p class=3DMsoNormal style=3D'margin-right:0cm;margin-bottom:12.0pt;margin-=
left:
36.0pt'><b><font size=3D2 face=3DTahoma><span lang=3DDE style=3D'font-size:=
10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span lang=3DDE style=3D'font-size:10.0pt;font-family:Tahoma'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b><span style=3D'font-w=
eight:
bold'>On Behalf Of </span></b>Steve Langstaff<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, November 10=
, 2004
5:16 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Christian Stredicke; Chr=
is
Boulton; Christer Holmberg (JO/LMF); fluffy@cisco.com<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [Sip] PING/PONG=
</span></font></p>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><b><i><font size=3D2 colo=
r=3Dblue
face=3DVerdana><span style=3D'font-size:10.0pt;font-family:Verdana;color:bl=
ue;
font-weight:bold;font-style:italic'>(CS) OPTIONS has the problem that it is
relatively expensive (size and CPU) and might result in responses that exce=
ed
the UDP fragmentation limit (there are many DSL routers that treat UDP
fragmentation as security problem including the one that I have at home).
Therefore I think it would not hurt to have a specific message that is desi=
gned
only for the keep-alive job.</span></font></i></b></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ue
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:blue'>=
...but
if you want your keep alive message to be routed over the same path and
transports as your original INVITE (and hence keep them alive), don't you n=
eed
most of the &quot;expensive&quot; content of (say) the OPTIONS&nbsp;message=
 -
e.g. the vias.</span></font></p>

</div>

<p style=3D'margin-left:36.0pt'><font size=3D2 face=3DArial><span style=3D'=
font-size:
10.0pt;font-family:Arial'>--</span></font> <br>
<font size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Ari=
al'>Steve
Langstaff.</span></font> </p>

</blockquote>

</div>

</body>

</html>
<br><br>
<FONT style=3D"BACKGROUND-COLOR: #ffffff">
<DIV>
<DIV><FONT face=3DVerdana color=3D#808080 size=3D1>This email and any attac=
hed files are confidential and copyright protected.&nbsp; If you are not th=
e addressee, any dissemination, distribution or copying of this communicati=
on is strictly prohibited.&nbsp; Unless otherwise expressly agreed in writi=
ng, nothing stated in this communication shall be legally binding.</FONT></=
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From: "Jesske, R" <R.Jesske@t-com.net>
To: sebastien.garcin@francetelecom.com, mary.barnes@nortelnetworks.com,
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Subject: AW: [Sip] Privacy statements and History (draft-ietf-sip-history-
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Dear Mary and Sebastien,
here are my Comments to Sebastien's statements:
=20
Your first proposal I can accept from my point of view.
On you second issue my impression is that this issue must be seen with =
regard to RFC3323 where privacy and the trust concept is described. =
Perhaps a reference to this document should be included.
On your last point, I think we can also refer to RFC 3323.
=20
What are you thinking?
=20
Best Regards
=20
Roland
=20

-----Urspr=FCngliche Nachricht-----
Von: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]Im Auftrag von =
GARCIN Sebastien RD-CORE-ISS
Gesendet: Dienstag, 9. November 2004 10:53
An: Mary Barnes; sip@ietf.org
Betreff: RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)


Mary,
=20
Responses below marked [SG]
=20
Best regards,=20
s=E9bastien=20

  _____ =20

De : sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] De la part de =
Mary Barnes
Envoy=E9 : lundi 8 novembre 2004 21:57
=C0 : Mary Barnes; GARCIN Sebastien RD-CORE-ISS; 'sip@ietf.org'
Objet : RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)


My initial response is awaiting approval by moderator due to the size, =
so I've snipped and am resending. =20

=20
Mary=20

-----Original Message-----
From: Barnes, Mary [NGC:B601:EXCH]=20
Sent: Monday, November 08, 2004 1:43 PM
To: 'GARCIN Sebastien RD-CORE-ISS'; sip@ietf.org
Subject: RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)


S=E9bastien,
=20
Thanks for reviewing the draft; my response to your comments is =
embedded below [MB].
=20
Mary=20


-----Original Message-----
From: GARCIN Sebastien RD-CORE-ISS =
[mailto:sebastien.garcin@francetelecom.com]=20
Sent: Monday, November 08, 2004 10:11 AM
To: Barnes, Mary [NGC:B601:EXCH]; sip@ietf.org
Subject: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)


Hi mary, all
=20
When reading draft-ietf-sip-history-info-04.txt, I have trouble in =
understanding some of the statements which relate to the forwarding =
rules for history-entries subject to privacy. It is an important =
requirement that History-entrie(s) with a Privacy=3Dhistory, session, =
or header are indeed forwarded to entities which belong to the same =
trust domain. The removal of specific history-entries should only occur =
if the peer does not belong to the trust domain.
=20
In the current text (section 4.3.3.1.1) :
If a request is  being forwarded to a Request URI associated with a =
domain for which the proxy is not responsible and there is a Privacy =
header in the request with a priv-value of "session", "header" or =
"history", the proxy MUST remove any hi-entry(s) prior to forwarding.=20
=20
The current wording is misleading since it gives the impression (maybe =
intentionnal) that it is not possible to forward history-entries with =
Privacy statements to domains under the responsability of e.g. another =
operator belonging to the same trust domain. =20
=20
[MB]: Current wording is consistent with terminology in RFC 3261 in =
terms of describing who is able to change the Request URI in a specific =
request (based on section 16.5):=20
   " A proxy MUST NOT add additional targets to the target set if the
   Request-URI of the original request does not indicate a resource =
this
   proxy is responsible for.

      A proxy can only change the Request-URI of a request during
      forwarding if it is responsible for that URI. "
Since History-Info (and associated privacy) are only added to the =
request, when an entity that is allowed to change the Request-URI =
retargets the request, it seemed sensible to use consistent wording to =
explain that.  =20
[/MB]=20
 [SG] I have no problem with the statements above. My concern is that =
if there is a hi-entry already embedded in the request with a Privacy =
statement, then, it should be up to local policy to decide whether or no=
t a proxy shall pass on those hi-entries to a trusted domain. The =
sentence in section 4.3.3.1.1 precludes this. I would propose to =
lighten the strenght of the sentence as follows:
=20
If a request is  being forwarded to a Request URI associated with a =
domain for which the proxy is not responsible and there is a Privacy =
header in the request with a priv-value of "session", "header" or =
"history", the proxy MAY remove any hi-entry(s) prior to forwarding.=20
=20
[SG] Another issue is the decision to add hi-entries in a privacy =
context. A proxy changing the target (i.e. the proxy is responsible of =
the resource reflected in the received Request-URI) of a request which =
contained a "privacy=3Dhistory" header MAY add a history-entries =
provided that it knows it can rely on other entities within the trust =
domain to apply the requested privacy. This affect item 4 in the list =
on conditions of section 4.3.3 and 4.3.3.1.=20
=20
The concept of "trust domain" should be used when discussing the =
forwarding rules pertaining to information subject to privacy. =
Furthermore, the requirement for forwarding history-entries to trusted =
entities should be stated more clearly in the draft.=20

[MB]: The whole concept of what defines privacy in terms of the proxy's =
use of the privacy header is outside the scope of History-Info =
functionality and really a matter of local policy.   I think the =
functionality that you want is a matter of local implementation and =
policy in terms of operators establishing this "trust domain" model to =
which you refer.  History-Info defines the mechanism to ensure the =
privacy of the requests, but it doesn't explicitly define how the proxy =
knows whether it is responsible for that resource.    I don't think =
this is a matter of standardization.  I thought the use of the term =
"domain" rather than "resouce" would be helpful, but perhaps changing =
it to the more general "resource" would resolve this concern and/or a =
statement clarifying what I've just described should be added in the =
draft.=20
[/MB]
[SG]   Changing the term "domain" to "resource" does not solve the =
problem I mentionned above. In order to reflect current operator =
requirements, the draft should not PRECLUDE the fowarding of existing =
history information towards trusted domains.
=20
Thank you for clarifying this point.
=20
Best regards,
s=E9bastien
=20
=20

------Remainder of non-related part of this thread has been deleted by =
Mary------------------


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3DISO-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.50.4937.800" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff size=3D2>Dear=20
Mary and Sebastien,</FONT></SPAN></DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff size=3D2>here=20
are my Comments to Sebastien's statements:</FONT></SPAN></DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff size=3D2>Your=20
first proposal I can accept from my point of view.</FONT></SPAN></DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff size=3D2>On you=20
second issue my impression is that this issue must be seen with regard =
to=20
RFC3323 where privacy and the trust concept is described. Perhaps a =
reference to=20
this document should be included.</FONT></SPAN></DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff size=3D2>On=20
your last point, I think we can also refer to RFC =
3323.</FONT></SPAN></DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2>What&nbsp;are you thinking?</FONT></SPAN></DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff size=3D2>Best=20
Regards</FONT></SPAN></DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2>Roland</FONT></SPAN></DIV>
<DIV><SPAN class=3D578113713-10112004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Urspr=FCngliche Nachricht-----<BR><B>Von:</B> =
sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org]<B>Im Auftrag von </B>GARCIN Sebastien=20
  RD-CORE-ISS<BR><B>Gesendet:</B> Dienstag, 9. November 2004 =
10:53<BR><B>An:</B>=20
  Mary Barnes; sip@ietf.org<BR><B>Betreff:</B> RE: [Sip] Privacy =
statements and=20
  History (draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D436390008-09112004>Mary,</SPAN></FONT></DIV>
  <DIV><SPAN class=3D436390008-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D436390008-09112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Responses below marked [SG]</FONT></SPAN></DIV>
  <DIV><SPAN class=3D436390008-09112004></SPAN>&nbsp;</DIV>
  <DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
  class=3D436390008-09112004>Best r</SPAN><SPAN=20
  =
class=3D436390008-09112004>egards,&nbsp;</SPAN></FONT></FONT></FONT></DI=
V>
  <DIV><FONT size=3D+0><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
  class=3D436390008-09112004></SPAN></FONT></FONT></FONT><SPAN=20
  class=3D436390008-09112004></SPAN><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
  size=3D2>s<SPAN=20
  =
class=3D436390008-09112004>=E9bastien&nbsp;</SPAN></FONT></FONT></FONT><=
BR></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Dfr dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>De&nbsp;:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>De la part de</B> Mary=20
  Barnes<BR><B>Envoy=E9&nbsp;:</B> lundi 8 novembre 2004 =
21:57<BR><B>=C0&nbsp;:</B>=20
  Mary Barnes; GARCIN Sebastien RD-CORE-ISS;=20
  'sip@ietf.org'<BR><B>Objet&nbsp;:</B> RE: [Sip] Privacy statements =
and History=20
  (draft-ietf-sip-history-info-04.txt)<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><SPAN class=3D761145520-08112004><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
  size=3D2>My initial response is awaiting approval by moderator due to =
the size,=20
  so I've snipped and am resending.&nbsp; =
<BR></FONT></FONT></FONT></SPAN></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN lang=3Den-us><FONT face=3DArial color=3D#0000ff =
size=3D2>Mary</FONT></SPAN>=20
  </DIV>
  <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
    <DIV></DIV>
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
    face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
Barnes, Mary=20
    [NGC:B601:EXCH] <BR><B>Sent:</B> Monday, November 08, 2004 1:43=20
    PM<BR><B>To:</B> 'GARCIN Sebastien RD-CORE-ISS';=20
    sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Privacy statements and =
History=20
    (draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
    <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D163370919-08112004><SPAN=20
    class=3D019105313-08112004><FONT face=3DArial color=3D#800080=20
    size=3D2>S=E9bastien,</FONT></SPAN></SPAN></FONT></DIV>
    <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
    class=3D163370919-08112004></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
    class=3D163370919-08112004>Thanks for reviewing the draft; my =
response to your=20
    comments is embedded below [MB].</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial color=3D#800080 =
size=3D2></FONT>&nbsp;</DIV>
    <DIV><FONT color=3D#800080><SPAN lang=3Den-us><FONT face=3DArial=20
    size=3D2>Mary</FONT></SPAN> </FONT></DIV>
    <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
      <DIV><FONT color=3D#0000ff></FONT></DIV>
      <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
      face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
GARCIN=20
      Sebastien RD-CORE-ISS [mailto:sebastien.garcin@francetelecom.com] =

      <BR><B>Sent:</B> Monday, November 08, 2004 10:11 AM<BR><B>To:</B> =
Barnes,=20
      Mary [NGC:B601:EXCH]; sip@ietf.org<BR><B>Subject:</B> [Sip] =
Privacy=20
      statements and History=20
      (draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
      <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
      class=3D019105313-08112004>Hi mary, all</SPAN></FONT></DIV>
      <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
      class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
      <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
      class=3D019105313-08112004>When=20
      reading&nbsp;draft-ietf-sip-history-info-04.txt, I&nbsp;have =
trouble in=20
      understanding some of the statements which relate to the =
forwarding=20
      rules&nbsp;for history-entries subject to=20
      privacy.&nbsp;</SPAN></FONT><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
      class=3D019105313-08112004>It is an&nbsp;important=20
      requirement&nbsp;that&nbsp;History-entrie(s) with&nbsp;a =
Privacy=3Dhistory,=20
      session, or&nbsp;header are indeed forwarded&nbsp;to entities =
which=20
      belong&nbsp;to&nbsp;the same&nbsp;trust domain. The removal of =
specific=20
      history-entries should only occur&nbsp;if the peer does not =
belong to the=20
      trust domain.</SPAN></FONT></DIV>
      <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
      class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
      <DIV dir=3Dltr align=3Dleft><FONT face=3DArial><SPAN=20
      class=3D019105313-08112004><FONT size=3D2><FONT =
color=3D#0000ff>In the current=20
      text (section 4.3.3.1.1)&nbsp;:</FONT></FONT></SPAN></FONT></DIV>
      <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><FONT =
face=3DArial=20
      color=3D#0000ff size=3D2></FONT>If a request is&nbsp; being =
forwarded to a=20
      Request URI associated with <STRONG>a domain for which&nbsp;the =
proxy is=20
      not responsible</STRONG> and there is a Privacy header in=20
      the&nbsp;request&nbsp;<SPAN =
class=3D019105313-08112004>w</SPAN>ith a=20
      priv-value of "session", "header" or "history", the&nbsp;proxy =
MUST remove=20
      any hi-entry(s) prior to forwarding. </DIV>
      <DIV><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>&nbsp;</DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
      color=3D#0000ff><FONT size=3D2>The&nbsp;current&nbsp;wording is =
misleading=20
      since it gives the impression (maybe intentionnal) that it is not =
possible=20
      to forward history-entries with Privacy statements to domains =
under the=20
      responsability of&nbsp;e.g. another operator belonging to the =
same trust=20
      domain.&nbsp;<SPAN=20
      =
class=3D163370919-08112004>&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DI=
V>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
      color=3D#0000ff><FONT size=3D2><SPAN=20
      =
class=3D163370919-08112004></SPAN></FONT></FONT></FONT></SPAN>&nbsp;</DI=
V>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
size=3D+0><FONT=20
      color=3D#800080 size=3D2><SPAN class=3D163370919-08112004>[MB]: =
Current wording=20
      is consistent with terminology in RFC 3261 in terms of describing =
who is=20
      able to&nbsp;change the Request URI in a specific request (based=20
      on&nbsp;section =
16.5):&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
      color=3D#800080><SPAN class=3D163370919-08112004>&nbsp;&nbsp; " A =
proxy MUST=20
      NOT add additional targets to the target set if =
the<BR>&nbsp;&nbsp;=20
      Request-URI of the original request does not indicate a resource=20
      this<BR>&nbsp;&nbsp; proxy is responsible=20
      for.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; A proxy can only =
change the=20
      Request-URI of a request during<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =

      forwarding if it is responsible for that URI.=20
      "</SPAN></FONT></FONT></SPAN></DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT =
size=3D+0><FONT=20
      color=3D#800080 size=3D2><SPAN class=3D163370919-08112004>Since =
History-Info=20
      (and associated privacy)&nbsp;are only added to the =
request,&nbsp;when=20
      an&nbsp;entity that is allowed to change&nbsp;the Request-URI =
retargets=20
      the request, it seemed sensible to use consistent wording to =
explain=20
      that.&nbsp; &nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT color=3D#800080><SPAN =

      class=3D163370919-08112004><FONT face=3DArial><FONT =
size=3D2>[/MB]<SPAN=20
      class=3D436390008-09112004><FONT=20
      =
color=3D#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></SPAN></FONT></SPAN><=
/DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT color=3D#800080><SPAN =

      class=3D163370919-08112004><FONT face=3DArial><FONT =
size=3D2><SPAN=20
      =
class=3D436390008-09112004>&nbsp;</SPAN></FONT></FONT></SPAN></FONT></SP=
AN><SPAN=20
      class=3D019105313-08112004><FONT face=3DArial color=3D#800080 =
size=3D2><SPAN=20
      class=3D436390008-09112004><STRONG><FONT color=3D#808080>[SG] I =
have no=20
      problem with the statements above.&nbsp;My concern is that if =
there is=20
      a&nbsp;hi-entry&nbsp;<U>already</U> embedded in the request with =
a Privacy=20
      statement, then, it should be&nbsp;up to local policy to decide =
whether or=20
      not&nbsp;a proxy shall pass on those hi-entries to a trusted =
domain. The=20
      sentence in section 4.3.3.1.1 precludes this.&nbsp;I would =
propose to=20
      lighten the strenght of the sentence as=20
      follows:</FONT></STRONG></SPAN></FONT></SPAN></DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#808080=20
      size=3D2><SPAN=20
      =
class=3D436390008-09112004><STRONG></STRONG></SPAN></FONT></SPAN>&nbsp;<=
/DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT size=3D+0><SPAN=20
      class=3D436390008-09112004>If a request is&nbsp; being forwarded =
to a=20
      Request URI associated with <STRONG>a domain for which&nbsp;the =
proxy is=20
      not responsible</STRONG> and there is a Privacy header in=20
      the&nbsp;request&nbsp;<SPAN =
class=3D019105313-08112004>w</SPAN>ith a=20
      priv-value of "session", "header" or "history", the&nbsp;proxy&nbs=
p;MAY=20
      remove any hi-entry(s) prior to forwarding. =
</SPAN></FONT></SPAN></DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#808080=20
      size=3D2><SPAN =
class=3D436390008-09112004></SPAN></FONT></SPAN><SPAN=20
      class=3D019105313-08112004><FONT face=3DArial color=3D#808080 =
size=3D2><SPAN=20
      =
class=3D436390008-09112004><STRONG></STRONG></SPAN></FONT></SPAN>&nbsp;<=
/DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#808080=20
      size=3D2><SPAN =
class=3D436390008-09112004><STRONG>[SG]&nbsp;Another issue is=20
      the decision to add hi-entries in a&nbsp;privacy context. A proxy =
changing=20
      the target (i.e. the proxy is responsible of the resource =
reflected in the=20
      received Request-URI) of a request which contained a=20
      "privacy=3Dhistory"&nbsp;header MAY add&nbsp;a =
history-entries&nbsp;provided=20
      that&nbsp;it&nbsp;knows it can rely on other entities within=20
      the&nbsp;trust domain to apply the requested privacy. This affect =
item 4=20
      in the list on conditions of section 4.3.3 and 4.3.3.1.=20
      </STRONG></SPAN></FONT></SPAN></DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><SPAN=20
      class=3D436390008-09112004></SPAN></FONT></SPAN>&nbsp;</DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
      color=3D#0000ff><FONT size=3D2>The&nbsp;concept of "trust =
domain"&nbsp;should=20
      be&nbsp;used when discussing the&nbsp;forwarding rules =
pertaining&nbsp;to=20
      information subject to privacy. Furthermore,&nbsp;the requirement =
for=20
      forwarding history-entries to trusted entities should =
be&nbsp;stated more=20
      clearly in the draft.<SPAN=20
      =
class=3D163370919-08112004>&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DI=
V>
      <DIV><SPAN class=3D019105313-08112004><FONT color=3D#0000ff><SPAN =

      class=3D163370919-08112004>
      <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
      class=3D163370919-08112004>[MB]: The whole concept of what =
defines privacy=20
      in terms of the proxy's use of the privacy header is outside the =
scope of=20
      History-Info functionality and really a matter of local policy. =
&nbsp; I=20
      think the functionality that you want is a matter of local =
implementation=20
      and policy in terms of&nbsp;operators establishing this "trust =
domain"=20
      model&nbsp;to which you refer.&nbsp; History-Info defines the =
mechanism=20
      to&nbsp;ensure the privacy of the requests, but it =
doesn't&nbsp;explicitly=20
      define how the&nbsp;proxy knows whether it is responsible for =
that=20
      resource.&nbsp;&nbsp;&nbsp;&nbsp;I don't think this is a matter =
of=20
      standardization.&nbsp; I thought the use of the term "domain" =
rather than=20
      "resouce" would be helpful, but perhaps changing it to the more =
general=20
      "resource" would resolve this concern and/or a statement =
clarifying what=20
      I've just described should be added in the draft. =
</SPAN></FONT></DIV>
      <DIV><FONT color=3D#800080><SPAN=20
      class=3D163370919-08112004></SPAN></FONT><FONT face=3DArial =
color=3D#800080=20
      size=3D2><SPAN =
class=3D163370919-08112004>[/MB]</SPAN></FONT></DIV><FONT=20
      face=3DArial><FONT size=3D2><FONT color=3D#808080><STRONG><SPAN=20
      =
class=3D436390008-09112004>[SG]&nbsp;</SPAN>&nbsp;</STRONG></FONT><SPAN =

      class=3D436390008-09112004><FONT =
color=3D#808080><STRONG>&nbsp;Changing the=20
      term&nbsp;"domain" to "resource" does not solve the problem I =
mentionned=20
      above. In order to reflect =
current&nbsp;operator&nbsp;requirements, the=20
      draft should not PRECLUDE the fowarding of existing history =
information=20
      towards trusted=20
      =
domains.</STRONG></FONT></SPAN></FONT></FONT></SPAN></FONT></SPAN></DIV>=

      <DIV><SPAN class=3D019105313-08112004></SPAN><SPAN=20
      class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff=20
      size=3D2></FONT></SPAN>&nbsp;</DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
      size=3D2>Thank you&nbsp;for&nbsp;clarifying&nbsp;this=20
      point.</FONT></SPAN></DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
      size=3D2></FONT></SPAN>&nbsp;</DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
      size=3D2>Best regards,</FONT></SPAN></DIV>
      <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial =
color=3D#0000ff=20
      size=3D2>s=E9bastien</FONT></SPAN></DIV>
      <DIV><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>&nbsp;</DIV>
      <DIV><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>&nbsp;</DIV>
      <DIV><BR><SPAN class=3D761145520-08112004><FONT face=3DTahoma=20
      size=3D2>------Remainder of non-related part of this thread has =
been deleted=20
      by=20
Mary------------------</FONT></SPAN></DIV></BLOCKQUOTE></BLOCKQUOTE></BL=
OCKQUOTE></BODY></HTML>

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Subject: RE: [Sip] PING/PONG
Date: Wed, 10 Nov 2004 13:55:17 -0000
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From: "Steve Langstaff" <steve.langstaff@citel.com>
To: "Christian Stredicke" <Christian.Stredicke@snom.de>,
        "Chris Boulton" <cboulton@ubiquity.net>,
        "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>,
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Fair enough.
=20
I must have missed the description of PING/PONG messages were for.
=20
I apologise.

--=20
Steve Langstaff.=20

-----Original Message-----
From: Christian Stredicke [mailto:Christian.Stredicke@snom.de]
Sent: 10 November 2004 10:36
To: Steve Langstaff; Chris Boulton; Christer Holmberg (JO/LMF); =
fluffy@cisco.com
Cc: sip@ietf.org
Subject: RE: [Sip] PING/PONG


True. For that purpose, use e.g. INFO messages with no body. These =
messages keep the dialog "alive".=20
=20
However, the goal of the PING/PONG keep-alive messages is to keep the =
NAT-binding alive. So that you are able to receive the first INVITE!
=20
CS


  _____ =20

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Steve Langstaff
Sent: Wednesday, November 10, 2004 5:16 AM
To: Christian Stredicke; Chris Boulton; Christer Holmberg (JO/LMF); =
fluffy@cisco.com
Cc: sip@ietf.org
Subject: RE: [Sip] PING/PONG


(CS) OPTIONS has the problem that it is relatively expensive (size and =
CPU) and might result in responses that exceed the UDP fragmentation =
limit (there are many DSL routers that treat UDP fragmentation as =
security problem including the one that I have at home). Therefore I =
think it would not hurt to have a specific message that is designed only =
for the keep-alive job.
=20
...but if you want your keep alive message to be routed over the same =
path and transports as your original INVITE (and hence keep them alive), =
don't you need most of the "expensive" content of (say) the OPTIONS =
message - e.g. the vias.

--=20
Steve Langstaff.=20


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<DIV><SPAN class=3D377085413-10112004><FONT face=3DArial color=3D#0000ff =
size=3D2>Fair=20
enough.</FONT></SPAN></DIV>
<DIV><SPAN class=3D377085413-10112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D377085413-10112004><FONT face=3DArial color=3D#0000ff =
size=3D2>I must=20
have missed the&nbsp;description of PING/PONG messages were=20
for.</FONT></SPAN></DIV>
<DIV><SPAN class=3D377085413-10112004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D377085413-10112004><FONT face=3DArial color=3D#0000ff =
size=3D2>I=20
apologise.</FONT></SPAN></DIV>
<P><FONT face=3DArial size=3D2>--</FONT> <BR><FONT face=3DArial =
size=3D2>Steve=20
Langstaff.</FONT> </P>
<DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
size=3D2>-----Original Message-----<BR><B>From:</B> Christian Stredicke=20
[mailto:Christian.Stredicke@snom.de]<BR><B>Sent:</B> 10 November 2004=20
10:36<BR><B>To:</B> Steve Langstaff; Chris Boulton; Christer Holmberg =
(JO/LMF);=20
fluffy@cisco.com<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> RE: [Sip] =

PING/PONG<BR><BR></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D548353310-10112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>True. For that purpose, use e.g. INFO messages =
with no=20
body. These messages keep the dialog "alive". </FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D548353310-10112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D548353310-10112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>However, the goal of the PING/PONG keep-alive =
messages is=20
to keep the NAT-binding alive. So that you are able to receive the first =

INVITE!</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D548353310-10112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D548353310-10112004><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>CS</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Steve=20
  Langstaff<BR><B>Sent:</B> Wednesday, November 10, 2004 5:16 =
AM<BR><B>To:</B>=20
  Christian Stredicke; Chris Boulton; Christer Holmberg (JO/LMF);=20
  fluffy@cisco.com<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> RE: =
[Sip]=20
  PING/PONG<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: navy; FONT-STYLE: =
italic; FONT-FAMILY: Arial"><SPAN=20
  class=3D546190215-09112004><FONT face=3DVerdana color=3D#0000ff>(CS) =
OPTIONS has the=20
  problem that it is relatively expensive (size and CPU) and might =
result in=20
  responses that exceed the UDP fragmentation limit (there are many DSL =
routers=20
  that treat UDP fragmentation as security problem including the one =
that I have=20
  at home). Therefore I think it would not hurt to have a specific =
message that=20
  is designed only for the keep-alive =
job.</FONT></SPAN></SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D563593009-10112004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>...but if you want your keep alive message to be routed over =
the same=20
  path and transports as your original INVITE (and hence keep them =
alive), don't=20
  you need most of the "expensive" content of (say) the =
OPTIONS&nbsp;message -=20
  e.g. the vias.</FONT></SPAN></DIV>
  <P><FONT face=3DArial size=3D2>--</FONT> <BR><FONT face=3DArial =
size=3D2>Steve=20
  Langstaff.</FONT> <FONT=20
style=3D"BACKGROUND-COLOR: =
#ffffff"></P></BLOCKQUOTE></FONT></BODY></HTML>

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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
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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'Jesske, R'" <R.Jesske@t-com.net>, sebastien.garcin@francetelecom.com,
        sip@ietf.org
Subject: RE: [Sip] Privacy statements and History (draft-ietf-sip-history-
	info-04.txt)
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Roland and Sebastien,
=20
I too agree with the proposal to change the strength of the referenced
statement from section 4.3.3.1.1  from MUST to SHOULD. This change is
consistent with the other normative statements in section 4.3.3.1.1, whic=
h
are SHOULDs rather than MUSTs.   I'll make that change in the next rev
unless someone raises concerns over that change.=20
=20
On the second concern, I'm not clear what the proposed change is for the
following:
[SG] Another issue is the decision to add hi-entries in a privacy context=
. A
proxy changing the target (i.e. the proxy is responsible of the resource
reflected in the received Request-URI) of a request which contained a
"privacy=3Dhistory" header MAY add a history-entries provided that it kno=
ws it
can rely on other entities within the trust domain to apply the requested
privacy. This affect item 4 in the list on conditions of section 4.3.3 an=
d
4.3.3.1.=20
=20
Item 4 in section 4.3.3 describes the general situation, thus the strengt=
h
(MAY) describes the general fact that the use of privacy is optional.  Th=
e
normative text provided in 4.3.3.1, provides the general model for adding
hi-entries, with the privacy specific considerations for adding the entri=
es
described in 4.3.3.1.1.  I think what you're suggesting is changing the
strength of the following statement in section 4.3.3.1:
" The hi-entry MUST be added following any hi-entry received in the reque=
st
being forwarded."
I can see that in the context of privacy considerations, this might seem
misleading.  That MUST was originally a SHOULD but was changed (as discus=
sed
on the list and at IETF-60) to MUST based on issue JRE- 4, so a simple
change of MUST to MAY would not at all work.  I thought it was clear from
the statement at the beginning of 4.3.3.1, that this section is describin=
g
the scenario under which the general privacy considerations had been
evaluated and the intention was to add an hi-entry, thus the statements
should be read in the context that the general screening indicates that a=
n
hi-entry SHOULD be added and if one is added it MUST be added following a=
ny
entry that's already in the request to preserve the ordering.   If you ha=
ve
explicit changes to the text that you think would further clarify the
functionality, we can discuss.=20
=20
Roland, if you have specific text on how I could incorporate a reference =
to
RFC3323, we can consider that; fundamentally any discussion of privacy
depends on RFC 3323, so it's not clear to me how you were suggesting we
incorporate such a reference.  The text in this document (history-info)
should accurately describe the processing impact of the new "history"
priv-value.=20
=20
Thanks for your input,
Mary=20


-----Original Message-----
From: Jesske, R [mailto:R.Jesske@t-com.net]=20
Sent: Wednesday, November 10, 2004 7:59 AM
To: sebastien.garcin@francetelecom.com; Barnes, Mary [NGC:B601:EXCH];
sip@ietf.org
Cc: VL-T-Com-T-TE332@vli.telekom.de
Subject: AW: [Sip] Privacy statements and History
(draft-ietf-sip-history-info-04.txt)


Dear Mary and Sebastien,
here are my Comments to Sebastien's statements:
=20
Your first proposal I can accept from my point of view.
On you second issue my impression is that this issue must be seen with
regard to RFC3323 where privacy and the trust concept is described. Perha=
ps
a reference to this document should be included.
On your last point, I think we can also refer to RFC 3323.
=20
What are you thinking?
=20
Best Regards
=20
Roland
=20

 ---- Mary snipped forwarding info to keep message smaller----------- =20



-----Original Message-----
From: GARCIN Sebastien RD-CORE-ISS
[mailto:sebastien.garcin@francetelecom.com]=20
Sent: Monday, November 08, 2004 10:11 AM
To: Barnes, Mary [NGC:B601:EXCH]; sip@ietf.org
Subject: [Sip] Privacy statements and History
(draft-ietf-sip-history-info-04.txt)


Hi mary, all
=20
When reading draft-ietf-sip-history-info-04.txt, I have trouble in
understanding some of the statements which relate to the forwarding rules
for history-entries subject to privacy. It is an important requirement th=
at
History-entrie(s) with a Privacy=3Dhistory, session, or header are indeed
forwarded to entities which belong to the same trust domain. The removal =
of
specific history-entries should only occur if the peer does not belong to
the trust domain.
=20
In the current text (section 4.3.3.1.1) :
If a request is  being forwarded to a Request URI associated with a domai=
n
for which the proxy is not responsible and there is a Privacy header in t=
he
request with a priv-value of "session", "header" or "history", the proxy
MUST remove any hi-entry(s) prior to forwarding.=20
=20
The current wording is misleading since it gives the impression (maybe
intentionnal) that it is not possible to forward history-entries with
Privacy statements to domains under the responsability of e.g. another
operator belonging to the same trust domain. =20
=20
[MB]: Current wording is consistent with terminology in RFC 3261 in terms=
 of
describing who is able to change the Request URI in a specific request
(based on section 16.5):=20
   " A proxy MUST NOT add additional targets to the target set if the
   Request-URI of the original request does not indicate a resource this
   proxy is responsible for.

      A proxy can only change the Request-URI of a request during
      forwarding if it is responsible for that URI. "
Since History-Info (and associated privacy) are only added to the request=
,
when an entity that is allowed to change the Request-URI retargets the
request, it seemed sensible to use consistent wording to explain that.  =20
[/MB]=20
 [SG] I have no problem with the statements above. My concern is that if
there is a hi-entry already embedded in the request with a Privacy
statement, then, it should be up to local policy to decide whether or not=
 a
proxy shall pass on those hi-entries to a trusted domain. The sentence in
section 4.3.3.1.1 precludes this. I would propose to lighten the strenght=
 of
the sentence as follows:
=20
If a request is  being forwarded to a Request URI associated with a domai=
n
for which the proxy is not responsible and there is a Privacy header in t=
he
request with a priv-value of "session", "header" or "history", the proxy =
MAY
remove any hi-entry(s) prior to forwarding.=20
=20
[SG] Another issue is the decision to add hi-entries in a privacy context=
. A
proxy changing the target (i.e. the proxy is responsible of the resource
reflected in the received Request-URI) of a request which contained a
"privacy=3Dhistory" header MAY add a history-entries provided that it kno=
ws it
can rely on other entities within the trust domain to apply the requested
privacy. This affect item 4 in the list on conditions of section 4.3.3 an=
d
4.3.3.1.=20
=20
The concept of "trust domain" should be used when discussing the forwardi=
ng
rules pertaining to information subject to privacy. Furthermore, the
requirement for forwarding history-entries to trusted entities should be
stated more clearly in the draft.=20

[MB]: The whole concept of what defines privacy in terms of the proxy's u=
se
of the privacy header is outside the scope of History-Info functionality =
and
really a matter of local policy.   I think the functionality that you wan=
t
is a matter of local implementation and policy in terms of operators
establishing this "trust domain" model to which you refer.  History-Info
defines the mechanism to ensure the privacy of the requests, but it doesn=
't
explicitly define how the proxy knows whether it is responsible for that
resource.    I don't think this is a matter of standardization.  I though=
t
the use of the term "domain" rather than "resouce" would be helpful, but
perhaps changing it to the more general "resource" would resolve this
concern and/or a statement clarifying what I've just described should be
added in the draft.=20
[/MB]
[SG]   Changing the term "domain" to "resource" does not solve the proble=
m I
mentionned above. In order to reflect current operator requirements, the
draft should not PRECLUDE the fowarding of existing history information
towards trusted domains.
=20
Thank you for clarifying this point.
=20
Best regards,
s=E9bastien
=20
=20

------Remainder of non-related part of this thread has been deleted by
Mary------------------


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; charset=3Diso-885=
9-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1476" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D160384614-=
10112004>Roland=20
and Sebastien,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D160384614-10112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D160384614-=
10112004>I too=20
agree with the proposal to change the strength of the referenced statemen=
t from=20
section&nbsp;4.3.3.1.1&nbsp; from MUST to SHOULD. This change is consiste=
nt with=20
the other normative statements in section 4.3.3.1.1, which are SHOULDs ra=
ther=20
than MUSTs.&nbsp; &nbsp;I'll make that change in the next rev unless some=
one=20
raises concerns over that change. </SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D160384614-10112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D160384614-=
10112004>On the=20
second concern, I'm not clear what the proposed change is for the=20
following:</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff><SPAN class=3D160384614-10112004>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D#808080 =
size=3D2><SPAN=20
class=3D436390008-09112004><STRONG>[SG]&nbsp;Another issue is the decisio=
n to add=20
hi-entries in a&nbsp;privacy context. A proxy changing the target (i.e. t=
he=20
proxy is responsible of the resource reflected in the received Request-UR=
I) of a=20
request which contained a "privacy=3Dhistory"&nbsp;header MAY add&nbsp;a=20
history-entries&nbsp;provided that&nbsp;it&nbsp;knows it can rely on othe=
r=20
entities within the&nbsp;trust domain to apply the requested privacy. Thi=
s=20
affect item 4 in the list on conditions of section 4.3.3 and 4.3.3.1.=20
</STRONG></SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D#808080 =
size=3D2><SPAN=20
class=3D436390008-09112004><STRONG></STRONG></SPAN></FONT></SPAN>&nbsp;</=
DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial size=3D2><SPAN=20
class=3D436390008-09112004><SPAN class=3D160384614-10112004>Item 4 in sec=
tion 4.3.3=20
describes the general situation, thus the strength (MAY) describes the ge=
neral=20
fact that the use of privacy is optional.&nbsp; The normative text provid=
ed in=20
4.3.3.1, provides the general model for adding hi-entries, with the priva=
cy=20
specific considerations for adding the entries described in 4.3.3.1.1.&nb=
sp; I=20
think what you're suggesting is changing the strength of&nbsp;the followi=
ng=20
statement in section 4.3.3.1:</SPAN></SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial size=3D2><SPAN=20
class=3D436390008-09112004><SPAN class=3D160384614-10112004>" The hi-entr=
y MUST be=20
added following any hi-entry received&nbsp;in the request being=20
forwarded."</SPAN></SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D#808080 =
size=3D2><SPAN=20
class=3D436390008-09112004><SPAN class=3D160384614-10112004><FONT color=3D=
#0000ff>I=20
can see that in the context of privacy considerations, this might seem=20
misleading.&nbsp; That MUST was originally a SHOULD but was changed (as=20
discussed on the list and at IETF-60) to MUST based on issue JRE-&nbsp;4,=
 so a=20
simple change of MUST to MAY would not at all work.&nbsp; I thought it wa=
s clear=20
from the statement at the beginning of 4.3.3.1, that this section is desc=
ribing=20
the scenario under which the general privacy considerations had been eval=
uated=20
and the intention was to add an hi-entry, thus the statements should be r=
ead in=20
the context that the general screening indicates that an hi-entry SHOULD =
be=20
added and if one is added it MUST be added following any entry that's alr=
eady in=20
the request to preserve the ordering. &nbsp; If you have explicit changes=
 to the=20
text that you think would further clarify the functionality, we can discu=
ss.=20
</FONT></SPAN></SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D436390008-09112004><SPAN=20
class=3D160384614-10112004></SPAN></SPAN></FONT></SPAN><FONT face=3DArial=
=20
size=3D2></FONT><FONT face=3DArial size=3D2></FONT></SPAN></FONT><FONT fa=
ce=3DArial=20
color=3D#0000ff size=3D2><SPAN=20
class=3D160384614-10112004></SPAN></FONT>&nbsp;</DIV></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D160384614-10112004>Roland, if you have specific text on how I cou=
ld=20
incorporate a reference to RFC3323, we can consider that; fundamentally a=
ny=20
discussion of privacy depends on RFC 3323, so it's not clear to me how yo=
u were=20
suggesting we incorporate such a reference.&nbsp; The text in this docume=
nt=20
(history-info) should accurately describe the processing impact of the ne=
w=20
"history" priv-value. </SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D160384614-10112004></SPAN></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D160384614-10112004><FONT face=3DArial color=3D#0000ff =
size=3D2>Thanks=20
for your input,</FONT></SPAN></DIV>
<DIV><FONT color=3D#0000ff><SPAN lang=3Den-us><FONT face=3DArial=20
size=3D2>Mary</FONT></SPAN> </FONT></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV><FONT color=3D#0000ff></FONT></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><=
FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> Jessk=
e, R=20
  [mailto:R.Jesske@t-com.net] <BR><B>Sent:</B> Wednesday, November 10, 20=
04 7:59=20
  AM<BR><B>To:</B> sebastien.garcin@francetelecom.com; Barnes, Mary=20
  [NGC:B601:EXCH]; sip@ietf.org<BR><B>Cc:</B>=20
  VL-T-Com-T-TE332@vli.telekom.de<BR><B>Subject:</B> AW: [Sip] Privacy=20
  statements and History=20
  (draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f size=3D2>Dear=20
  Mary and Sebastien,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f size=3D2>here=20
  are my Comments to Sebastien's statements:</FONT></SPAN></DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f size=3D2>Your=20
  first proposal I can accept from my point of view.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f size=3D2>On=20
  you second issue my impression is that this issue must be seen with reg=
ard to=20
  RFC3323 where privacy and the trust concept is described. Perhaps a ref=
erence=20
  to this document should be included.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f size=3D2>On=20
  your last point, I think we can also refer to RFC 3323.</FONT></SPAN></=
DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f=20
  size=3D2>What&nbsp;are you thinking?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f size=3D2>Best=20
  Regards</FONT></SPAN></DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f=20
  size=3D2>Roland</FONT></SPAN></DIV>
  <DIV><SPAN class=3D578113713-10112004><FONT face=3DArial color=3D#0000f=
f=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT face=3D=
Arial=20
    color=3D#0000ff size=3D2><SPAN class=3D160384614-10112004>&nbsp;---- =
Mary snipped=20
    forwarding info to keep message smaller-----------&nbsp;</SPAN></FONT=
>
    <DIV><FONT face=3DTahoma size=3D2></FONT></DIV></DIV>
    <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
      <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
        <DIV><FONT color=3D#0000ff></FONT></DIV>
        <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3D=
left><FONT=20
        face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=
 GARCIN=20
        Sebastien RD-CORE-ISS [mailto:sebastien.garcin@francetelecom.com]=
=20
        <BR><B>Sent:</B> Monday, November 08, 2004 10:11 AM<BR><B>To:</B>=
=20
        Barnes, Mary [NGC:B601:EXCH]; sip@ietf.org<BR><B>Subject:</B> [Si=
p]=20
        Privacy statements and History=20
        (draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
        <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff si=
ze=3D2><SPAN=20
        class=3D019105313-08112004>Hi mary, all</SPAN></FONT></DIV>
        <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff si=
ze=3D2><SPAN=20
        class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
        <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff si=
ze=3D2><SPAN=20
        class=3D019105313-08112004>When=20
        reading&nbsp;draft-ietf-sip-history-info-04.txt, I&nbsp;have trou=
ble in=20
        understanding some of the statements which relate to the forwardi=
ng=20
        rules&nbsp;for history-entries subject to=20
        privacy.&nbsp;</SPAN></FONT><FONT face=3DArial color=3D#0000ff si=
ze=3D2><SPAN=20
        class=3D019105313-08112004>It is an&nbsp;important=20
        requirement&nbsp;that&nbsp;History-entrie(s) with&nbsp;a=20
        Privacy=3Dhistory, session, or&nbsp;header are indeed forwarded&n=
bsp;to=20
        entities which belong&nbsp;to&nbsp;the same&nbsp;trust domain. Th=
e=20
        removal of specific history-entries should only occur&nbsp;if the=
 peer=20
        does not belong to the trust domain.</SPAN></FONT></DIV>
        <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff si=
ze=3D2><SPAN=20
        class=3D019105313-08112004></SPAN></FONT>&nbsp;</DIV>
        <DIV dir=3Dltr align=3Dleft><FONT face=3DArial><SPAN=20
        class=3D019105313-08112004><FONT size=3D2><FONT color=3D#0000ff>I=
n the current=20
        text (section 4.3.3.1.1)&nbsp;:</FONT></FONT></SPAN></FONT></DIV>
        <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><FONT fac=
e=3DArial=20
        color=3D#0000ff size=3D2></FONT>If a request is&nbsp; being forwa=
rded to a=20
        Request URI associated with <STRONG>a domain for which&nbsp;the p=
roxy is=20
        not responsible</STRONG> and there is a Privacy header in=20
        the&nbsp;request&nbsp;<SPAN class=3D019105313-08112004>w</SPAN>it=
h a=20
        priv-value of "session", "header" or "history", the&nbsp;proxy MU=
ST=20
        remove any hi-entry(s) prior to forwarding. </DIV>
        <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</D=
IV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
        color=3D#0000ff><FONT size=3D2>The&nbsp;current&nbsp;wording is m=
isleading=20
        since it gives the impression (maybe intentionnal) that it is not=
=20
        possible to forward history-entries with Privacy statements to do=
mains=20
        under the responsability of&nbsp;e.g. another operator belonging =
to the=20
        same trust domain.&nbsp;<SPAN=20
        class=3D163370919-08112004>&nbsp;</SPAN></FONT></FONT></FONT></SP=
AN></DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
        color=3D#0000ff><FONT size=3D2><SPAN=20
        class=3D163370919-08112004></SPAN></FONT></FONT></FONT></SPAN>&nb=
sp;</DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT si=
ze=3D+0><FONT=20
        color=3D#800080 size=3D2><SPAN class=3D163370919-08112004>[MB]: C=
urrent=20
        wording is consistent with terminology in RFC 3261 in terms of=20
        describing who is able to&nbsp;change the Request URI in a specif=
ic=20
        request (based on&nbsp;section=20
        16.5):&nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
        color=3D#800080><SPAN class=3D163370919-08112004>&nbsp;&nbsp; " A=
 proxy MUST=20
        NOT add additional targets to the target set if the<BR>&nbsp;&nbs=
p;=20
        Request-URI of the original request does not indicate a resource=20
        this<BR>&nbsp;&nbsp; proxy is responsible=20
        for.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; A proxy can only chang=
e the=20
        Request-URI of a request during<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
=20
        forwarding if it is responsible for that URI.=20
        "</SPAN></FONT></FONT></SPAN></DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT si=
ze=3D+0><FONT=20
        color=3D#800080 size=3D2><SPAN class=3D163370919-08112004>Since H=
istory-Info=20
        (and associated privacy)&nbsp;are only added to the request,&nbsp=
;when=20
        an&nbsp;entity that is allowed to change&nbsp;the Request-URI ret=
argets=20
        the request, it seemed sensible to use consistent wording to expl=
ain=20
        that.&nbsp; &nbsp;</SPAN></FONT></FONT></FONT></SPAN></DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT color=3D#800080><SPAN=
=20
        class=3D163370919-08112004><FONT face=3DArial><FONT size=3D2>[/MB=
]<SPAN=20
        class=3D436390008-09112004><FONT=20
        color=3D#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></SPAN></FONT><=
/SPAN></DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT color=3D#800080><SPAN=
=20
        class=3D163370919-08112004><FONT face=3DArial><FONT size=3D2><SPA=
N=20
        class=3D436390008-09112004>&nbsp;</SPAN></FONT></FONT></SPAN></FO=
NT></SPAN><SPAN=20
        class=3D019105313-08112004><FONT face=3DArial color=3D#800080 siz=
e=3D2><SPAN=20
        class=3D436390008-09112004><STRONG><FONT color=3D#808080>[SG] I h=
ave no=20
        problem with the statements above.&nbsp;My concern is that if the=
re is=20
        a&nbsp;hi-entry&nbsp;<U>already</U> embedded in the request with =
a=20
        Privacy statement, then, it should be&nbsp;up to local policy to =
decide=20
        whether or not&nbsp;a proxy shall pass on those hi-entries to a t=
rusted=20
        domain. The sentence in section 4.3.3.1.1 precludes this.&nbsp;I =
would=20
        propose to lighten the strenght of the sentence as=20
        follows:</FONT></STRONG></SPAN></FONT></SPAN></DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D=
#808080=20
        size=3D2><SPAN=20
        class=3D436390008-09112004><STRONG></STRONG></SPAN></FONT></SPAN>=
&nbsp;</DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT size=3D+0><SPAN=20
        class=3D436390008-09112004>If a request is&nbsp; being forwarded =
to a=20
        Request URI associated with <STRONG>a domain for which&nbsp;the p=
roxy is=20
        not responsible</STRONG> and there is a Privacy header in=20
        the&nbsp;request&nbsp;<SPAN class=3D019105313-08112004>w</SPAN>it=
h a=20
        priv-value of "session", "header" or "history", the&nbsp;proxy&nb=
sp;MAY=20
        remove any hi-entry(s) prior to forwarding. </SPAN></FONT></SPAN>=
</DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D=
#808080=20
        size=3D2><SPAN class=3D436390008-09112004></SPAN></FONT></SPAN><S=
PAN=20
        class=3D019105313-08112004><FONT face=3DArial color=3D#808080 siz=
e=3D2><SPAN=20
        class=3D436390008-09112004><STRONG></STRONG></SPAN></FONT></SPAN>=
&nbsp;</DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D=
#808080=20
        size=3D2><SPAN class=3D436390008-09112004><STRONG>[SG]&nbsp;Anoth=
er issue is=20
        the decision to add hi-entries in a&nbsp;privacy context. A proxy=
=20
        changing the target (i.e. the proxy is responsible of the resourc=
e=20
        reflected in the received Request-URI) of a request which contain=
ed a=20
        "privacy=3Dhistory"&nbsp;header MAY add&nbsp;a=20
        history-entries&nbsp;provided that&nbsp;it&nbsp;knows it can rely=
 on=20
        other entities within the&nbsp;trust domain to apply the requeste=
d=20
        privacy. This affect item 4 in the list on conditions of section =
4.3.3=20
        and 4.3.3.1. </STRONG></SPAN></FONT></SPAN></DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><SPAN=20
        class=3D436390008-09112004></SPAN></FONT></SPAN>&nbsp;</DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial><FONT=20
        color=3D#0000ff><FONT size=3D2>The&nbsp;concept of "trust=20
        domain"&nbsp;should be&nbsp;used when discussing the&nbsp;forward=
ing=20
        rules pertaining&nbsp;to information subject to privacy.=20
        Furthermore,&nbsp;the requirement for forwarding history-entries =
to=20
        trusted entities should be&nbsp;stated more clearly in the draft.=
<SPAN=20
        class=3D163370919-08112004>&nbsp;</SPAN></FONT></FONT></FONT></SP=
AN></DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT color=3D#0000ff><SPAN=
=20
        class=3D163370919-08112004>
        <DIV><FONT face=3DArial color=3D#800080 size=3D2><SPAN=20
        class=3D163370919-08112004>[MB]: The whole concept of what define=
s privacy=20
        in terms of the proxy's use of the privacy header is outside the =
scope=20
        of History-Info functionality and really a matter of local policy=
.=20
        &nbsp; I think the functionality that you want is a matter of loc=
al=20
        implementation and policy in terms of&nbsp;operators establishing=
 this=20
        "trust domain" model&nbsp;to which you refer.&nbsp; History-Info =
defines=20
        the mechanism to&nbsp;ensure the privacy of the requests, but it=20
        doesn't&nbsp;explicitly define how the&nbsp;proxy knows whether i=
t is=20
        responsible for that resource.&nbsp;&nbsp;&nbsp;&nbsp;I don't thi=
nk this=20
        is a matter of standardization.&nbsp; I thought the use of the te=
rm=20
        "domain" rather than "resouce" would be helpful, but perhaps chan=
ging it=20
        to the more general "resource" would resolve this concern and/or =
a=20
        statement clarifying what I've just described should be added in =
the=20
        draft. </SPAN></FONT></DIV>
        <DIV><FONT color=3D#800080><SPAN=20
        class=3D163370919-08112004></SPAN></FONT><FONT face=3DArial color=
=3D#800080=20
        size=3D2><SPAN class=3D163370919-08112004>[/MB]</SPAN></FONT></DI=
V><FONT=20
        face=3DArial><FONT size=3D2><FONT color=3D#808080><STRONG><SPAN=20
        class=3D436390008-09112004>[SG]&nbsp;</SPAN>&nbsp;</STRONG></FONT=
><SPAN=20
        class=3D436390008-09112004><FONT color=3D#808080><STRONG>&nbsp;Ch=
anging the=20
        term&nbsp;"domain" to "resource" does not solve the problem I men=
tionned=20
        above. In order to reflect current&nbsp;operator&nbsp;requirement=
s, the=20
        draft should not PRECLUDE the fowarding of existing history infor=
mation=20
        towards trusted=20
        domains.</STRONG></FONT></SPAN></FONT></FONT></SPAN></FONT></SPAN=
></DIV>
        <DIV><SPAN class=3D019105313-08112004></SPAN><SPAN=20
        class=3D019105313-08112004><FONT face=3DArial color=3D#0000ff=20
        size=3D2></FONT></SPAN>&nbsp;</DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D=
#0000ff=20
        size=3D2>Thank you&nbsp;for&nbsp;clarifying&nbsp;this=20
        point.</FONT></SPAN></DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D=
#0000ff=20
        size=3D2></FONT></SPAN>&nbsp;</DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D=
#0000ff=20
        size=3D2>Best regards,</FONT></SPAN></DIV>
        <DIV><SPAN class=3D019105313-08112004><FONT face=3DArial color=3D=
#0000ff=20
        size=3D2>s=E9bastien</FONT></SPAN></DIV>
        <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</D=
IV>
        <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</D=
IV>
        <DIV><BR><SPAN class=3D761145520-08112004><FONT face=3DTahoma=20
        size=3D2>------Remainder of non-related part of this thread has b=
een=20
        deleted by=20
    Mary------------------</FONT></SPAN></DIV></BLOCKQUOTE></BLOCKQUOTE><=
/BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org  Wed Nov 10 10:59:12 2004
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Subject: [Sip] More general question on  S-S-L mode in RPH
To: "James M. Polk" <jmpolk@cisco.com>
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What do the S-S-L modes apply to?

In this case, I am just looking just at the dimension of which messages are
accepted, and which are rejected- ignoring the other features which have
been associated with the mode.

4.3.2 says that "mode" is a characteristic of the SIP element:

"When handling requests with unknown namespaces or priority values,
   elements can operate in one of three modes, "strict", "semi-strict"
   and "loose"."

This makes some sense to me (comments below).

There are other places that imply that the mode is a characteristic of the
domain.  4.3.3 says:

"Strict Mode is for administrative domains (or realms) that ..."

But table 1 in section 7 says:

"                                            New   New Error
Namespace  Levels   Mode    Operation   Rej. code  code    Reference
---------  ------   ----    ---------   ---------  ----    ---------
   dsn       5     strict   preemption      no      417    [This RFC]

   drsn      6     strict   preemption      no      417    [This RFC]

   q735      5     strict   preemption      no      417    [This RFC]

   ets       5  semi-strict preferential    no      417    [This RFC]

   wps       5  semi-strict preferential    no      417    [This RFC]

                   Table 1. Namespace Parameters"

Which makes "mode" a characteristic of the namespace.

This makes less sense to me.  If any of these namespaces appear in an RPH
in a SIP message which is being processed by a SIP element operating in
loose mode, they will NOT get the strict or semi-strict behavior specified
in Table 1.

It seems to me that the mode is/should be a function of the (SIP element,
namespace) pair.

For instance a SIP element could be "strict" for namespace A and
"semi-strict" for namespace B.

So a message with just A would succeed
A message with just B would be rejected
A message with A and B would succeed
A message with just C would  be rejected
A message with A and C would be rejected

Or the SIP element could be "strict" for namespace A, and "loose" for
everything else.

So a message with just A would succeed
A message with just B would be rejected
A message with A and B would succeed
A message with just C would  be rejected
A message with A and C would be succeed.

I don't think you could combine "semi-strict" and "loose" in the same SIP
element.

Am I missing something?

Janet




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Date: Wed, 10 Nov 2004 10:43:00 -0600
To: Janet P Gunn <jgunn6@csc.com>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] More general question on  S-S-L mode in RPH
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Clearly this document needs to address the behaviors you want in a mode (if 
they remain in the document), and clearly semi-strict as defined is not 
exactly as you want it.

Are you missing something...?

Domains will operate their network as they see fit, and this document will 
not dictate how that is accomplished. This is a document that describes a 
header with several field values. Some field values can be used to prevent 
communications in known networks, so the expected behaviors of those field 
values should be articulated somewhere in a document where the header is 
defined (the IETF).

A concern is 2 years from now someone operating a network will want to use 
this header in their network and want guidance as to which field values to 
use from a list of predefined values, or suggest a new one be created for 
unique conditions.

Since SIP is an end-to-end protocol (not accounting for B2BUAs), a SIP 
message with this header may have one behavior in one network and a 
completely different (and potentially undesirable) behavior in another 
network.

The modes articulated in this document are to explain usages that "can" 
(not must) be experienced in some networks when using specific namespaces 
created for specific purposes. Each are as much to caution as they are to 
define the behavior of usage.

If a namespace has a general characteristic of preemption as a behavior in 
a known network(s) - that namespace should not be used by endpoints in 
another network that does not include the behavior of preemption. 
Undesirable consequences will occur when communicated between those networks.

This is all guidance, it's not a law anywhere


At 10:39 AM 11/10/2004 -0500, Janet P Gunn wrote:

>What do the S-S-L modes apply to?
>
>In this case, I am just looking just at the dimension of which messages are
>accepted, and which are rejected- ignoring the other features which have
>been associated with the mode.
>
>4.3.2 says that "mode" is a characteristic of the SIP element:
>
>"When handling requests with unknown namespaces or priority values,
>    elements can operate in one of three modes, "strict", "semi-strict"
>    and "loose"."
>
>This makes some sense to me (comments below).
>
>There are other places that imply that the mode is a characteristic of the
>domain.  4.3.3 says:
>
>"Strict Mode is for administrative domains (or realms) that ..."
>
>But table 1 in section 7 says:
>
>"                                            New   New Error
>Namespace  Levels   Mode    Operation   Rej. code  code    Reference
>---------  ------   ----    ---------   ---------  ----    ---------
>    dsn       5     strict   preemption      no      417    [This RFC]
>
>    drsn      6     strict   preemption      no      417    [This RFC]
>
>    q735      5     strict   preemption      no      417    [This RFC]
>
>    ets       5  semi-strict preferential    no      417    [This RFC]
>
>    wps       5  semi-strict preferential    no      417    [This RFC]
>
>                    Table 1. Namespace Parameters"
>
>Which makes "mode" a characteristic of the namespace.
>
>This makes less sense to me.  If any of these namespaces appear in an RPH
>in a SIP message which is being processed by a SIP element operating in
>loose mode, they will NOT get the strict or semi-strict behavior specified
>in Table 1.
>
>It seems to me that the mode is/should be a function of the (SIP element,
>namespace) pair.
>
>For instance a SIP element could be "strict" for namespace A and
>"semi-strict" for namespace B.
>
>So a message with just A would succeed
>A message with just B would be rejected
>A message with A and B would succeed
>A message with just C would  be rejected
>A message with A and C would be rejected
>
>Or the SIP element could be "strict" for namespace A, and "loose" for
>everything else.
>
>So a message with just A would succeed
>A message with just B would be rejected
>A message with A and B would succeed
>A message with just C would  be rejected
>A message with A and C would be succeed.
>
>I don't think you could combine "semi-strict" and "loose" in the same SIP
>element.
>
>Am I missing something?
>
>Janet
>
>
>
>
>----------------------------------------------------------------------------------------
>
>This is a PRIVATE message. If you are not the intended recipient, please
>delete without copying and kindly advise us by e-mail of the mistake in
>delivery. NOTE: Regardless of content, this e-mail shall not operate to
>bind CSC to any order or other contract unless pursuant to explicit written
>agreement or government initiative expressly permitting the use of e-mail
>for such purpose.
>----------------------------------------------------------------------------------------


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


_______________________________________________
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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Subject: Re: [Sip] More general question on  S-S-L mode in RPH
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In a message dated 11/10/2004 11:00:13 AM Eastern Standard Time, 
jgunn6@csc.com writes:


> For instance a SIP element could be "strict" for namespace A and
> "semi-strict" for namespace B.
> 


This doesn't make sense. Part of the use of the mode still is to determine 
what to do when the receiving element does not understand the namespace, that 
is, it thinks it needs a namespace but doesn't see one it understands. (This was 
the only use of the mode in version -04.) Becasue of this, the mode must be a 
function of the domain, or rather, of every element in the domain. It is not 
associated with a namespace. Each element must know what mode it is supposed 
to operate in independent of what namespace is being used.

Version -04 was based on an element or UAS operating in a specific mode and 
did not try to associate the mode with the namespace. This technical change is 
another reason why I believe that version -05 should be rejected.

Mike


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/10/2004 11:00:13 AM Eastern Standard Time, jgunn6@csc.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">For instance a SIP element=20=
could be "strict" for namespace A and
<BR>"semi-strict" for namespace B.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>This doesn't make sense. Part of the use of the mode still is to determi=
ne what to do when the receiving element does not understand the namespace,=20=
that is, it thinks it needs a namespace but doesn't see one it understands.=20=
(This was the only use of the mode in version -04.) Becasue of this, the mod=
e must be a function of the domain, or rather, of every element in the domai=
n. It is not associated with a namespace. Each element must know what mode i=
t is supposed to operate in independent of what namespace is being used.
<BR>
<BR>Version -04 was based on an element or UAS operating in a specific mode=20=
and did not try to associate the mode with the namespace. This technical cha=
nge is another reason why I believe that version -05 should be rejected.
<BR>
<BR>Mike
<BR></FONT></HTML>

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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
--===============1527209579==--



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Subject: RE: [Sip] Privacy statements and History
	(draft-ietf-sip-history-info-04.txt)
Date: Wed, 10 Nov 2004 19:27:34 +0100
Message-ID: <49E7012A614B024B80A7D175CB9A64EC8A0B6E@ftrdmel1.rd.francetelecom.fr>
Thread-Topic: [Sip] Privacy statements and History
	(draft-ietf-sip-history-info-04.txt)
Thread-Index: AcTHORYP2JVBhFc9QaK//OB7bdk51AACvm3A
From: "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>
To: "Mary Barnes" <mary.barnes@nortelnetworks.com>,
        "Jesske, R" <R.Jesske@t-com.net>, <sip@ietf.org>
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Mary and roland,

Actually, the statement regarding the forwarding of hi-entries beyond =
the domain for which the proxy is responsible should rather be a matter =
of local policy and thus a MAY should be used instead of SHOULD. We are =
potentially dealing with network boundaries where agreements for =
forwarding such kind information can be reached.=20

Section 4.3.3.1.1
This section should be re-reworded in accordance with the statement =
above (I can provide some text is we can agree).

Section 4.3.3.1 is ok (apologies for not being explicit)

Best regards,
s=E9bastien



________________________________

De : Mary Barnes [mailto:mary.barnes@nortelnetworks.com]=20
Envoy=E9 : mercredi 10 novembre 2004 16:21
=C0 : 'Jesske, R'; GARCIN Sebastien RD-CORE-ISS; sip@ietf.org
Cc : VL-T-Com-T-TE332@vli.telekom.de
Objet : RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)


Roland and Sebastien,
=20
I too agree with the proposal to change the strength of the referenced =
statement from section 4.3.3.1.1  from MUST to SHOULD. This change is =
consistent with the other normative statements in section 4.3.3.1.1, =
which are SHOULDs rather than MUSTs.   I'll make that change in the next =
rev unless someone raises concerns over that change.=20
=20
On the second concern, I'm not clear what the proposed change is for the =
following:
[SG] Another issue is the decision to add hi-entries in a privacy =
context. A proxy changing the target (i.e. the proxy is responsible of =
the resource reflected in the received Request-URI) of a request which =
contained a "privacy=3Dhistory" header MAY add a history-entries =
provided that it knows it can rely on other entities within the trust =
domain to apply the requested privacy. This affect item 4 in the list on =
conditions of section 4.3.3 and 4.3.3.1.=20
=20
Item 4 in section 4.3.3 describes the general situation, thus the =
strength (MAY) describes the general fact that the use of privacy is =
optional.  The normative text provided in 4.3.3.1, provides the general =
model for adding hi-entries, with the privacy specific considerations =
for adding the entries described in 4.3.3.1.1.  I think what you're =
suggesting is changing the strength of the following statement in =
section 4.3.3.1:
" The hi-entry MUST be added following any hi-entry received in the =
request being forwarded."
I can see that in the context of privacy considerations, this might seem =
misleading.  That MUST was originally a SHOULD but was changed (as =
discussed on the list and at IETF-60) to MUST based on issue JRE- 4, so =
a simple change of MUST to MAY would not at all work.  I thought it was =
clear from the statement at the beginning of 4.3.3.1, that this section =
is describing the scenario under which the general privacy =
considerations had been evaluated and the intention was to add an =
hi-entry, thus the statements should be read in the context that the =
general screening indicates that an hi-entry SHOULD be added and if one =
is added it MUST be added following any entry that's already in the =
request to preserve the ordering.   If you have explicit changes to the =
text that you think would further clarify the functionality, we can =
discuss.=20
=20
Roland, if you have specific text on how I could incorporate a reference =
to RFC3323, we can consider that; fundamentally any discussion of =
privacy depends on RFC 3323, so it's not clear to me how you were =
suggesting we incorporate such a reference.  The text in this document =
(history-info) should accurately describe the processing impact of the =
new "history" priv-value.=20
=20
Thanks for your input,
Mary=20

		-----Original Message-----
	From: Jesske, R [mailto:R.Jesske@t-com.net]=20
	Sent: Wednesday, November 10, 2004 7:59 AM
	To: sebastien.garcin@francetelecom.com; Barnes, Mary [NGC:B601:EXCH]; =
sip@ietf.org
	Cc: VL-T-Com-T-TE332@vli.telekom.de
	Subject: AW: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)
=09
=09
	Dear Mary and Sebastien,
	here are my Comments to Sebastien's statements:
	=20
	Your first proposal I can accept from my point of view.
	On you second issue my impression is that this issue must be seen with =
regard to RFC3323 where privacy and the trust concept is described. =
Perhaps a reference to this document should be included.
	On your last point, I think we can also refer to RFC 3323.
	=20
	What are you thinking?
	=20
	Best Regards
	=20
	Roland
	=20

		 ---- Mary snipped forwarding info to keep message smaller-----------  =

	=09
								-----Original Message-----
				From: GARCIN Sebastien RD-CORE-ISS =
[mailto:sebastien.garcin@francetelecom.com]=20
				Sent: Monday, November 08, 2004 10:11 AM
				To: Barnes, Mary [NGC:B601:EXCH]; sip@ietf.org
				Subject: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)
			=09
			=09
				Hi mary, all
				=20
				When reading draft-ietf-sip-history-info-04.txt, I have trouble in =
understanding some of the statements which relate to the forwarding =
rules for history-entries subject to privacy. It is an important =
requirement that History-entrie(s) with a Privacy=3Dhistory, session, or =
header are indeed forwarded to entities which belong to the same trust =
domain. The removal of specific history-entries should only occur if the =
peer does not belong to the trust domain.
				=20
				In the current text (section 4.3.3.1.1) :
				If a request is  being forwarded to a Request URI associated with a =
domain for which the proxy is not responsible and there is a Privacy =
header in the request with a priv-value of "session", "header" or =
"history", the proxy MUST remove any hi-entry(s) prior to forwarding.=20
				=20
				The current wording is misleading since it gives the impression =
(maybe intentionnal) that it is not possible to forward history-entries =
with Privacy statements to domains under the responsability of e.g. =
another operator belonging to the same trust domain. =20
				=20
				[MB]: Current wording is consistent with terminology in RFC 3261 in =
terms of describing who is able to change the Request URI in a specific =
request (based on section 16.5):=20
				   " A proxy MUST NOT add additional targets to the target set if =
the
				   Request-URI of the original request does not indicate a resource =
this
				   proxy is responsible for.
			=09
				      A proxy can only change the Request-URI of a request during
				      forwarding if it is responsible for that URI. "
				Since History-Info (and associated privacy) are only added to the =
request, when an entity that is allowed to change the Request-URI =
retargets the request, it seemed sensible to use consistent wording to =
explain that.  =20
				[/MB]=20
				 [SG] I have no problem with the statements above. My concern is =
that if there is a hi-entry already embedded in the request with a =
Privacy statement, then, it should be up to local policy to decide =
whether or not a proxy shall pass on those hi-entries to a trusted =
domain. The sentence in section 4.3.3.1.1 precludes this. I would =
propose to lighten the strenght of the sentence as follows:
				=20
				If a request is  being forwarded to a Request URI associated with a =
domain for which the proxy is not responsible and there is a Privacy =
header in the request with a priv-value of "session", "header" or =
"history", the proxy MAY remove any hi-entry(s) prior to forwarding.=20
				=20
				[SG] Another issue is the decision to add hi-entries in a privacy =
context. A proxy changing the target (i.e. the proxy is responsible of =
the resource reflected in the received Request-URI) of a request which =
contained a "privacy=3Dhistory" header MAY add a history-entries =
provided that it knows it can rely on other entities within the trust =
domain to apply the requested privacy. This affect item 4 in the list on =
conditions of section 4.3.3 and 4.3.3.1.=20
				=20
				The concept of "trust domain" should be used when discussing the =
forwarding rules pertaining to information subject to privacy. =
Furthermore, the requirement for forwarding history-entries to trusted =
entities should be stated more clearly in the draft.=20
								[MB]: The whole concept of what defines privacy in terms of the =
proxy's use of the privacy header is outside the scope of History-Info =
functionality and really a matter of local policy.   I think the =
functionality that you want is a matter of local implementation and =
policy in terms of operators establishing this "trust domain" model to =
which you refer.  History-Info defines the mechanism to ensure the =
privacy of the requests, but it doesn't explicitly define how the proxy =
knows whether it is responsible for that resource.    I don't think this =
is a matter of standardization.  I thought the use of the term "domain" =
rather than "resouce" would be helpful, but perhaps changing it to the =
more general "resource" would resolve this concern and/or a statement =
clarifying what I've just described should be added in the draft.=20
				[/MB]
				[SG]   Changing the term "domain" to "resource" does not solve the =
problem I mentionned above. In order to reflect current operator =
requirements, the draft should not PRECLUDE the fowarding of existing =
history information towards trusted domains.
				=20
				Thank you for clarifying this point.
				=20
				Best regards,
				s=E9bastien
				=20
				=20

				------Remainder of non-related part of this thread has been deleted =
by Mary------------------


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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'GARCIN Sebastien RD-CORE-ISS'" <sebastien.garcin@francetelecom.com>,
        "Jesske, R" <R.Jesske@t-com.net>, sip@ietf.org
Subject: RE: [Sip] Privacy statements and History (draft-ietf-sip-history-
	info-04.txt)
Date: Wed, 10 Nov 2004 13:53:53 -0500
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I still contend that the SHOULD is sufficient.  SHOULD means that in =
general
processing, if the hi-entry has a privacy header, then the usual =
processing
would be that it would be removed (i.e it was added for a specific =
reason
and in general should be used to remove the entries based on well =
defined
criteria).  If there are reasons, such as local policy, that would =
allow the
forwarding in specific cases, then it's okay that it is forwarded.  I =
think
the use of MAY results in less precision and I think the value and =
critera
for associating and removing the privacy header with the hi-entry =
becomes
much less clear. =20

I'd like to hear more opinions on this topic, prior to agreeing to =
making
the change (from the MUST to MAY rather than MUST to SHOULD).=20

Regards,=20
Mary


-----Original Message-----
From: GARCIN Sebastien RD-CORE-ISS
[mailto:sebastien.garcin@francetelecom.com]=20
Sent: Wednesday, November 10, 2004 12:28 PM
To: Barnes, Mary [NGC:B601:EXCH]; Jesske, R; sip@ietf.org
Cc: VL-T-Com-T-TE332@vli.telekom.de
Subject: RE: [Sip] Privacy statements and History
(draft-ietf-sip-history-info-04.txt)


Mary and roland,

Actually, the statement regarding the forwarding of hi-entries beyond =
the
domain for which the proxy is responsible should rather be a matter of =
local
policy and thus a MAY should be used instead of SHOULD. We are =
potentially
dealing with network boundaries where agreements for forwarding such =
kind
information can be reached.=20

Section 4.3.3.1.1
This section should be re-reworded in accordance with the statement =
above (I
can provide some text is we can agree).

Section 4.3.3.1 is ok (apologies for not being explicit)

Best regards,
s=E9bastien



________________________________

De : Mary Barnes [mailto:mary.barnes@nortelnetworks.com]=20
Envoy=E9 : mercredi 10 novembre 2004 16:21
=C0 : 'Jesske, R'; GARCIN Sebastien RD-CORE-ISS; sip@ietf.org
Cc : VL-T-Com-T-TE332@vli.telekom.de
Objet : RE: [Sip] Privacy statements and History
(draft-ietf-sip-history-info-04.txt)


Roland and Sebastien,
=20
I too agree with the proposal to change the strength of the referenced
statement from section 4.3.3.1.1  from MUST to SHOULD. This change is
consistent with the other normative statements in section 4.3.3.1.1, =
which
are SHOULDs rather than MUSTs.   I'll make that change in the next rev
unless someone raises concerns over that change.=20
=20
On the second concern, I'm not clear what the proposed change is for =
the
following:
[SG] Another issue is the decision to add hi-entries in a privacy =
context. A
proxy changing the target (i.e. the proxy is responsible of the =
resource
reflected in the received Request-URI) of a request which contained a
"privacy=3Dhistory" header MAY add a history-entries provided that it =
knows it
can rely on other entities within the trust domain to apply the =
requested
privacy. This affect item 4 in the list on conditions of section 4.3.3 =
and
4.3.3.1.=20
=20
Item 4 in section 4.3.3 describes the general situation, thus the =
strength
(MAY) describes the general fact that the use of privacy is optional.  =
The
normative text provided in 4.3.3.1, provides the general model for =
adding
hi-entries, with the privacy specific considerations for adding the =
entries
described in 4.3.3.1.1.  I think what you're suggesting is changing the
strength of the following statement in section 4.3.3.1:
" The hi-entry MUST be added following any hi-entry received in the =
request
being forwarded."
I can see that in the context of privacy considerations, this might =
seem
misleading.  That MUST was originally a SHOULD but was changed (as =
discussed
on the list and at IETF-60) to MUST based on issue JRE- 4, so a simple
change of MUST to MAY would not at all work.  I thought it was clear =
from
the statement at the beginning of 4.3.3.1, that this section is =
describing
the scenario under which the general privacy considerations had been
evaluated and the intention was to add an hi-entry, thus the statements
should be read in the context that the general screening indicates that =
an
hi-entry SHOULD be added and if one is added it MUST be added following =
any
entry that's already in the request to preserve the ordering.   If you =
have
explicit changes to the text that you think would further clarify the
functionality, we can discuss.=20
=20
Roland, if you have specific text on how I could incorporate a =
reference to
RFC3323, we can consider that; fundamentally any discussion of privacy
depends on RFC 3323, so it's not clear to me how you were suggesting we
incorporate such a reference.  The text in this document (history-info)
should accurately describe the processing impact of the new "history"
priv-value.=20
=20
Thanks for your input,
Mary=20

		-----Original Message-----
	From: Jesske, R [mailto:R.Jesske@t-com.net]=20
	Sent: Wednesday, November 10, 2004 7:59 AM
	To: sebastien.garcin@francetelecom.com; Barnes, Mary
[NGC:B601:EXCH]; sip@ietf.org
	Cc: VL-T-Com-T-TE332@vli.telekom.de
	Subject: AW: [Sip] Privacy statements and History
(draft-ietf-sip-history-info-04.txt)
=09
=09
	Dear Mary and Sebastien,
	here are my Comments to Sebastien's statements:
	=20
	Your first proposal I can accept from my point of view.
	On you second issue my impression is that this issue must be seen
with regard to RFC3323 where privacy and the trust concept is =
described.
Perhaps a reference to this document should be included.
	On your last point, I think we can also refer to RFC 3323.
	=20
	What are you thinking?
	=20
	Best Regards
	=20
	Roland
	=20

		 ---- Mary snipped forwarding info to keep message
smaller----------- =20
	=09
=09
-----Original Message-----
				From: GARCIN Sebastien RD-CORE-ISS
[mailto:sebastien.garcin@francetelecom.com]=20
				Sent: Monday, November 08, 2004 10:11 AM
				To: Barnes, Mary [NGC:B601:EXCH];
sip@ietf.org
				Subject: [Sip] Privacy statements and
History (draft-ietf-sip-history-info-04.txt)
			=09
			=09
				Hi mary, all
				=20
				When reading
draft-ietf-sip-history-info-04.txt, I have trouble in understanding =
some of
the statements which relate to the forwarding rules for history-entries
subject to privacy. It is an important requirement that =
History-entrie(s)
with a Privacy=3Dhistory, session, or header are indeed forwarded to =
entities
which belong to the same trust domain. The removal of specific
history-entries should only occur if the peer does not belong to the =
trust
domain.
				=20
				In the current text (section 4.3.3.1.1) :
				If a request is  being forwarded to a
Request URI associated with a domain for which the proxy is not =
responsible
and there is a Privacy header in the request with a priv-value of =
"session",
"header" or "history", the proxy MUST remove any hi-entry(s) prior to
forwarding.=20
				=20
				The current wording is misleading since it
gives the impression (maybe intentionnal) that it is not possible to =
forward
history-entries with Privacy statements to domains under the =
responsability
of e.g. another operator belonging to the same trust domain. =20
				=20
				[MB]: Current wording is consistent with
terminology in RFC 3261 in terms of describing who is able to change =
the
Request URI in a specific request (based on section 16.5):=20
				   " A proxy MUST NOT add additional targets
to the target set if the
				   Request-URI of the original request does
not indicate a resource this
				   proxy is responsible for.
			=09
				      A proxy can only change the
Request-URI of a request during
				      forwarding if it is responsible for
that URI. "
				Since History-Info (and associated privacy)
are only added to the request, when an entity that is allowed to change =
the
Request-URI retargets the request, it seemed sensible to use consistent
wording to explain that.  =20
				[/MB]=20
				 [SG] I have no problem with the statements
above. My concern is that if there is a hi-entry already embedded in =
the
request with a Privacy statement, then, it should be up to local policy =
to
decide whether or not a proxy shall pass on those hi-entries to a =
trusted
domain. The sentence in section 4.3.3.1.1 precludes this. I would =
propose to
lighten the strenght of the sentence as follows:
				=20
				If a request is  being forwarded to a
Request URI associated with a domain for which the proxy is not =
responsible
and there is a Privacy header in the request with a priv-value of =
"session",
"header" or "history", the proxy MAY remove any hi-entry(s) prior to
forwarding.=20
				=20
				[SG] Another issue is the decision to add
hi-entries in a privacy context. A proxy changing the target (i.e. the =
proxy
is responsible of the resource reflected in the received Request-URI) =
of a
request which contained a "privacy=3Dhistory" header MAY add a =
history-entries
provided that it knows it can rely on other entities within the trust =
domain
to apply the requested privacy. This affect item 4 in the list on =
conditions
of section 4.3.3 and 4.3.3.1.=20
				=20
				The concept of "trust domain" should be used
when discussing the forwarding rules pertaining to information subject =
to
privacy. Furthermore, the requirement for forwarding history-entries to
trusted entities should be stated more clearly in the draft.=20
								[MB]: The
whole concept of what defines privacy in terms of the proxy's use of =
the
privacy header is outside the scope of History-Info functionality and =
really
a matter of local policy.   I think the functionality that you want is =
a
matter of local implementation and policy in terms of operators =
establishing
this "trust domain" model to which you refer.  History-Info defines the
mechanism to ensure the privacy of the requests, but it doesn't =
explicitly
define how the proxy knows whether it is responsible for that resource. =
   I
don't think this is a matter of standardization.  I thought the use of =
the
term "domain" rather than "resouce" would be helpful, but perhaps =
changing
it to the more general "resource" would resolve this concern and/or a
statement clarifying what I've just described should be added in the =
draft.=20
				[/MB]
				[SG]   Changing the term "domain" to
"resource" does not solve the problem I mentionned above. In order to
reflect current operator requirements, the draft should not PRECLUDE =
the
fowarding of existing history information towards trusted domains.
				=20
				Thank you for clarifying this point.
				=20
				Best regards,
				s=E9bastien
				=20
				=20

				------Remainder of non-related part of this
thread has been deleted by Mary------------------



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<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 =
5.5.2658.2">
<TITLE>RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>I still contend that the SHOULD is sufficient.&nbsp; =
SHOULD means that in general processing, if the hi-entry has a privacy =
header, then the usual processing would be that it would be removed =
(i.e it was added for a specific reason and in general should be used =
to remove the entries based on well defined criteria).&nbsp; If there =
are reasons, such as local policy, that would allow the forwarding in =
specific cases, then it's okay that it is forwarded.&nbsp; I think the =
use of MAY results in less precision and I think the value and critera =
for associating and removing the privacy header with the hi-entry =
becomes much less clear.&nbsp; </FONT></P>

<P><FONT SIZE=3D2>I'd like to hear more opinions on this topic, prior =
to agreeing to making the change (from the MUST to MAY rather than MUST =
to SHOULD). </FONT></P>

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

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: GARCIN Sebastien RD-CORE-ISS [<A =
HREF=3D"mailto:sebastien.garcin@francetelecom.com">mailto:sebastien.garc=
in@francetelecom.com</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: Wednesday, November 10, 2004 12:28 PM</FONT>
<BR><FONT SIZE=3D2>To: Barnes, Mary [NGC:B601:EXCH]; Jesske, R; =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Cc: VL-T-Com-T-TE332@vli.telekom.de</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Mary and roland,</FONT>
</P>

<P><FONT SIZE=3D2>Actually, the statement regarding the forwarding of =
hi-entries beyond the domain for which the proxy is responsible should =
rather be a matter of local policy and thus a MAY should be used =
instead of SHOULD. We are potentially dealing with network boundaries =
where agreements for forwarding such kind information can be reached. =
</FONT></P>

<P><FONT SIZE=3D2>Section 4.3.3.1.1</FONT>
<BR><FONT SIZE=3D2>This section should be re-reworded in accordance =
with the statement above (I can provide some text is we can =
agree).</FONT>
</P>

<P><FONT SIZE=3D2>Section 4.3.3.1 is ok (apologies for not being =
explicit)</FONT>
</P>

<P><FONT SIZE=3D2>Best regards,</FONT>
<BR><FONT SIZE=3D2>s=E9bastien</FONT>
</P>
<BR>
<BR>

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

<P><FONT SIZE=3D2>De : Mary Barnes [<A =
HREF=3D"mailto:mary.barnes@nortelnetworks.com">mailto:mary.barnes@nortel=
networks.com</A>] </FONT>
<BR><FONT SIZE=3D2>Envoy=E9 : mercredi 10 novembre 2004 16:21</FONT>
<BR><FONT SIZE=3D2>=C0 : 'Jesske, R'; GARCIN Sebastien RD-CORE-ISS; =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Cc : VL-T-Com-T-TE332@vli.telekom.de</FONT>
<BR><FONT SIZE=3D2>Objet : RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Roland and Sebastien,</FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>I too agree with the proposal to change the strength =
of the referenced statement from section 4.3.3.1.1&nbsp; from MUST to =
SHOULD. This change is consistent with the other normative statements =
in section 4.3.3.1.1, which are SHOULDs rather than MUSTs.&nbsp;&nbsp; =
I'll make that change in the next rev unless someone raises concerns =
over that change. </FONT></P>

<P><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>On the second concern, I'm not clear what the =
proposed change is for the following:</FONT>
<BR><FONT SIZE=3D2>[SG] Another issue is the decision to add hi-entries =
in a privacy context. A proxy changing the target (i.e. the proxy is =
responsible of the resource reflected in the received Request-URI) of a =
request which contained a &quot;privacy=3Dhistory&quot; header MAY add =
a history-entries provided that it knows it can rely on other entities =
within the trust domain to apply the requested privacy. This affect =
item 4 in the list on conditions of section 4.3.3 and 4.3.3.1. =
</FONT></P>

<P><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>Item 4 in section 4.3.3 describes the general =
situation, thus the strength (MAY) describes the general fact that the =
use of privacy is optional.&nbsp; The normative text provided in =
4.3.3.1, provides the general model for adding hi-entries, with the =
privacy specific considerations for adding the entries described in =
4.3.3.1.1.&nbsp; I think what you're suggesting is changing the =
strength of the following statement in section 4.3.3.1:</FONT></P>

<P><FONT SIZE=3D2>&quot; The hi-entry MUST be added following any =
hi-entry received in the request being forwarded.&quot;</FONT>
<BR><FONT SIZE=3D2>I can see that in the context of privacy =
considerations, this might seem misleading.&nbsp; That MUST was =
originally a SHOULD but was changed (as discussed on the list and at =
IETF-60) to MUST based on issue JRE- 4, so a simple change of MUST to =
MAY would not at all work.&nbsp; I thought it was clear from the =
statement at the beginning of 4.3.3.1, that this section is describing =
the scenario under which the general privacy considerations had been =
evaluated and the intention was to add an hi-entry, thus the statements =
should be read in the context that the general screening indicates that =
an hi-entry SHOULD be added and if one is added it MUST be added =
following any entry that's already in the request to preserve the =
ordering.&nbsp;&nbsp; If you have explicit changes to the text that you =
think would further clarify the functionality, we can discuss. =
</FONT></P>

<P><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>Roland, if you have specific text on how I could =
incorporate a reference to RFC3323, we can consider that; fundamentally =
any discussion of privacy depends on RFC 3323, so it's not clear to me =
how you were suggesting we incorporate such a reference.&nbsp; The text =
in this document (history-info) should accurately describe the =
processing impact of the new &quot;history&quot; priv-value. =
</FONT></P>

<P><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>Thanks for your input,</FONT>
<BR><FONT SIZE=3D2>Mary </FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>-----Original =
Message-----</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>From: =
Jesske, R [<A =
HREF=3D"mailto:R.Jesske@t-com.net">mailto:R.Jesske@t-com.net</A>] =
</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Sent: =
Wednesday, November 10, 2004 7:59 AM</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>To: =
sebastien.garcin@francetelecom.com; Barnes, Mary [NGC:B601:EXCH]; =
sip@ietf.org</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Cc: =
VL-T-Com-T-TE332@vli.telekom.de</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Subject: =
AW: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Dear Mary =
and Sebastien,</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>here are =
my Comments to Sebastien's statements:</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> =
</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Your =
first proposal I can accept from my point of view.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>On you =
second issue my impression is that this issue must be seen with regard =
to RFC3323 where privacy and the trust concept is described. Perhaps a =
reference to this document should be included.</FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>On your =
last point, I think we can also refer to RFC 3323.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> =
</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>What are =
you thinking?</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> =
</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Best =
Regards</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> =
</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Roland</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> =
</FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> ---- =
Mary snipped forwarding info to keep message smaller-----------&nbsp; =
</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>-----Original =
Message-----</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>From: GARCIN =
Sebastien RD-CORE-ISS [<A =
HREF=3D"mailto:sebastien.garcin@francetelecom.com">mailto:sebastien.garc=
in@francetelecom.com</A>] </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Sent: Monday, =
November 08, 2004 10:11 AM</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>To: Barnes, =
Mary [NGC:B601:EXCH]; sip@ietf.org</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Subject: =
[Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Hi mary, =
all</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>When reading =
draft-ietf-sip-history-info-04.txt, I have trouble in understanding =
some of the statements which relate to the forwarding rules for =
history-entries subject to privacy. It is an important requirement that =
History-entrie(s) with a Privacy=3Dhistory, session, or header are =
indeed forwarded to entities which belong to the same trust domain. The =
removal of specific history-entries should only occur if the peer does =
not belong to the trust domain.</FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>In the =
current text (section 4.3.3.1.1) :</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>If a request =
is&nbsp; being forwarded to a Request URI associated with a domain for =
which the proxy is not responsible and there is a Privacy header in the =
request with a priv-value of &quot;session&quot;, &quot;header&quot; or =
&quot;history&quot;, the proxy MUST remove any hi-entry(s) prior to =
forwarding. </FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>The current =
wording is misleading since it gives the impression (maybe =
intentionnal) that it is not possible to forward history-entries with =
Privacy statements to domains under the responsability of e.g. another =
operator belonging to the same trust domain.&nbsp; </FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>[MB]: Current =
wording is consistent with terminology in RFC 3261 in terms of =
describing who is able to change the Request URI in a specific request =
(based on section 16.5): </FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>&nbsp;&nbsp; =
&quot; A proxy MUST NOT add additional targets to the target set if =
the</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>&nbsp;&nbsp; =
Request-URI of the original request does not indicate a resource =
this</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>&nbsp;&nbsp; =
proxy is responsible for.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; A proxy can only change the =
Request-URI of a request during</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; forwarding if it is responsible =
for that URI. &quot;</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Since =
History-Info (and associated privacy) are only added to the request, =
when an entity that is allowed to change the Request-URI retargets the =
request, it seemed sensible to use consistent wording to explain =
that.&nbsp;&nbsp; </FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>[/MB] </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> [SG] I =
have no problem with the statements above. My concern is that if there =
is a hi-entry already embedded in the request with a Privacy statement, =
then, it should be up to local policy to decide whether or not a proxy =
shall pass on those hi-entries to a trusted domain. The sentence in =
section 4.3.3.1.1 precludes this. I would propose to lighten the =
strenght of the sentence as follows:</FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>If a request =
is&nbsp; being forwarded to a Request URI associated with a domain for =
which the proxy is not responsible and there is a Privacy header in the =
request with a priv-value of &quot;session&quot;, &quot;header&quot; or =
&quot;history&quot;, the proxy MAY remove any hi-entry(s) prior to =
forwarding. </FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>[SG] Another =
issue is the decision to add hi-entries in a privacy context. A proxy =
changing the target (i.e. the proxy is responsible of the resource =
reflected in the received Request-URI) of a request which contained a =
&quot;privacy=3Dhistory&quot; header MAY add a history-entries provided =
that it knows it can rely on other entities within the trust domain to =
apply the requested privacy. This affect item 4 in the list on =
conditions of section 4.3.3 and 4.3.3.1. </FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>The concept =
of &quot;trust domain&quot; should be used when discussing the =
forwarding rules pertaining to information subject to privacy. =
Furthermore, the requirement for forwarding history-entries to trusted =
entities should be stated more clearly in the draft. </FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>[MB]: The =
whole concept of what defines privacy in terms of the proxy's use of =
the privacy header is outside the scope of History-Info functionality =
and really a matter of local policy.&nbsp;&nbsp; I think the =
functionality that you want is a matter of local implementation and =
policy in terms of operators establishing this &quot;trust domain&quot; =
model to which you refer.&nbsp; History-Info defines the mechanism to =
ensure the privacy of the requests, but it doesn't explicitly define =
how the proxy knows whether it is responsible for that =
resource.&nbsp;&nbsp;&nbsp; I don't think this is a matter of =
standardization.&nbsp; I thought the use of the term &quot;domain&quot; =
rather than &quot;resouce&quot; would be helpful, but perhaps changing =
it to the more general &quot;resource&quot; would resolve this concern =
and/or a statement clarifying what I've just described should be added =
in the draft. </FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>[/MB]</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>[SG]&nbsp;&nbsp; Changing the term &quot;domain&quot; to =
&quot;resource&quot; does not solve the problem I mentionned above. In =
order to reflect current operator requirements, the draft should not =
PRECLUDE the fowarding of existing history information towards trusted =
domains.</FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Thank you for =
clarifying this point.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Best =
regards,</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>s=E9bastien</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT SIZE=3D2> </FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>------Remainder of non-related part of this thread has been =
deleted by Mary------------------</FONT></P>
<BR>

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From sip-bounces@ietf.org  Wed Nov 10 16:44:31 2004
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From: "Spencer Dawkins" <spencer@mcsr-labs.org>
To: <sip@ietf.org>
References: <F8EFC4B4A8C016428BC1F589296D4FBF07B0876E@esealnt630.al.sw.ericsson.se>
	<00b001c4c658$4e32f2c0$5757430a@BPenfield2>
Subject: Re: [Sip] PING/PONG
Date: Wed, 10 Nov 2004 16:24:42 -0500
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Call me old-fashioned, but I'm requesting a point of clarification 
about PING/PONG.

As I should have said in the working group session when we discussed 
this, I have no interest in chatting about UDP versus TCP. If you 
choose UDP, I assume you did it for a reason, and the discussion at 
the working group session makes sense to me.

Having said that,

> With UDP, the PING will create a new NAT binding and the server can 
> update
> its mapping for the UA.
>
> With TCP, if you want to be able to detect that the NAT binding is 
> gone on
> the order of a transaction timeout, you would need to send a PING 
> every 32
> seconds or so. Given that NAT bindings for TCP live longer than 
> that, I
> don't see what the CRLF buys you.

it seems to me that if you have chosen to use TCP for some reason, you 
need to at least consider living with TCP behavior on retries.

The TCP protocol is extremely persistent, uses exponential backoff 
(typically to a maximum of 60-64 seconds per retry), and in most 
vanilla implementations will not declare a connection unusable for 
something like 10-15 minutes.

If a path is lost, for any reason, TCP will use this behavior (the 
behavior that allows the Internet to survive the World Wide Web 
today).

If you are using TCP and you don't like this behavior, and you don't 
get a response back when you wanted to get a response back, the 
application choices are:

- sit on your hands and wait for the TCP connection to fail,

- close the TCP connection and open another one, or

- close the TCP connection, sit on your hands for some period of time, 
and open another one.

The second choice (immediate retry) requires that you reset the TCP 
connection (one packet), send a SYN to open a second TCP connection 
(one packet), and potentially ACK an incoming SYN/ACK (one packet), 
and then resend your application message (could be piggybacked but 
most TCP stacks will use a separate packet).

Immediately sending up to four new packets when you detect loss of one 
packet, with the possibility that the single-packet loss is due to 
network congestion, seems too aggressive.

Sitting on your hands and letting TCP retry doesn't seem much worse 
than sitting on your hands and then setting up a new TCP connection.

Are we assuming that network congestion is not an issue any more?

Thanks for any guidance you can provide!

Spencer 



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Subject: RE: [Sip] PING/PONG
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Thread-Topic: [Sip] PING/PONG
thread-index: AcTHcMo5CP2b5eryTiy5Nl/L8i5PrAABvRvg
From: "Christian Stredicke" <Christian.Stredicke@snom.de>
To: "Spencer Dawkins" <spencer@mcsr-labs.org>
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Hi Spencer!

You just mentioned another way of keeping TCP connections alive. Thats
good!=20

So far we have:

- CRLF (frequenly used today, but not indicated by user agent)
- STUN (same as CRLF)
- PING/PONG or whatever (tbd, maybe superflous)
- OPTIONS (big overhead, but compatible)
- REGISTER (same as OPTIONS, but even bigger overhead)
- TCP low-level (how can this be addressed by the application?)
- No refreshing ("legacy device")

Lets have a way to indicate these options and let the SBC decide what
method it likes/supports.

CS

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Spencer Dawkins
> Sent: Wednesday, November 10, 2004 5:01 PM
> To: sip@ietf.org
> Subject: Re: [Sip] PING/PONG
>=20
> Call me old-fashioned, but I'm requesting a point of clarification=20
> about PING/PONG.
>=20
> As I should have said in the working group session when we discussed=20
> this, I have no interest in chatting about UDP versus TCP. If you=20
> choose UDP, I assume you did it for a reason, and the discussion at=20
> the working group session makes sense to me.
>=20
> Having said that,
>=20
> > With UDP, the PING will create a new NAT binding and the server can=20
> > update
> > its mapping for the UA.
> >
> > With TCP, if you want to be able to detect that the NAT binding is=20
> > gone on
> > the order of a transaction timeout, you would need to send a PING=20
> > every 32
> > seconds or so. Given that NAT bindings for TCP live longer than=20
> > that, I
> > don't see what the CRLF buys you.
>=20
> it seems to me that if you have chosen to use TCP for some=20
> reason, you=20
> need to at least consider living with TCP behavior on retries.
>=20
> The TCP protocol is extremely persistent, uses exponential backoff=20
> (typically to a maximum of 60-64 seconds per retry), and in most=20
> vanilla implementations will not declare a connection unusable for=20
> something like 10-15 minutes.
>=20
> If a path is lost, for any reason, TCP will use this behavior (the=20
> behavior that allows the Internet to survive the World Wide Web=20
> today).
>=20
> If you are using TCP and you don't like this behavior, and you don't=20
> get a response back when you wanted to get a response back, the=20
> application choices are:
>=20
> - sit on your hands and wait for the TCP connection to fail,
>=20
> - close the TCP connection and open another one, or
>=20
> - close the TCP connection, sit on your hands for some period=20
> of time,=20
> and open another one.
>=20
> The second choice (immediate retry) requires that you reset the TCP=20
> connection (one packet), send a SYN to open a second TCP connection=20
> (one packet), and potentially ACK an incoming SYN/ACK (one packet),=20
> and then resend your application message (could be piggybacked but=20
> most TCP stacks will use a separate packet).
>=20
> Immediately sending up to four new packets when you detect=20
> loss of one=20
> packet, with the possibility that the single-packet loss is due to=20
> network congestion, seems too aggressive.
>=20
> Sitting on your hands and letting TCP retry doesn't seem much worse=20
> than sitting on your hands and then setting up a new TCP connection.
>=20
> Are we assuming that network congestion is not an issue any more?
>=20
> Thanks for any guidance you can provide!
>=20
> Spencer=20
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20
>=20
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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There has been a lot of discussion in the past on the method of
indicating a change of identity to a UA, as an explicit user action at
the remote UA, because of transfer within a legacy network or by 3PCC in
a B2BUA, or even because of aliasing, forwarding or
redirection+recursion within a target domain when there is no mechanism
to indicate identity in a response. At San Diego the topic was revisited
with 3 options on the table:

      1) Use of INVITE/Replaces as a kind of self-transfer

      2) Changing the From header during a dialog

      3) Inventing a new header

Based on the arguments put forward during that session, there was a hum
in favor of INVITE/Replaces as the only mechanism supported.=20

=20

On reflection, I don't think all the issues were aired at the time or in
the small amount of subsequent list discussion. The INVITE/Replaces
mechanism certainly has some appealing similarities to other scenarios
(particularly the pure SIP transfer case, of course). I don't know that
these are compelling when the application is to update information about
a single session, and it seems to me there are going to be cases where
it is not the most efficient or convenient way of doing so. Some
specific issues:

=20

1) How does the SDP in the INVITE/Replaces relate to the media session
for the original dialog? There are more possibilities than normal
because the new dialog has the same participants as the old:

      a) Same SDP session ID and version

      b) Same SDP session ID, new version

      c) New SDP session

I personally would find a) and b) a bit odd because they mean that the
media session migrates from one SIP dialog to another. Not mentioned in
the Replaces spec as far as I can see, though not ruled out either, I
suppose. I'm not sure what everyone involved in the discussions had in
mind (or what implementations might do in cases a) and b) ...).

=20

2) I seem to recall claims that INVITE/Replaces would not introduce
significant loss of efficiency compared with UPDATE (or re-INVITE). If
case c) is the only reasonable one above, this may not be so, since QoS
and media encryption negotiation could be necessary. (It might even be
the case that the new request fails call admission, even though the
intent is to reuse the resources already allocated to the original call,
because the entity making the decision does not know this is a
replacement.) An in-dialog update mechanism might be able to avoid this.
Indeed, with John Elwell's proposed amendments to the relevant RFCs,
there might be no mention of media session parameters at all. I think
this more lightweight procedure should at least be possible.

=20

3) INVITE/Replaces does not necessarily pass through the same
intermediaries as the original dialog. While this has to be dealt with
in other scenarios (and some in the WG appear to regard this as a Good
Thing), applications that like to "misuse" dialog stickiness will surely
find it at least convenient to be in the path of the
post-identity-change session. Presumably the answer to this is for such
an application to be a B2BUA that provides GRUU+crypto-grid Contacts
such that an INV/R from either side will pass through the application -
but such a B2BUA is not transparent to the mechanism proposed in
draft-ietf-sip-identity.

=20

4) The hum preference for INVITE/Replaces seemed in large part to be
motivated by difficulties with the From header. Since there may be other
state information that will need updating in the future, it would be a
bad idea if the adoption of INVITE/Replaces for updating From were to be
taken as a precedent for any other state update, which would not be
bound by this restriction. A consistent and efficient way of updating
state information is required, and INVITE/Replaces doesn't meet the
efficiency criterion. The second hum choice was to introduce another
header. This would work, at the cost of a certain amount of redundancy
with From. In particular, draft-ietf-sip-identity would need enhancing
to allow authentication of the new header as well as (or instead of) the
>From header.

=20

5) All of which leads back to the least-favorite hum choice, which I
feel is the best - to allow From to change. The motivation in 3261 for
keeping the From URI immutable was (temporary) 2543-compatibility, and
there is pretty clear language about not depending on that feature. I
question whether there are 2543-only endpoints out there that are
capable of deployment in realistic environments (that require SIPS and
GRUU support for INVITE/Replaces authorization, for example). As pointed
out in San Diego, the worst that would happen would be that a
2543-compliant UA would reject the re-INVITE or UPDATE with a 481. If
the intent is at some point to follow through on the threat in 3261 to
allow From to be changed, why not do it now, and allow Identity to be
used for update within a dialog?

=20

Peter.

=20

Peter Mataga

Converged Systems Division

Avaya, Inc.

=20

=20


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<div class=3DSection1>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>There has been a =
lot of
discussion in the past on the method of indicating a change of identity =
to a
UA, as an explicit user action at the remote UA, because of transfer =
within a
legacy network or by 3PCC in a B2BUA, or even because of aliasing, =
forwarding
or redirection+recursion within a target domain when there is no =
mechanism to
indicate identity in a response. At <st1:City w:st=3D"on"><st1:place =
w:st=3D"on">San
  Diego</st1:place></st1:City> the topic was revisited with 3 options on =
the
table:<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1)
Use of INVITE/Replaces as a kind of =
self-transfer<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2)
Changing the From header during a dialog<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3)
Inventing a new header<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>Based on the =
arguments put
forward during that session, there was a hum in favor of INVITE/Replaces =
as the
only mechanism supported. <o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>On reflection, I =
don't think
all the issues were aired at the time or in the small amount of =
subsequent list
discussion. The INVITE/Replaces mechanism certainly has some appealing
similarities to other scenarios (particularly the pure SIP transfer =
case, of
course). I don&#8217;t know that these are compelling when the =
application is
to update information about a single session, and it seems to me there =
are
going to be cases where it is not the most efficient or convenient way =
of doing
so. Some specific issues:<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>1) How does the SDP =
in the
INVITE/Replaces relate to the media session for the original dialog? =
There are
more possibilities than normal because the new dialog has the same =
participants
as the old:<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a)
Same SDP session ID and version<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; b)
Same SDP session ID, new version<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; c)
New SDP session<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>I personally would =
find a)
and b) a bit odd because they mean that the media session migrates from =
one SIP
dialog to another. Not mentioned in the Replaces spec as far as I can =
see,
though not ruled out either, I suppose. I'm not sure what everyone =
involved in
the discussions had in mind (or what implementations might do in cases =
a) and
b) ...).<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>2) I seem to recall =
claims
that INVITE/Replaces would not introduce significant loss of efficiency
compared with UPDATE (or re-INVITE). If case c) is the only reasonable =
one
above, this may not be so, since QoS and media encryption negotiation =
could be
necessary. (It might even be the case that the new request fails call
admission, even though the intent is to reuse the resources already =
allocated
to the original call, because the entity making the decision does not =
know this
is a replacement.) An in-dialog update mechanism might be able to avoid =
this.
Indeed, with John Elwell's proposed amendments to the relevant RFCs, =
there
might be no mention of media session parameters at all. I think this =
more
lightweight procedure should at least be =
possible.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>3) INVITE/Replaces =
does not
necessarily pass through the same intermediaries as the original dialog. =
While
this has to be dealt with in other scenarios (and some in the WG appear =
to
regard this as a Good Thing), applications that like to =
&#8220;misuse&#8221;
dialog stickiness will surely find it at least convenient to be in the =
path of
the post-identity-change session. Presumably the answer to this is for =
such an
application to be a B2BUA that provides GRUU+crypto-grid Contacts such =
that an
INV/R from either side will pass through the application &#8211; but =
such a
B2BUA is not transparent to the mechanism proposed in =
draft-ietf-sip-identity.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>4) The hum =
preference for
INVITE/Replaces seemed in large part to be motivated by difficulties =
with the
>From header. Since there may be other state information that will need =
updating
in the future, it would be a bad idea if the adoption of INVITE/Replaces =
for
updating From were to be taken as a precedent for any other state =
update, which
would not be bound by this restriction. A consistent and efficient way =
of
updating state information is required, and INVITE/Replaces doesn't meet =
the
efficiency criterion. The second hum choice was to introduce another =
header. This
would work, at the cost of a certain amount of redundancy with From. In
particular, draft-ietf-sip-identity would need enhancing to allow
authentication of the new header as well as (or instead of) the From =
header.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>5) All of which =
leads back
to the least-favorite hum choice, which I feel is the best &#8211; to =
allow
>From to change. The motivation in 3261 for keeping the From URI =
immutable was
(temporary) 2543-compatibility, and there is pretty clear language about =
not depending
on that feature. I question whether there are 2543-only endpoints out =
there
that are capable of deployment in realistic environments (that require =
SIPS and
GRUU support for INVITE/Replaces authorization, for example). As pointed =
out in
<st1:City w:st=3D"on"><st1:place w:st=3D"on">San =
Diego</st1:place></st1:City>, the
worst that would happen would be that a 2543-compliant UA would reject =
the
re-INVITE or UPDATE with a 481. If the intent is at some point to follow
through on the threat in 3261 to allow From to be changed, why not do it =
now,
and allow Identity to be used for update within a =
dialog?<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Peter.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>Peter =
Mataga<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>Converged Systems =
Division<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>Avaya, =
Inc.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><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'><o:p>&nbsp;</o:p></span></font></p>

</div>

</body>

</html>

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Subject: RE: [Sip] PING/PONG [bcc][faked-from][heur]
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From: "Christian Jansson" <christian.jansson@hotsip.com>
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        "Spencer Dawkins" <spencer@mcsr-labs.org>
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>Lets have a way to indicate these options and let the SBC decide what
>method it likes/supports.

So, this discussion is about how to make UAs interop with SBCs? That was
new to me. As SBC are not even well defined does it make sense to try to
make UAs interop with them? Don't get me wrong, I do think it is very
good to get a standardized way of handling the keep-alive problem, but
putting a SBC as the focus of the requirements before it is well defined
seems strange.

>You just mentioned another way of keeping TCP connections alive. Thats
>good!=20
> So far we have:

>- CRLF (frequenly used today, but not indicated by user agent)

My gut feeling was that could be done without any
indication/negotiation, but now I understand why you want this feature,
you have a SBC in mind. If that had been written somewhere in this
thread, at least I would have understood your motivation.

>- STUN (same as CRLF)

Just checking, STUN over TCP?

>- PING/PONG or whatever (tbd, maybe superflous)
>- OPTIONS (big overhead, but compatible)
>- REGISTER (same as OPTIONS, but even bigger overhead)
>- TCP low-level (how can this be addressed by the application?)
>- No refreshing ("legacy device")

REGISTER without a contact header, would that make it light enough? Bad
to overload REGISTER? Backwards compatible? Perhaps the response would
still need to include all registered contacts for the AOR, so the
registrar still has to access the DB.

My opinion is that none of the OPTIONS, REGISTER, TCP low-level
fiddling, no refreshing at all, is the right method for solving the
problem, exactly because of the problems you stated with each method.

/ Christian Jansson

=20


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From: "Christian Stredicke" <Christian.Stredicke@snom.de>
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Christian Jansson
> Sent: Thursday, November 11, 2004 4:32 AM
> To: Christian Stredicke; Spencer Dawkins
> Cc: sip@ietf.org
> Subject: RE: [Sip] PING/PONG [bcc][faked-from][heur]
>=20
>=20
> >Lets have a way to indicate these options and let the SBC decide what
> >method it likes/supports.
>=20
> So, this discussion is about how to make UAs interop with=20
> SBCs? That was
> new to me. As SBC are not even well defined does it make=20
> sense to try to
> make UAs interop with them? Don't get me wrong, I do think it is very
> good to get a standardized way of handling the keep-alive problem, but
> putting a SBC as the focus of the requirements before it is=20
> well defined
> seems strange.

Lets say a SBC is an example. I dont want to be pulled into the SBC
discussion.

>=20
> >You just mentioned another way of keeping TCP connections=20
> alive. Thats
> >good!=20
> > So far we have:
>=20
> >- CRLF (frequenly used today, but not indicated by user agent)
>=20
> My gut feeling was that could be done without any
> indication/negotiation, but now I understand why you want=20
> this feature,
> you have a SBC in mind. If that had been written somewhere in this
> thread, at least I would have understood your motivation.

Well the [refreshing device] needs to know if it has to do something or
not (eg. send OPTIONS periodically). This needs to be indicated. Looking
at the User-Agent header is surely not a good way...

>=20
> >- STUN (same as CRLF)
>=20
> Just checking, STUN over TCP?

Why not?

>=20
> >- PING/PONG or whatever (tbd, maybe superflous)
> >- OPTIONS (big overhead, but compatible)
> >- REGISTER (same as OPTIONS, but even bigger overhead)
> >- TCP low-level (how can this be addressed by the application?)
> >- No refreshing ("legacy device")
>=20
> REGISTER without a contact header, would that make it light=20
> enough? Bad
> to overload REGISTER? Backwards compatible? Perhaps the response would
> still need to include all registered contacts for the AOR, so the
> registrar still has to access the DB.
>=20
> My opinion is that none of the OPTIONS, REGISTER, TCP low-level
> fiddling, no refreshing at all, is the right method for solving the
> problem, exactly because of the problems you stated with each method.

STUN sounds like a very good method to me...

>=20
> / Christian Jansson
>=20
> =20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20
>=20
>=20

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Date: Thu, 11 Nov 2004 11:01:58 -0500
From: Jonathan Rosenberg <jdrosen@cisco.com>
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To: Adam Roach <adam@nostrum.com>
Subject: Re: [Sip] Event Lists: Back-End Credentials
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Its worth noting that this issue has come up in the context of 3gpp, 
where it is actually even harder of a problem than it is here. I suggest 
  folks read section 12.1 of RFC 3725.

Basically, what is says is that delegation is a hard problem, we don't 
have a solution for it. So, in the interim, the 3pcc controller set the 
 From in a specific way in order to be usable in applications which 
(perhaps sadly) don't authenticate. And that's it.

I'm not sure if precedent of past acceptability of this kind of 
statement can help us, but at least there is a precedent.

-Jonathan R.



Adam Roach wrote:

> hisham.khartabil@nokia.com wrote:
> 
>>
>>> A consequence of this is that, lacking any other mechanism, the 
>>> resources in a resource list for which the RLS is not authoritative 
>>> will contain only the information that their notifier is willing to 
>>> reveal to the RLS -- which is generally the same information they 
>>> would render to unauthenticated users. In some cases, this might 
>>> result in no useful information being transferred.
>>>   
>>
>>
>> This limitation by itself kills the idea, IMO. Most presence lists 
>> will rely on the event list, and if its gonna mean that no useful 
>> information will come to the watcher, then the whole mecahnism is 
>> useless.
>>  
>>
> 
> That's a bit of a strong characterization, and it seems to ignore the 
> remainder of the mail that I sent out; specifically:
> 
>   1. Users have the ability to subscribe to any resources that the RLS
>      marks itself as non-authoritative, and
>   2. (More Importantly) Provisions for delegation of authority are
>      included -- they're just not in-scope for this document.
> 
> /a
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Subject: RE: [Sip] PING/PONG [bcc][faked-from][heur] [mx][spf]
Date: Thu, 11 Nov 2004 17:40:23 +0100
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From: "Christian Jansson" <christian.jansson@hotsip.com>
To: "Christian Stredicke" <Christian.Stredicke@snom.de>,
        "Spencer Dawkins" <spencer@mcsr-labs.org>
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Let's step back and look at the requirements.=20

We have a problem that UAs are placed behind NATs and FWs. Those NATs
and FWs give us two big problems:
Problem1. Machines from the outside cannot easily connect/communicate
with the UAs on the inside, at least not without initiative from the UAs
on the inside.
Problem2. Even though the UAs on the inside open up for communication
through the NAT/FW, the NAT/FW doesn't keep the state for as long as we
would wish.

This gives us two requirements:
Req1. Open up for two way communication through the NAT/FW
Req2. Ensure req1 for as long as it is needed for the UA on the inside.

There are also some facts:
Fact1. In almost all normal situations it will only be the UA that can
open up the NAT/FW for two way communication.
Fact2. It is the UA that knows for how long the two way communication
through the NAT/FW is needed.
Fact3. It is generally good to put less work on servers and let the UAs
do as much of the work as possible.=20

>From fact1 I draw the conclusion that the UA must open up the two way
communication because it will be the only reliable solution.
Fact2 and fact3 together with the conclusion that the UA must fulfill
fact1 gives me a strong feeling that the UA should be the only one that
should do the job of fulfilling the requirements.

I understand that there could be other solutions where machines on the
outer side of the NAT/FW could partly fulfill req2, that is, they could
try to keep the communication path open. But such a machine cannot be
sure exactly when to stop, and, this is the big one, it can't do
anything if it detects that the communication path fails ... nothing! If
the UA on the other hand fulfilled req2 and detected a problem, it could
open up a new communication path and everything would start working
again.

I do not think we should fool UA implementers into thinking that they
can solve the NAT/FW problem by negotiating with another device on the
outside of the NAT/FW. I think it is better to write down how to open,
use, and maintain a two way SIP signaling path through NATs and FWs.

My suggested solution:
Follow what's in draft-jennings-sipping-outbound-00:
- Keep-alives should be sent by the UA.
- Keep-alives for TCP should be CRLF (default-interval TBD).=20
- Keep-alives for UDP could be the SIP methods REGISTER, or a new
light-weight SIP PING method, to be able to get the rport and received
from the response, or STUN. Frequent REGISTERs seems to heavy-weight for
the servers. That leaves us with PING or STUN. PING advantage, it is
SIP, disadvantage, it is new. STUN advantage, have been around for a
while, disadvantage, it is not SIP.

/ Christian Jansson



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Thu Nov 11 12:09:13 2004
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Guys,

The attached drafts, submitted on time, haven't made their way to the
IETF repository. Please, take a look at them before the meeting today.
There have been no changes since the previous version
draft-olson-sipping-refer-extensions-02 rather than splitting the draft
into two and submitting them as SIP WG items.

=20

Thanks a lot,

Orit.


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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>The attached drafts, submitted on time, haven&#8217;t =
made
their way to the IETF repository. Please, take a look at them before the
meeting today. There have been no changes since the previous version =
</span></font><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>draft-olson-sipping-refer-extensions-02
rather than splitting the draft into two and submitting them as SIP WG =
items.</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New"'>Thanks a lot,</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New"'>Orit.</span></font></p>

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------_=_NextPart_001_01C4C80E.CDA3C5ED
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_______________________________________________
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From sip-bounces@ietf.org  Thu Nov 11 14:05:29 2004
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From: "Spencer Dawkins" <spencer@mcsr-labs.org>
To: "Christian Jansson" <christian.jansson@hotsip.com>,
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Subject: Re: [Sip] PING/PONG [bcc][faked-from][heur] [mx][spf]
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OK, I'm still confused here. Please help me.

The apparent choices include "Keep-alives for TCP should be CRLF 
(default-interval TBD)".

Sending keep-alives over TCP may be interesting, but I'm really 
confused about what you think is a reasonable action when you send a 
PING over TCP and don't get a PONG back.

If you change a status indicator on a user display that says "may have 
lost connectivity", that's fine. If you try to retransmit requests, 
I'm concerned.

My definition of "reasonable" includes "don't send multiple new 
packets into a network path that is not returning PONGs, when the 
problem may be that the network path is losing packets because it is 
already congested." Retransmitting requests requires exactly this 
(tear down existing TCP connection, because it's still retransmitting, 
open a new TCP connection, and then retransmit the request at the 
application level).

There is no way for the endpoints to know that loss is NOT due to 
congestion, and putting TCP-based "lost packet multipliers" in place 
in end systems that can't know they aren't facing congestion is not 
something I'm comfortable with.

If TCP isn't the right answer, using TCP and sending multiple new TCP 
packets when a connection won't PING/PONG isn't helpful.

Spencer 



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From sip-bounces@ietf.org  Thu Nov 11 16:17:15 2004
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Jonathan Rosenberg wrote:
> Its worth noting that this issue has come up in the context of 3gpp, 

Oops - s/3gpp/3pcc

> where it is actually even harder of a problem than it is here. I suggest 
>  folks read section 12.1 of RFC 3725.
> 
> Basically, what is says is that delegation is a hard problem, we don't 
> have a solution for it. So, in the interim, the 3pcc controller set the 
>  From in a specific way in order to be usable in applications which 
> (perhaps sadly) don't authenticate. And that's it.
> 
> I'm not sure if precedent of past acceptability of this kind of 
> statement can help us, but at least there is a precedent.
> 
> -Jonathan R.
> 
> 
> 
> Adam Roach wrote:
> 
>> hisham.khartabil@nokia.com wrote:
>>
>>>
>>>> A consequence of this is that, lacking any other mechanism, the 
>>>> resources in a resource list for which the RLS is not authoritative 
>>>> will contain only the information that their notifier is willing to 
>>>> reveal to the RLS -- which is generally the same information they 
>>>> would render to unauthenticated users. In some cases, this might 
>>>> result in no useful information being transferred.
>>>>   
>>>
>>>
>>>
>>> This limitation by itself kills the idea, IMO. Most presence lists 
>>> will rely on the event list, and if its gonna mean that no useful 
>>> information will come to the watcher, then the whole mecahnism is 
>>> useless.
>>>  
>>>
>>
>> That's a bit of a strong characterization, and it seems to ignore the 
>> remainder of the mail that I sent out; specifically:
>>
>>   1. Users have the ability to subscribe to any resources that the RLS
>>      marks itself as non-authoritative, and
>>   2. (More Importantly) Provisions for delegation of authority are
>>      included -- they're just not in-scope for this document.
>>
>> /a
>>
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Hi,

Just a minor note on this: if/when mandating the support for asserted 
identity in the event-list draft, please remember to say that p-asserted 
identities also count (to some extent).

Cheers,
Aki

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Subject: RE: [Sip] Privacy statements and History
	(draft-ietf-sip-history-info-04.txt)
Date: Thu, 11 Nov 2004 15:44:12 +0100
Message-ID: <AB50A99C736B2B45BCA142A99A51F205632079@ftrdmel1.rd.francetelecom.fr>
Thread-Topic: [Sip] Privacy statements and History
	(draft-ietf-sip-history-info-04.txt)
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From: "PROUVOST Sebastien RD-CORE-ISS" <sebastien.prouvost@francetelecom.com>
To: "Mary Barnes" <mary.barnes@nortelnetworks.com>,
        "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>,
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Mary, Sebastien, Roland,=20
=20
I agree that we should let the possibility for a proxy not to remove the =
hi-entry even if privacy is requested and even if the request is =
forwarded to a Request-URI associated with a domain for which the proxy =
is not responsible (if there is an agreement between the domains that =
ensures the proxy that privacy will be applied to the request).=20
I suggest a text that would look like (in case privacy is requested): =
"the hi-entry SHOULD be removed by the proxy unless it knows that it can =
rely on a downstream privacy service to apply the requested privacy ".
=20
Sebastien.


________________________________

De : sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] De la part de =
Mary Barnes
Envoy=E9 : mercredi 10 novembre 2004 19:54
=C0 : GARCIN Sebastien RD-CORE-ISS; Jesske, R; sip@ietf.org
Objet : RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)



I still contend that the SHOULD is sufficient.  SHOULD means that in =
general processing, if the hi-entry has a privacy header, then the usual =
processing would be that it would be removed (i.e it was added for a =
specific reason and in general should be used to remove the entries =
based on well defined criteria).  If there are reasons, such as local =
policy, that would allow the forwarding in specific cases, then it's =
okay that it is forwarded.  I think the use of MAY results in less =
precision and I think the value and critera for associating and removing =
the privacy header with the hi-entry becomes much less clear. =20

I'd like to hear more opinions on this topic, prior to agreeing to =
making the change (from the MUST to MAY rather than MUST to SHOULD).=20

Regards,=20
Mary=20


-----Original Message-----=20
From: GARCIN Sebastien RD-CORE-ISS =
[mailto:sebastien.garcin@francetelecom.com]=20
Sent: Wednesday, November 10, 2004 12:28 PM=20
To: Barnes, Mary [NGC:B601:EXCH]; Jesske, R; sip@ietf.org=20
Cc: VL-T-Com-T-TE332@vli.telekom.de=20
Subject: RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)=20


Mary and roland,=20

Actually, the statement regarding the forwarding of hi-entries beyond =
the domain for which the proxy is responsible should rather be a matter =
of local policy and thus a MAY should be used instead of SHOULD. We are =
potentially dealing with network boundaries where agreements for =
forwarding such kind information can be reached.=20

Section 4.3.3.1.1=20
This section should be re-reworded in accordance with the statement =
above (I can provide some text is we can agree).=20

Section 4.3.3.1 is ok (apologies for not being explicit)=20

Best regards,=20
s=E9bastien=20



________________________________=20

De : Mary Barnes [mailto:mary.barnes@nortelnetworks.com]=20
Envoy=E9 : mercredi 10 novembre 2004 16:21=20
=C0 : 'Jesske, R'; GARCIN Sebastien RD-CORE-ISS; sip@ietf.org=20
Cc : VL-T-Com-T-TE332@vli.telekom.de=20
Objet : RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)=20


Roland and Sebastien,=20
 =20
I too agree with the proposal to change the strength of the referenced =
statement from section 4.3.3.1.1  from MUST to SHOULD. This change is =
consistent with the other normative statements in section 4.3.3.1.1, =
which are SHOULDs rather than MUSTs.   I'll make that change in the next =
rev unless someone raises concerns over that change.=20


On the second concern, I'm not clear what the proposed change is for the =
following:=20
[SG] Another issue is the decision to add hi-entries in a privacy =
context. A proxy changing the target (i.e. the proxy is responsible of =
the resource reflected in the received Request-URI) of a request which =
contained a "privacy=3Dhistory" header MAY add a history-entries =
provided that it knows it can rely on other entities within the trust =
domain to apply the requested privacy. This affect item 4 in the list on =
conditions of section 4.3.3 and 4.3.3.1.=20


Item 4 in section 4.3.3 describes the general situation, thus the =
strength (MAY) describes the general fact that the use of privacy is =
optional.  The normative text provided in 4.3.3.1, provides the general =
model for adding hi-entries, with the privacy specific considerations =
for adding the entries described in 4.3.3.1.1.  I think what you're =
suggesting is changing the strength of the following statement in =
section 4.3.3.1:

" The hi-entry MUST be added following any hi-entry received in the =
request being forwarded."=20
I can see that in the context of privacy considerations, this might seem =
misleading.  That MUST was originally a SHOULD but was changed (as =
discussed on the list and at IETF-60) to MUST based on issue JRE- 4, so =
a simple change of MUST to MAY would not at all work.  I thought it was =
clear from the statement at the beginning of 4.3.3.1, that this section =
is describing the scenario under which the general privacy =
considerations had been evaluated and the intention was to add an =
hi-entry, thus the statements should be read in the context that the =
general screening indicates that an hi-entry SHOULD be added and if one =
is added it MUST be added following any entry that's already in the =
request to preserve the ordering.   If you have explicit changes to the =
text that you think would further clarify the functionality, we can =
discuss.=20


Roland, if you have specific text on how I could incorporate a reference =
to RFC3323, we can consider that; fundamentally any discussion of =
privacy depends on RFC 3323, so it's not clear to me how you were =
suggesting we incorporate such a reference.  The text in this document =
(history-info) should accurately describe the processing impact of the =
new "history" priv-value.=20


Thanks for your input,=20
Mary=20

                -----Original Message-----=20
        From: Jesske, R [mailto:R.Jesske@t-com.net]=20
        Sent: Wednesday, November 10, 2004 7:59 AM=20
        To: sebastien.garcin@francetelecom.com; Barnes, Mary =
[NGC:B601:EXCH]; sip@ietf.org=20
        Cc: VL-T-Com-T-TE332@vli.telekom.de=20
        Subject: AW: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)=20
       =20
       =20
        Dear Mary and Sebastien,=20
        here are my Comments to Sebastien's statements:=20
        =20
        Your first proposal I can accept from my point of view.=20
        On you second issue my impression is that this issue must be =
seen with regard to RFC3323 where privacy and the trust concept is =
described. Perhaps a reference to this document should be included.

        On your last point, I think we can also refer to RFC 3323.=20
        =20
        What are you thinking?=20
        =20
        Best Regards=20
        =20
        Roland=20
        =20

                 ---- Mary snipped forwarding info to keep message =
smaller----------- =20
               =20
                                                                =
-----Original Message-----=20
                                From: GARCIN Sebastien RD-CORE-ISS =
[mailto:sebastien.garcin@francetelecom.com]=20
                                Sent: Monday, November 08, 2004 10:11 AM =

                                To: Barnes, Mary [NGC:B601:EXCH]; =
sip@ietf.org=20
                                Subject: [Sip] Privacy statements and =
History (draft-ietf-sip-history-info-04.txt)=20
                               =20
                               =20
                                Hi mary, all=20
                                =20
                                When reading =
draft-ietf-sip-history-info-04.txt, I have trouble in understanding some =
of the statements which relate to the forwarding rules for =
history-entries subject to privacy. It is an important requirement that =
History-entrie(s) with a Privacy=3Dhistory, session, or header are =
indeed forwarded to entities which belong to the same trust domain. The =
removal of specific history-entries should only occur if the peer does =
not belong to the trust domain.

                                =20
                                In the current text (section 4.3.3.1.1) =
:=20
                                If a request is  being forwarded to a =
Request URI associated with a domain for which the proxy is not =
responsible and there is a Privacy header in the request with a =
priv-value of "session", "header" or "history", the proxy MUST remove =
any hi-entry(s) prior to forwarding.=20

                                =20
                                The current wording is misleading since =
it gives the impression (maybe intentionnal) that it is not possible to =
forward history-entries with Privacy statements to domains under the =
responsability of e.g. another operator belonging to the same trust =
domain. =20

                                =20
                                [MB]: Current wording is consistent with =
terminology in RFC 3261 in terms of describing who is able to change the =
Request URI in a specific request (based on section 16.5):=20

                                   " A proxy MUST NOT add additional =
targets to the target set if the=20
                                   Request-URI of the original request =
does not indicate a resource this=20
                                   proxy is responsible for.=20
                               =20
                                      A proxy can only change the =
Request-URI of a request during=20
                                      forwarding if it is responsible =
for that URI. "=20
                                Since History-Info (and associated =
privacy) are only added to the request, when an entity that is allowed =
to change the Request-URI retargets the request, it seemed sensible to =
use consistent wording to explain that.  =20

                                [/MB]=20
                                 [SG] I have no problem with the =
statements above. My concern is that if there is a hi-entry already =
embedded in the request with a Privacy statement, then, it should be up =
to local policy to decide whether or not a proxy shall pass on those =
hi-entries to a trusted domain. The sentence in section 4.3.3.1.1 =
precludes this. I would propose to lighten the strenght of the sentence =
as follows:

                                =20
                                If a request is  being forwarded to a =
Request URI associated with a domain for which the proxy is not =
responsible and there is a Privacy header in the request with a =
priv-value of "session", "header" or "history", the proxy MAY remove any =
hi-entry(s) prior to forwarding.=20

                                =20
                                [SG] Another issue is the decision to =
add hi-entries in a privacy context. A proxy changing the target (i.e. =
the proxy is responsible of the resource reflected in the received =
Request-URI) of a request which contained a "privacy=3Dhistory" header =
MAY add a history-entries provided that it knows it can rely on other =
entities within the trust domain to apply the requested privacy. This =
affect item 4 in the list on conditions of section 4.3.3 and 4.3.3.1.=20

                                =20
                                The concept of "trust domain" should be =
used when discussing the forwarding rules pertaining to information =
subject to privacy. Furthermore, the requirement for forwarding =
history-entries to trusted entities should be stated more clearly in the =
draft.=20

                                                                [MB]: =
The whole concept of what defines privacy in terms of the proxy's use of =
the privacy header is outside the scope of History-Info functionality =
and really a matter of local policy.   I think the functionality that =
you want is a matter of local implementation and policy in terms of =
operators establishing this "trust domain" model to which you refer.  =
History-Info defines the mechanism to ensure the privacy of the =
requests, but it doesn't explicitly define how the proxy knows whether =
it is responsible for that resource.    I don't think this is a matter =
of standardization.  I thought the use of the term "domain" rather than =
"resouce" would be helpful, but perhaps changing it to the more general =
"resource" would resolve this concern and/or a statement clarifying what =
I've just described should be added in the draft.=20

                                [/MB]=20
                                [SG]   Changing the term "domain" to =
"resource" does not solve the problem I mentionned above. In order to =
reflect current operator requirements, the draft should not PRECLUDE the =
fowarding of existing history information towards trusted domains.

                                =20
                                Thank you for clarifying this point.=20
                                =20
                                Best regards,=20
                                s=E9bastien=20
                                =20
                                =20

                                ------Remainder of non-related part of =
this thread has been deleted by Mary------------------



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1476" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Mary, Sebastien, Roland, </FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I agree that we should let the possibility for =
a proxy not=20
to remove the hi-entry even if privacy is requested and even if the =
request is=20
forwarded to a Request-URI associated with a domain for which the proxy =
is not=20
responsible&nbsp;(if there&nbsp;is an agreement&nbsp;between =
the&nbsp;domains=20
that ensures the proxy that privacy will be applied to the request).=20
</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I suggest a&nbsp;text that would look =
like&nbsp;(in case=20
privacy is requested): "the hi-entry SHOULD be removed by the proxy =
unless it=20
knows that it can rely on a downstream privacy service to apply the =
requested=20
privacy ".</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Sebastien.</FONT></SPAN></DIV><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT><FONT=20
face=3DArial color=3D#0000ff size=3D2></FONT><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT><FONT face=3DArial color=3D#0000ff size=3D2></FONT><BR>
<DIV class=3DOutlookMessageHeader lang=3Dfr dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>De&nbsp;:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>De la part de</B> Mary=20
Barnes<BR><B>Envoy=E9&nbsp;:</B> mercredi 10 novembre 2004=20
19:54<BR><B>=C0&nbsp;:</B> GARCIN Sebastien RD-CORE-ISS; Jesske, R;=20
sip@ietf.org<BR><B>Objet&nbsp;:</B> RE: [Sip] Privacy statements and =
History=20
(draft-ietf-sip-history-info-04.txt)<BR></FONT><BR></DIV>
<DIV></DIV>
<P><FONT size=3D2>I still contend that the SHOULD is sufficient.&nbsp; =
SHOULD=20
means that in general processing, if the hi-entry has a privacy header, =
then the=20
usual processing would be that it would be removed (i.e it was added for =
a=20
specific reason and in general should be used to remove the entries =
based on=20
well defined criteria).&nbsp; If there are reasons, such as local =
policy, that=20
would allow the forwarding in specific cases, then it's okay that it is=20
forwarded.&nbsp; I think the use of MAY results in less precision and I =
think=20
the value and critera for associating and removing the privacy header =
with the=20
hi-entry becomes much less clear.&nbsp; </FONT></P>
<P><FONT size=3D2>I'd like to hear more opinions on this topic, prior to =
agreeing=20
to making the change (from the MUST to MAY rather than MUST to SHOULD).=20
</FONT></P>
<P><FONT size=3D2>Regards, </FONT><BR><FONT size=3D2>Mary</FONT> =
</P><BR>
<P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From: GARCIN=20
Sebastien RD-CORE-ISS [<A=20
href=3D"mailto:sebastien.garcin@francetelecom.com">mailto:sebastien.garci=
n@francetelecom.com</A>]=20
</FONT><BR><FONT size=3D2>Sent: Wednesday, November 10, 2004 12:28 =
PM</FONT>=20
<BR><FONT size=3D2>To: Barnes, Mary [NGC:B601:EXCH]; Jesske, R;=20
sip@ietf.org</FONT> <BR><FONT size=3D2>Cc: =
VL-T-Com-T-TE332@vli.telekom.de</FONT>=20
<BR><FONT size=3D2>Subject: RE: [Sip] Privacy statements and History=20
(draft-ietf-sip-history-info-04.txt)</FONT> </P><BR>
<P><FONT size=3D2>Mary and roland,</FONT> </P>
<P><FONT size=3D2>Actually, the statement regarding the forwarding of =
hi-entries=20
beyond the domain for which the proxy is responsible should rather be a =
matter=20
of local policy and thus a MAY should be used instead of SHOULD. We are=20
potentially dealing with network boundaries where agreements for =
forwarding such=20
kind information can be reached. </FONT></P>
<P><FONT size=3D2>Section 4.3.3.1.1</FONT> <BR><FONT size=3D2>This =
section should be=20
re-reworded in accordance with the statement above (I can provide some =
text is=20
we can agree).</FONT> </P>
<P><FONT size=3D2>Section 4.3.3.1 is ok (apologies for not being =
explicit)</FONT>=20
</P>
<P><FONT size=3D2>Best regards,</FONT> <BR><FONT =
size=3D2>s=E9bastien</FONT>=20
</P><BR><BR>
<P><FONT size=3D2>________________________________</FONT> </P>
<P><FONT size=3D2>De : Mary Barnes [<A=20
href=3D"mailto:mary.barnes@nortelnetworks.com">mailto:mary.barnes@norteln=
etworks.com</A>]=20
</FONT><BR><FONT size=3D2>Envoy=E9 : mercredi 10 novembre 2004 =
16:21</FONT>=20
<BR><FONT size=3D2>=C0 : 'Jesske, R'; GARCIN Sebastien RD-CORE-ISS;=20
sip@ietf.org</FONT> <BR><FONT size=3D2>Cc : =
VL-T-Com-T-TE332@vli.telekom.de</FONT>=20
<BR><FONT size=3D2>Objet : RE: [Sip] Privacy statements and History=20
(draft-ietf-sip-history-info-04.txt)</FONT> </P><BR>
<P><FONT size=3D2>Roland and Sebastien,</FONT> <BR><FONT =
size=3D2>&nbsp;</FONT>=20
<BR><FONT size=3D2>I too agree with the proposal to change the strength =
of the=20
referenced statement from section 4.3.3.1.1&nbsp; from MUST to SHOULD. =
This=20
change is consistent with the other normative statements in section =
4.3.3.1.1,=20
which are SHOULDs rather than MUSTs.&nbsp;&nbsp; I'll make that change =
in the=20
next rev unless someone raises concerns over that change. </FONT></P>
<P><FONT size=3D2></FONT> <BR><FONT size=3D2>On the second concern, I'm =
not clear=20
what the proposed change is for the following:</FONT> <BR><FONT =
size=3D2>[SG]=20
Another issue is the decision to add hi-entries in a privacy context. A =
proxy=20
changing the target (i.e. the proxy is responsible of the resource =
reflected in=20
the received Request-URI) of a request which contained a =
"privacy=3Dhistory"=20
header MAY add a history-entries provided that it knows it can rely on =
other=20
entities within the trust domain to apply the requested privacy. This =
affect=20
item 4 in the list on conditions of section 4.3.3 and 4.3.3.1. =
</FONT></P>
<P><FONT size=3D2></FONT> <BR><FONT size=3D2>Item 4 in section 4.3.3 =
describes the=20
general situation, thus the strength (MAY) describes the general fact =
that the=20
use of privacy is optional.&nbsp; The normative text provided in =
4.3.3.1,=20
provides the general model for adding hi-entries, with the privacy =
specific=20
considerations for adding the entries described in 4.3.3.1.1.&nbsp; I =
think what=20
you're suggesting is changing the strength of the following statement in =
section=20
4.3.3.1:</FONT></P>
<P><FONT size=3D2>" The hi-entry MUST be added following any hi-entry =
received in=20
the request being forwarded."</FONT> <BR><FONT size=3D2>I can see that =
in the=20
context of privacy considerations, this might seem misleading.&nbsp; =
That MUST=20
was originally a SHOULD but was changed (as discussed on the list and at =

IETF-60) to MUST based on issue JRE- 4, so a simple change of MUST to =
MAY would=20
not at all work.&nbsp; I thought it was clear from the statement at the=20
beginning of 4.3.3.1, that this section is describing the scenario under =
which=20
the general privacy considerations had been evaluated and the intention =
was to=20
add an hi-entry, thus the statements should be read in the context that =
the=20
general screening indicates that an hi-entry SHOULD be added and if one =
is added=20
it MUST be added following any entry that's already in the request to =
preserve=20
the ordering.&nbsp;&nbsp; If you have explicit changes to the text that =
you=20
think would further clarify the functionality, we can discuss. =
</FONT></P>
<P><FONT size=3D2></FONT> <BR><FONT size=3D2>Roland, if you have =
specific text on=20
how I could incorporate a reference to RFC3323, we can consider that;=20
fundamentally any discussion of privacy depends on RFC 3323, so it's not =
clear=20
to me how you were suggesting we incorporate such a reference.&nbsp; The =
text in=20
this document (history-info) should accurately describe the processing =
impact of=20
the new "history" priv-value. </FONT></P>
<P><FONT size=3D2></FONT> <BR><FONT size=3D2>Thanks for your =
input,</FONT> <BR><FONT=20
size=3D2>Mary </FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>-----Original=20
Message-----</FONT> <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =

size=3D2>From: Jesske, R [<A=20
href=3D"mailto:R.Jesske@t-com.net">mailto:R.Jesske@t-com.net</A>]=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
size=3D2>Sent:=20
Wednesday, November 10, 2004 7:59 AM</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>To:=20
sebastien.garcin@francetelecom.com; Barnes, Mary [NGC:B601:EXCH];=20
sip@ietf.org</FONT> <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =

size=3D2>Cc: VL-T-Com-T-TE332@vli.telekom.de</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>Subject: =
AW: [Sip]=20
Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>Dear Mary =
and=20
Sebastien,</FONT> <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT=20
size=3D2>here are my Comments to Sebastien's statements:</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
size=3D2>Your first=20
proposal I can accept from my point of view.</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>On you =
second issue=20
my impression is that this issue must be seen with regard to RFC3323 =
where=20
privacy and the trust concept is described. Perhaps a reference to this =
document=20
should be included.</FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>On your =
last point, I=20
think we can also refer to RFC 3323.</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
size=3D2>What are you=20
thinking?</FONT> =
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT=20
size=3D2> </FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
size=3D2>Best=20
Regards</FONT> <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT =
size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
size=3D2>Roland</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2> =
</FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2> ---- =
Mary snipped=20
forwarding info to keep message smaller-----------&nbsp;=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>-----Original=20
Message-----</FONT> <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>From: GARCIN =
Sebastien=20
RD-CORE-ISS [<A=20
href=3D"mailto:sebastien.garcin@francetelecom.com">mailto:sebastien.garci=
n@francetelecom.com</A>]=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>Sent: Monday, =
November=20
08, 2004 10:11 AM</FONT> <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>To: Barnes, =
Mary=20
[NGC:B601:EXCH]; sip@ietf.org</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>Subject: [Sip] =
Privacy=20
statements and History (draft-ietf-sip-history-info-04.txt)</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>Hi mary, =
all</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>When reading=20
draft-ietf-sip-history-info-04.txt, I have trouble in understanding some =
of the=20
statements which relate to the forwarding rules for history-entries =
subject to=20
privacy. It is an important requirement that History-entrie(s) with a=20
Privacy=3Dhistory, session, or header are indeed forwarded to entities =
which=20
belong to the same trust domain. The removal of specific history-entries =
should=20
only occur if the peer does not belong to the trust domain.</FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>In the current =
text=20
(section 4.3.3.1.1) :</FONT> =
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>If a request =
is&nbsp;=20
being forwarded to a Request URI associated with a domain for which the =
proxy is=20
not responsible and there is a Privacy header in the request with a =
priv-value=20
of "session", "header" or "history", the proxy MUST remove any =
hi-entry(s) prior=20
to forwarding. </FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>The current =
wording is=20
misleading since it gives the impression (maybe intentionnal) that it is =
not=20
possible to forward history-entries with Privacy statements to domains =
under the=20
responsability of e.g. another operator belonging to the same trust=20
domain.&nbsp; </FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>[MB]: Current =
wording is=20
consistent with terminology in RFC 3261 in terms of describing who is =
able to=20
change the Request URI in a specific request (based on section 16.5):=20
</FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>&nbsp;&nbsp; " =
A proxy=20
MUST NOT add additional targets to the target set if the</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>&nbsp;&nbsp; =
Request-URI=20
of the original request does not indicate a resource this</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>&nbsp;&nbsp; =
proxy is=20
responsible for.</FONT> <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; A proxy can only change the =
Request-URI of=20
a request during</FONT> <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; forwarding if it is responsible =
for that=20
URI. "</FONT> <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>Since =
History-Info (and=20
associated privacy) are only added to the request, when an entity that =
is=20
allowed to change the Request-URI retargets the request, it seemed =
sensible to=20
use consistent wording to explain that.&nbsp;&nbsp; </FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>[/MB]=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2> [SG] I =
have no=20
problem with the statements above. My concern is that if there is a =
hi-entry=20
already embedded in the request with a Privacy statement, then, it =
should be up=20
to local policy to decide whether or not a proxy shall pass on those =
hi-entries=20
to a trusted domain. The sentence in section 4.3.3.1.1 precludes this. I =
would=20
propose to lighten the strenght of the sentence as follows:</FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>If a request =
is&nbsp;=20
being forwarded to a Request URI associated with a domain for which the =
proxy is=20
not responsible and there is a Privacy header in the request with a =
priv-value=20
of "session", "header" or "history", the proxy MAY remove any =
hi-entry(s) prior=20
to forwarding. </FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>[SG] Another =
issue is=20
the decision to add hi-entries in a privacy context. A proxy changing =
the target=20
(i.e. the proxy is responsible of the resource reflected in the received =

Request-URI) of a request which contained a "privacy=3Dhistory" header =
MAY add a=20
history-entries provided that it knows it can rely on other entities =
within the=20
trust domain to apply the requested privacy. This affect item 4 in the =
list on=20
conditions of section 4.3.3 and 4.3.3.1. </FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>The concept of =
"trust=20
domain" should be used when discussing the forwarding rules pertaining =
to=20
information subject to privacy. Furthermore, the requirement for =
forwarding=20
history-entries to trusted entities should be stated more clearly in the =
draft.=20
</FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>[MB]: The =
whole concept=20
of what defines privacy in terms of the proxy's use of the privacy =
header is=20
outside the scope of History-Info functionality and really a matter of =
local=20
policy.&nbsp;&nbsp; I think the functionality that you want is a matter =
of local=20
implementation and policy in terms of operators establishing this "trust =
domain"=20
model to which you refer.&nbsp; History-Info defines the mechanism to =
ensure the=20
privacy of the requests, but it doesn't explicitly define how the proxy =
knows=20
whether it is responsible for that resource.&nbsp;&nbsp;&nbsp; I don't =
think=20
this is a matter of standardization.&nbsp; I thought the use of the term =

"domain" rather than "resouce" would be helpful, but perhaps changing it =
to the=20
more general "resource" would resolve this concern and/or a statement =
clarifying=20
what I've just described should be added in the draft. </FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>[/MB]</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
size=3D2>[SG]&nbsp;&nbsp;=20
Changing the term "domain" to "resource" does not solve the problem I =
mentionned=20
above. In order to reflect current operator requirements, the draft =
should not=20
PRECLUDE the fowarding of existing history information towards trusted=20
domains.</FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>Thank you for =
clarifying=20
this point.</FONT> <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>Best =
regards,</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
size=3D2>s=E9bastien</FONT>=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2>=20
</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT size=3D2> =
</FONT></P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
size=3D2>------Remainder of=20
non-related part of this thread has been deleted by=20
Mary------------------</FONT></P><BR></BODY></HTML>

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Date: Thu, 11 Nov 2004 16:34:42 -0600
From: Adam Roach <adam@nostrum.com>
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To: Aki Niemi <aki.niemi@nokia.com>
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Cc: SIP WG <sip@ietf.org>
Subject: [Sip] Re: RLS and identity
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Aki Niemi wrote:

>
> Just a minor note on this: if/when mandating the support for asserted 
> identity in the event-list draft, please remember to say that 
> p-asserted identities also count (to some extent).


Based on Rohan's suggestion, the text will effectively say:

 - Jon's Identity draft will be mandatory to implement, optional to use.

 - Other mechanisms that have properties such that they can adequately
   convey the identity of the subscriber and the permission of the RLS
   to subscribe on the user's behalf can also be used.

If you think that your architecture, in combination with 
P-Asserted-Identity, satisfies the second bullet, then you're probably 
fine. However, I think it is a really bad idea to specifically include 
any mention of P-Asserted-Identity in the draft.

/a

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Date: Thu, 11 Nov 2004 14:43:27 -0800
From: Charles Eckel <eckelcu@cisco.com>
Organization: Cisco Systems
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To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
Subject: Re: [Sip] mib-08
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Cc: sip@ietf.org, Cullen Jennings <fluffy@cisco.com>,
        Kevin Lingle <klingle@cisco.com>,
        Steve Langstaff <steve.langstaff@citel.com>
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Just a few recommended clarifications:

sipStatsInbounds OBJECT-TYPE
Description: should state excludes retransmissions

sipStatsRetries OBJECT-TYPE
Description: should replace "total number of INVITE retries" with "total 
number of request retransmissions, could be multiple per request"

sipStatsRetryFinalResponses
Description: should state there could be multiple per response

Cheers,
Charles


Romascanu, Dan (Dan) wrote:

> Steve,
> 
> Actually implementation experience and interoperability is required only 
> when time comes to advance a standards track Proposed Standard to Draft 
> Standard phase. Although 'pre-standard' implementations are not 
> discouraged in I-D phase, one cannot ensure interoperable versions 
> before the standard, because the OID allocation for the root of the MIB 
> module under mib-2 is being performed by IANA only very late, prior to 
> the RFC publication.
> 
> Whether the document will be a standards track, or not, is an IESG 
> decision based on the WG recommendation. 
> 
> Regards,
> 
> Dan
> 
> 
> 
>  > -----Original Message-----
>  > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On
>  > Behalf Of Steve Langstaff
>  > Sent: 09 November, 2004 6:37 PM
>  > To: Kevin Lingle
>  > Cc: sip@ietf.org; Cullen Jennings
>  > Subject: RE: [Sip] mib-08
>  >
>  >
>  > KL: "we need to move the I-D forward so we can get a RFC that
>  > can get some implementation experience."
>  >
>  > I thought that the I-D phase was the time to get the
>  > implementation experience. Is it your intention that the RFC
>  > will be "Experimental" or "Informational" before getting this
>  > implementation experience, and then become a standard afterwards?
>  >
>  > --
>  > Steve Langstaff.
>  >
>  >
>  > -----Original Message-----
>  > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
>  > Kevin Lingle
>  > Sent: 08 November 2004 17:26
>  > To: Cullen Jennings
>  > Cc: sip@ietf.org
>  > Subject: Re: [Sip] mib-08
>  > [snip]
>  >
>  > we need to move the
>  > I-D forward so we can get a RFC that can get some implementation
>  > experience.
>  >
>  > kevin
>  >
>  > _______________________________________________
>  > 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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> 

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From sip-bounces@ietf.org  Thu Nov 11 18:11:03 2004
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From: "Corey W. Alexander" <coreya@nortelnetworks.com>
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Subject: [Sip] -resource-priority-05.txt: Comments and Recommendations
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This should spur some discussion...  I'd rather send this directly to 
the authors, but to keep everything out in the open...

I spoke briefly with Mr. Polk following the IEPREP session on Tuesday. 
I had three general questions.  I'm paraphrasing the responses, and so 
invite James to correct me if I'm misrepresenting him here.

1. Why support multiple namespace/priority (n-p) values in a single 
message, in one or more R-P headers?  And if supported, should not the 
n-p values in the message equate to the same priority level (for 
example, dsn.flash and q735.1 would be equal)?  Would it not be better 
to support only one n-p value per message, and recommend that a 
proxy/gateway between networks supporting two different namespaces 
handle mapping n-p values?
---- There may be/are networks which would support multiple levels.  The 
draft is not dictating that there must be multiple levels.
---- As for having the n-p levels be "equal", James cited a GETS call 
with a specific precedence which becomes a Routine call on the DSN.
---- Having a proxy/gateway mapped between namespaces is an viable 
alternative.

2. Is it wise or valid to explicitly allow a proxy to "upgrade" or 
"downgrade" the  n-p values in a message?
---- I'm coming at this from the DSN perspective, but wps and/or ets may 
allow this kind of thing.  While the DSN/DSRN/(Q735) would not, that 
does not invalidate the concept.

3. With "preferential treatment" of messages based on R-P content, 
should not scenarios be considered and listed in which a lower priority 
message  would need to be handled before a higher priority message?  My 
example: it would be wise to handle a BYE message with dsn.routine 
before an INVITE with dsn.flash-override...  At least in certain 
situations, say when the BYE is sent as a result of a preemption event 
(which the Reason header should likely indicate).
---- Certainly something to consider, especially for DSN.


Having witnessed the majority of the discussion on this draft in the 
final SIP session, and stepping back from the questions I had, it occurs 
to me that, while the information presented is not invalid, it is 
certainly confusing, to a varying degree.  In my opinion, the confusion 
comes from trying to define (describe/outline, whatever) the behavior of 
the various namespaces, while at the same time intending to keep draft 
general, and the use of the R-P header open to those who implement it.

Right before the comments on this draft were brought to a close, the 
thread was leading to there ultimately being independent documents which 
would describe the overall behavior of networks which implement one or 
more of these namespaces.

RECOMMENDATION:  Any namespace-specific behaviors should be relegated to 
documents specific to the namespaces themselves.  For instance, there 
are currently a number of documents, and certainly other to come, 
dealing with the requirements/architecture/implementation of Assured 
Service (MLPP in IP):
draft-baker-tsvwg-mlpp-that-works-02.txt
draft-pierce-tsvwg-assured-service-req-01.txt
draft-pierce-tsvwg-assured-service-arch-01.txt
draft-pierce-tsvwg-pref-treat-examples-01.txt

It is in documents like these - more specifically, documents specific to 
expected behaviors - where the use of the R-P header should ultimately 
be laid out.

I would like to see the R-P draft limited to
- a general description of what the header is: What is a namespace? 
What is the priority values mean?  Why is it needed?  Basically, 
background and definitions, but NOT explicit behaviors.
- the format of the Resource-Priority header
- 417 response code
- error conditions
--- If the R-P header is not supported: 420
--- If the n-p value is not supported or unknown: 417
*** This document does not have to define those circumstances in which 
each defined namespace considers the n-p value unsupported.
- A MINIMUM number of namespaces.
*** I would recommend providing only a single generic namespace, with 5, 
6, whatever, priority values, and leave the actual namespace definitions 
(dsn, dsrn, ets, wps; and q735, if necessary) to documents specific to 
those namespaces/architectures.

This would remove all of the confusion.  No need to describe the 
behaviors expected with the current namespaces, let applications 
specific to the namespaces to that.

Further, the document could then keep the strict, semi-strict, and loose 
mode concepts intact (or with modification) without interfering with 
existing behaviors in the current namespaces.  New namespaces, when 
defined, can align themselves with these modes, or define their own 
behaviors.

Finally, since preferential treatment of messages would likely find 
greater use outside of a private network than preemption, I would also 
recommend one more thing.  If the suggestion of a single generic 
namespace in this draft is taken, behaviors surrounding the concept of 
preferential treatment should be included for this namespace.

Digest and comment.

-- 
Corey W. Alexander

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Subject: RE: [Sip] PING/PONG [bcc][faked-from][heur] [mx][spf]
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>From: Spencer Dawkins [mailto:spencer@mcsr-labs.org]
>
>OK, I'm still confused here. Please help me.
>
>The apparent choices include "Keep-alives for TCP should be CRLF
>(default-interval TBD)".
>
>Sending keep-alives over TCP may be interesting, but I'm really
>confused about what you think is a reasonable action when you send a
>PING over TCP and don't get a PONG back.


When sending the CRLF over TCP you should not get anything back if the
connection is still there. If the TCP connection is down or fails while
sending the CRLF on the other hand you will get a notice from your IP
stack of the failure. Then that socket should be closed, and the UA may
try to set up a new connection if wants to.


>
>If you change a status indicator on a user display that says "may have
>lost connectivity", that's fine. If you try to retransmit requests,
>I'm concerned.
>
>My definition of "reasonable" includes "don't send multiple new
>packets into a network path that is not returning PONGs, when the
>problem may be that the network path is losing packets because it is
>already congested." Retransmitting requests requires exactly this
>(tear down existing TCP connection, because it's still retransmitting,
>open a new TCP connection, and then retransmit the request at the
>application level).
>
>There is no way for the endpoints to know that loss is NOT due to
>congestion, and putting TCP-based "lost packet multipliers" in place
>in end systems that can't know they aren't facing congestion is not
>something I'm comfortable with.


I'm not sure how often keep-alives have to be sent over TCP to ensure
that NAT/FWs do not remove the state. Preferabely NAT/FWs never remove
the state, but I have heard that it happens. Does anyone have a
reference to test statistics of NAT/FWs behavior? The keep-alive default
interval should probably be set just below the shortest of all detected
values. My guess is that the value will be around 5 minutes. Sending a
TCP packet with CRLF every 5 minutes can't really be a big source for
congestion. The keep-alives would also not be sent if any other traffic
had been sent to or from the UA.=20

>
>If TCP isn't the right answer, using TCP and sending multiple new TCP
>packets when a connection won't PING/PONG isn't helpful.

Given that the keep-alives are spaced with a couple of minutes, and that
this interval is much greater than the TCP error timeouts, I do not see
the problem. If those keep-alives would have been sent every 10 seconds
I would have agreed on the problems you describe.

/ Christian Jansson


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From sip-bounces@ietf.org  Fri Nov 12 08:19:19 2004
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Subject: [Sip] comments on sipping-certs
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Cullen, Jon

Some comments on the sipping-certs draft.

The document is in need of a picture that shows the overall architecture 
- something along the lines of a protocol model as described in:

http://www.ietf.org/internet-drafts/draft-iab-model-02.txt

I'd suggest that the picture show the credential server, identity 
service, and clients on both sides, along with a single section 
overviewing operation.

One technical aspect of this bothered me. My understanding is that the 
NOTIFY from the credential server containing the cert has to be signed 
by an identity service in order to provide an indentity binding to the 
cert. However, in this case, the notification is *not* sent by the user 
in the From/identity field! It is sent by an entity which is authorized 
to send requests for that identity, and is going to have a tight trust 
relationship with the authentication service.

The spec is quite loose with terminology; I point out some of these 
below, but it needs a careful scrub in this area.

You need to formally define the various event packages you are using. I 
would suggest aligning the package names with the terminology you are 
using (i.e., a credential and certificate package).

Generally, a lot of details are needed in various places. It reads much 
like a proposal and needs to move on to become a protocol specification.

I think section 7 and 10.2 are out of scope. This is a fairly orthogonal 
piece of work that treats how offer/answers are done using s/mime. It 
does not appear to depend at all on the fact that the certificates were 
obtain through the mechanism in this draft.

section 11.1 is out of place. It belongs in a section that defines 
client behavior for obtained a cert.

also, thinking about this, it seems to me that the mechanism is 
susceptible to SBCs, which would be able to tap into the media stream. 
They would do this by handing out a different cert to subcsribers than 
the one you hand it. Then, any encrypted media for you comes to the SBC, 
which decrypts and re-encrypts with your key. Badness.

Might want to mention this in the security considerations section, and 
recommend that a UA subscribe to its own cert using a different identity 
in order to verify that the proper cert is being handed out.



detailed comments:

> SIP provides a mechanism for end to end encryption and integrity
>    using S/MIME,

s/end to end/end-to-end

> S/MIME has not been widely implemented or deployed due to the
>    complexity of providing a reasonable certificate distribution
>    infrastructure.

This is a bold statement; there are many reasons, of which this is one.

> Combined with the Identity [2] work, this work allows users to have
>    certificates that are not signed by any well known certificate

s/work/specification

> This mechanism allows UAs such as IP phones to enroll
>    and get their credentials without any more configuration information
>    than they commonly have today, without any extra effort or key clicks
>    by the end user, and without any extra expense for the end user.

this is one long run on sentence

> This mechanism also lets the UA discover and retrieve the public
>    certificate for any other user and find out about certificate
>    revocations.

worth noting that it actually provides more than this; its not just 
users, but any resource associated with an AOR.

> The general approach is to provide a new SIP service referred to as a
>    Credential Server that allows UAs to subscribe to some other user's
>    certificate. 

a server provides a service, it isn't a service

> The identity of the certificate can be vouched for
>    using Identity [2] work, which uses the domain's certificate to sign
>    that the NOTIFY really is from the Credential server for the request
>    user in that domain.

This is not true. The signature does not verify this; the signature 
verifies that the request came from a server that has access to the 
domain keys.

> The Credential Service can manage public
>    certificates as well as credentials that include the user's private

you sometimes use credential service, sometimes credential server

also, this text is confusing since credentials apparently contain 
certificates according to your definition. So, saying that they manage 
certifications *as well as* private keys is redundant

> The Credential Server authenticates UAs that
>    are changing credentials or requesting private keys using a shared
>    secret that both the UA and the Server know.

in this sentence, it is not obvious that "using a shared secret" applies 
to both changing credentials and requesting private keys.

Also, my understanding is that you would request credentials as well, 
not just the private key (i.e., there is no way to get just one)

> Typically this will be
>    the same shared secret that is used in Register with the Registrar
>    for the domain.

s/Register/REGISTER

> The mechanism described in this document works for both self signed
>    certificates and certificates signed by a well known certificate
>    authority;

you just said that in the preceding paragraph

> Previous versions of this work proposed using HTTP instead of SIP for
>    communicating with the Credential Server. 

this needs to be changed now

generally, on its way to an rfc, it needs to become less conversational, 
  statements like "this work" and "previous versions" need to go

> The key difference with
>    using SIP is that a certificate can be revoked by sending a new
>    NOTIFY; in the HTTP based scheme, the certificates were cached for a
>    predefined period of time, typically one day, so that a revocation
>    could take effect only after the cache had expired.

there are other differences too, i mentioned some of the benefits of 
presence integration. Its a good idea to compare the pros and cons now 
that http becomes optional.

also the text is confusing because the certificate can also be revoked 
in sip by having it expire

> 2.  Conventions

change to definitions

>  The certificates discussed in this draft are generally self signed
>    and use the mechanisms in the Identity work [2] to vouch for their
>    validity.

the "work" thing again

> 3.  Goals
> 
>    This work allows S/MIME to be used to meet the following goals and
>    requirements:

I think this section needs to become the introduction. Furthermore, much 
of the introduction is sort of a weak overview of operation. You should 
move that text into a stronger overview of operation section.

>  Not require any efforts (zero clicks) on behalf of an end user to
>       acquire and start using end to end security.

s/efforts/effort
s/end to end/end-to-end



>  Not require any extra cost on a per user basis in deploying
>       systems that support this.  It does require that the domain have a
>       web server style certificate that is either self signed or signed
>       by a well known CA.

this is clearly false; I'll require incremental CPU for each user to 
process subscriptions for their cert, and incremental storage.

> Allow negotiation of E2E encrypted SRTP sessions.

please use end-to-end consistently

> Allow end to end encryption and integrity of SIP bodies that may
>       be delivered in SIP signaling, such as page mode MESSAGEs or
>       NOTIFY bodies in presence.

integrity? i usually think of signature as the way we provide integrity, 
and the certs mechanism isn't used for that

hmm. if you want to encrypt presence documents in NOTIFY, what uri do 
you use to get the certificate? is it the identity used in the 
subscription, or the gruu in the contact? should a credential server 
support subscriptiosn to gruu to get certificates?

> Allow end to end encryption and integrity of SIP bodies that may
>       be delivered in SIP signaling, such as page mode MESSAGEs or
>       NOTIFY bodies in presence.

i had to read this a few times to realize you were talking about the 
case where a user wants to discover someone elses certificate; since a 
certificate is included in the credential i initially found this confusing

> If the identity asserted by the Authentication Service
>    does not match the identity requests, t

what are "identitiy requests"?

in this section, you need to say more. discuss when a ua might initiate 
the subscription, how long it will hold it. Need ot mention the 
keepalive issue. Also mention that these subscriptions won't generally 
be challenged.

> which will result in a message containing both an
>    application/pkix-cert body and an application/pkcs8 body that has the
>    associated private key information for the certificate. 

I think you mean that the notifications will contain bodies of this type.

> credential that contains both an
>    application/pkix-cert and an application/pkcs8 body.

MUST contain

>  may be configured with a specific name for the Credential Server;
>    otherwise it defaults to the name of the domain in the User's AOR.

is this a new dns usage? I think you mean that we do rfc3263 
resolutionon a sip uri

> The TLS connection MUST present a certificate that matches the
>    expected name for the credential server, so that the UA knows it is
>    talking to the correct server.

this is not detailed enough. anyway i think it is covered in rfc3261, 
you should reference that

section 6 blends together a discussion of handling of credentials and 
certs. For example, it is unclear from the last paragraph on page 5 
whether or not the server digest challenges requests for certs (it 
should not)

> Otherwise it sets up a subscription and forms a NOTIFY with the
>    certificate in the body and the From header field value set to the
>    request URI of the SUBSCRIBE. 

how does this work if the request is proxied or retargeted in any way?

> It MUST send this NOTIFY through an
>    Authentication Service (as described in Identity [2]) or implement an
>    Authentication Service itself.

it can't just do that; the notifies are sent within the dialog. The 
subscribe request has to be routed through the authentication service.

> When a Credential Server receives a SUBSCRIBE for a credential, the
>    Server has to authenticate and authorize the UA and validate that
>    adequate transport security is being used. 

how is authorization done?

> Once the UA has authenticated with the Server, the Server can set up
>    a subscription and send a Notify message that MUST contain the
>    credentials.  This NOTIFY message is sent thought an Authorization
>    Service in the same way as the certificate subscriptions.

s/thought/through

authorization service? perhaps authentication

>  Next Alice sends a SIP MESSAGE message to Bob and can encrypt the
>    body using Bob's public key as shown below.  Thought outside the
>    scope of this document, it is worth noting that IM messages often

s/thought/though

>  Likewise, the authentication
>    of the user submitting the private key should be equivalent to or
>    higher than the key being transmitted.

its going to be digest; i'm not sure what this means

> If a UA receives a
>    certificate in a TLS connection that it cannot chain back to a well
>    known CA but is otherwise valid (i.e.  a self signed certificate), it
>    MAY ask the user if it wishes to allow this certificate to be used,

why should we allow this? seems like a recipe for trouble

> If a particular credential needs to be revoked, the new credential is
>    simply published to the Credential Server.  Every device keeping this
>    current in its cache will have a subscription to the credential and

this current? i think you mean the credential that was just revoked

 >  certificate for the identity in the Original URI.  This means that
 >       the certificate should only be indexed in the certificate cache by
 >       the value of the original URI, not by the value of all the
 >       identities found in the SubjectAltName list.

that said, i think the cert should be a valid standalone cert. i think 
we should require, at should or possibly must, that the subjectaltname 
match the identity that the authentication service will assert for that 
user. You need a section on generating self-signed certs, i think.

 > The Authentication Service validates
 >    that this message did come from the identity claimed in the From and

it won't though; it will come from the credential service which cannot 
claim it is the identity in the from

 > Only the UA that can authenticate as this user can tamper with this
 >    body, so the owner of the identity can provide a false public key but
 >    other users cannot.

or the authentication service itself, or anything with the private 
domain keys





-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (1)
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Corey,

Thanks for the great analysis of the situation and the suggestions. I agree 
almost completely with what you suggested. I'll follow this up with a few minor 
points on your suggestions. But first, some comments on your questions, which 
might help understand why this approach was taken.

In a message dated 11/11/2004 6:12:39 PM Eastern Standard Time, 
coreya@nortelnetworks.com writes:


> 1. Why support multiple namespace/priority (n-p) values in a single 
> message, in one or more R-P headers?  And if supported, should not the 
> n-p values in the message equate to the same priority level (for 
> example, dsn.flash and q735.1 would be equal)?  Would it not be better 
> to support only one n-p value per message, and recommend that a 
> proxy/gateway between networks supporting two different namespaces 
> handle mapping n-p values?
> ---- There may be/are networks which would support multiple levels.  The 
> draft is not dictating that there must be multiple levels.
> ---- As for having the n-p levels be "equal", James cited a GETS call 
> with a specific precedence which becomes a Routine call on the DSN.
> ---- Having a proxy/gateway mapped between namespaces is an viable 
> alternative.
> 


While it is a little hard to describe specific cases for the use of multiple 
namespaces, I believe it could be used, such as a GETS call originated in a 
military network that normally uses only the dsn namespace. The caller, knowing 
that the call will likely go outside the dsn, might be able to specify both 
dsn=flash and ETS=3. It is not possible to say that they "equal", since they are 
two separately defined lists of priorities which have no direct relationship. 
In one application, only the dsn namespace would have an effect while the 
call remains within the private network (DOD) for which it was defined. When the 
call gets to a PSTN gateway to jump into the "public" network, the dsn 
namespace/priority is discarded, since the "public" network does not support it. At 
the GW, the ETS namespace/priority is converted to the appropriate PSTN 
signaling which that netwrok supports. Note that in this case, the PSTN signaling 
does not have any further signaling of this value beyond the initial call setup 
message, which is why the SIP signaling would not include this R-P header in 
anything beyond the INVITE.

When an ETS call from the public network enters the dsn, it could be mapped 
to Routine or to some higher priority level if the PSTN signaling provided that 
information. That will be determine by the administrator of the network and 
what the GW is told to do. It certainly can't be defined in the R-P header 
draft.

Another application is a gateway between two networks which each use the R-P 
header but with different definitions. Lets say between the US DOD network 
with 5 values and another country's military network when that other country uses 
only 3 levels. The GW must map between the two sets according to a mutually 
agreed mapping. In each network, only one R-P header is used.

I believe it was agreed long ago that the R-P header draft should not try to 
define the interoperation/mapping between different namespaces. I think this 
approach should remain, since any attempt to define relationships would be 
overly complex and may conflict with what network operators want to do.

The first example above, with two namespaces being carried, is the reason why 
the required mode of operation in the dsn is that unknown namespaces are 
simply ignored and passed on. Known namespaces with unknown priorities need to be 
treated as errors, but not necessarily by failing a call.

Mike Pierce
Artel


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>Corey,
<BR>
<BR>Thanks for the great analysis of the situation and the suggestions. I ag=
ree almost completely with what you suggested. I'll follow this up with a fe=
w minor points on your suggestions. But first, some comments on your questio=
ns, which might help understand why this approach was taken.
<BR>
<BR>In a message dated 11/11/2004 6:12:39 PM Eastern Standard Time, coreya@n=
ortelnetworks.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">1. Why support multiple nam=
espace/priority (n-p) values in a single=20
<BR>message, in one or more R-P headers? &nbsp;And if supported, should not=20=
the=20
<BR>n-p values in the message equate to the same priority level (for=20
<BR>example, dsn.flash and q735.1 would be equal)? &nbsp;Would it not be bet=
ter=20
<BR>to support only one n-p value per message, and recommend that a=20
<BR>proxy/gateway between networks supporting two different namespaces=20
<BR>handle mapping n-p values?
<BR>---- There may be/are networks which would support multiple levels. &nbs=
p;The=20
<BR>draft is not dictating that there must be multiple levels.
<BR>---- As for having the n-p levels be "equal", James cited a GETS call=20
<BR>with a specific precedence which becomes a Routine call on the DSN.
<BR>---- Having a proxy/gateway mapped between namespaces is an viable=20
<BR>alternative.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>While it is a little hard to describe specific cases for the use of mult=
iple namespaces, I believe it could be used, such as a GETS call originated=20=
in a military network that normally uses only the dsn namespace. The caller,=
 knowing that the call will likely go outside the dsn, might be able to spec=
ify both dsn=3Dflash and ETS=3D3. It is not possible to say that they "equal=
", since they are two separately defined lists of priorities which have no d=
irect relationship. In one application, only the dsn namespace would have an=
 effect while the call remains within the private network (DOD) for which it=
 was defined. When the call gets to a PSTN gateway to jump into the "public"=
 network, the dsn namespace/priority is discarded, since the "public" networ=
k does not support it. At the GW, the ETS namespace/priority is converted to=
 the appropriate PSTN signaling which that netwrok supports. Note that in th=
is case, the PSTN signaling does not have any further signaling of this valu=
e beyond the initial call setup message, which is why the SIP signaling woul=
d not include this R-P header in anything beyond the INVITE.
<BR>
<BR>When an ETS call from the public network enters the dsn, it could be map=
ped to Routine or to some higher priority level if the PSTN signaling provid=
ed that information. That will be determine by the administrator of the netw=
ork and what the GW is told to do. It certainly can't be defined in the R-P=20=
header draft.
<BR>
<BR>Another application is a gateway between two networks which each use the=
 R-P header but with different definitions. Lets say between the US DOD netw=
ork with 5 values and another country's military network when that other cou=
ntry uses only 3 levels. The GW must map between the two sets according to a=
 mutually agreed mapping. In each network, only one R-P header is used.
<BR>
<BR>I believe it was agreed long ago that the R-P header draft should not tr=
y to define the interoperation/mapping between different namespaces. I think=
 this approach should remain, since any attempt to define relationships woul=
d be overly complex and may conflict with what network operators want to do.
<BR>
<BR>The first example above, with two namespaces being carried, is the reaso=
n why the required mode of operation in the dsn is that unknown namespaces a=
re simply ignored and passed on. Known namespaces with unknown priorities ne=
ed to be treated as errors, but not necessarily by failing a call.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR></FONT></HTML>

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From sip-bounces@ietf.org  Fri Nov 12 10:17:39 2004
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From: "Spencer Dawkins" <spencer@mcsr-labs.org>
To: "Christian Jansson" <christian.jansson@hotsip.com>,
        "Christian Stredicke" <Christian.Stredicke@snom.de>
References: <B7192C0D8D60754DADA9E22294C573696316D3@mailserver.hotsip.com>
Subject: Re: [Sip] PING/PONG [bcc][faked-from][heur] [mx][spf]
Date: Fri, 12 Nov 2004 10:06:30 -0500
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Hi, Christian (J),

Ah, your note was very helpful. thank you for continuing to follow up 
when we weren't together yet. I've been responding to a vague uneasy 
feeling, and this note helped me understand my own concerns.

The behavior of applications potentially attempting to set up TCP 
connections very aggressively is what concerns me. Think, "close the 
connection, wait some minimal period of time before trying to reopen 
it, what's the minimum period of time, and is there a backoff at the 
application level?"

I would be concerned about large numbers of clients reopening TCP 
connections through a NAT (for instance) trying to restart.

I agree with the rest of what we're talking about here.

Spencer

----- Original Message ----- 
From: "Christian Jansson" <christian.jansson@hotsip.com>
To: "Spencer Dawkins" <spencer@mcsr-labs.org>; "Christian Stredicke" 
<Christian.Stredicke@snom.de>
Cc: <sip@ietf.org>
Sent: Friday, November 12, 2004 4:00 AM
Subject: RE: [Sip] PING/PONG [bcc][faked-from][heur] [mx][spf]



>From: Spencer Dawkins [mailto:spencer@mcsr-labs.org]
>
>OK, I'm still confused here. Please help me.
>
>The apparent choices include "Keep-alives for TCP should be CRLF
>(default-interval TBD)".
>
>Sending keep-alives over TCP may be interesting, but I'm really
>confused about what you think is a reasonable action when you send a
>PING over TCP and don't get a PONG back.


When sending the CRLF over TCP you should not get anything back if the
connection is still there. If the TCP connection is down or fails 
while
sending the CRLF on the other hand you will get a notice from your IP
stack of the failure. Then that socket should be closed, and the UA 
may
try to set up a new connection if wants to.


>
>If you change a status indicator on a user display that says "may 
>have
>lost connectivity", that's fine. If you try to retransmit requests,
>I'm concerned.
>
>My definition of "reasonable" includes "don't send multiple new
>packets into a network path that is not returning PONGs, when the
>problem may be that the network path is losing packets because it is
>already congested." Retransmitting requests requires exactly this
>(tear down existing TCP connection, because it's still 
>retransmitting,
>open a new TCP connection, and then retransmit the request at the
>application level).
>
>There is no way for the endpoints to know that loss is NOT due to
>congestion, and putting TCP-based "lost packet multipliers" in place
>in end systems that can't know they aren't facing congestion is not
>something I'm comfortable with.


I'm not sure how often keep-alives have to be sent over TCP to ensure
that NAT/FWs do not remove the state. Preferabely NAT/FWs never remove
the state, but I have heard that it happens. Does anyone have a
reference to test statistics of NAT/FWs behavior? The keep-alive 
default
interval should probably be set just below the shortest of all 
detected
values. My guess is that the value will be around 5 minutes. Sending a
TCP packet with CRLF every 5 minutes can't really be a big source for
congestion. The keep-alives would also not be sent if any other 
traffic
had been sent to or from the UA.

>
>If TCP isn't the right answer, using TCP and sending multiple new TCP
>packets when a connection won't PING/PONG isn't helpful.

Given that the keep-alives are spaced with a couple of minutes, and 
that
this interval is much greater than the TCP error timeouts, I do not 
see
the problem. If those keep-alives would have been sent every 10 
seconds
I would have agreed on the problems you describe.

/ Christian Jansson




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From sip-bounces@ietf.org  Fri Nov 12 10:55:52 2004
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Multiple namespaces.

It is highly likely that an NS/EP (National Security/Emergency
Preparedness) call will carry both the ets and wps namespaces.  This is
because you can not determine ahead of time whether the call will progress
to a WIRELINE gateway (in which case the mapping associated with the ets
namespace is needed) or to a WIRELESS gateway (in which case the mapping
associated with the wps namespace is needed).  We initially tired to define
a single namespace that would work for both situations, but were unable to,
given the constraints on namespace structure.

There is also the POSSIBILITY of a call (within one of the military
networks) carrying both the ets namespace and one of the MLPP based
namespaces (especially a call that is expected to terminate on the Public
network where the MLPP based namespace could be dropped while the ets
namespace is retained).  But this situation is still hypothetical.

Janet

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                      Mpierce1                                                                                                 
                      @aol.com                 To:      coreya@nortelnetworks.com, sip@ietf.org                                
                      Sent by:                 cc:                                                                             
                      sip-bounces              Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (1)  
                                                                                                                               
                                                                                                                               
                      11/12/2004 09:53                                                                                         
                      AM                                                                                                       
                                                                                                                               
                                                                                                                               




Corey,

Thanks for the great analysis of the situation and the suggestions. I agree
almost completely with what you suggested. I'll follow this up with a few
minor points on your suggestions. But first, some comments on your
questions, which might help understand why this approach was taken.

In a message dated 11/11/2004 6:12:39 PM Eastern Standard Time,
coreya@nortelnetworks.com writes:


 1. Why support multiple namespace/priority (n-p) values in a single
 message, in one or more R-P headers?  And if supported, should not the
 n-p values in the message equate to the same priority level (for
 example, dsn.flash and q735.1 would be equal)?  Would it not be better
 to support only one n-p value per message, and recommend that a
 proxy/gateway between networks supporting two different namespaces
 handle mapping n-p values?
 ---- There may be/are networks which would support multiple levels.  The
 draft is not dictating that there must be multiple levels.
 ---- As for having the n-p levels be "equal", James cited a GETS call
 with a specific precedence which becomes a Routine call on the DSN.
 ---- Having a proxy/gateway mapped between namespaces is an viable
 alternative.



While it is a little hard to describe specific cases for the use of
multiple namespaces, I believe it could be used, such as a GETS call
originated in a military network that normally uses only the dsn namespace.
The caller, knowing that the call will likely go outside the dsn, might be
able to specify both dsn=flash and ETS=3. It is not possible to say that
they "equal", since they are two separately defined lists of priorities
which have no direct relationship. In one application, only the dsn
namespace would have an effect while the call remains within the private
network (DOD) for which it was defined. When the call gets to a PSTN
gateway to jump into the "public" network, the dsn namespace/priority is
discarded, since the "public" network does not support it. At the GW, the
ETS namespace/priority is converted to the appropriate PSTN signaling which
that netwrok supports. Note that in this case, the PSTN signaling does not
have any further signaling of this value beyond the initial call setup
message, which is why the SIP signaling would not include this R-P header
in anything beyond the INVITE.

When an ETS call from the public network enters the dsn, it could be mapped
to Routine or to some higher priority level if the PSTN signaling provided
that information. That will be determine by the administrator of the
network and what the GW is told to do. It certainly can't be defined in the
R-P header draft.

Another application is a gateway between two networks which each use the
R-P header but with different definitions. Lets say between the US DOD
network with 5 values and another country's military network when that
other country uses only 3 levels. The GW must map between the two sets
according to a mutually agreed mapping. In each network, only one R-P
header is used.

I believe it was agreed long ago that the R-P header draft should not try
to define the interoperation/mapping between different namespaces. I think
this approach should remain, since any attempt to define relationships
would be overly complex and may conflict with what network operators want
to do.

The first example above, with two namespaces being carried, is the reason
why the required mode of operation in the dsn is that unknown namespaces
are simply ignored and passed on. Known namespaces with unknown priorities
need to be treated as errors, but not necessarily by failing a call.

Mike Pierce
Artel _______________________________________________
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In a message dated 11/12/2004 11:58:14 AM Eastern Standard Time, 
An.Nguyen@ncs.gov writes:


> Can we just add a sentence to the draft saying mapping between namespaces 
> is out of scope and it should be addressed separately?

Yes, such a statement would be fine, but it should be along with a statement 
that all interrelationship between namespaces is out of scope.

Mike


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/12/2004 11:58:14 AM Eastern Standard Time, An.Nguyen@ncs.gov writes=
:
<BR>
<BR>
<BR></FONT><FONT  COLOR=3D"#0000ff" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARG=
IN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Can we just add a senten=
ce to the draft saying mapping between namespaces is out of scope and it sho=
uld be addressed separately?</FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff"=
 style=3D"BACKGROUND-COLOR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERI=
F" FACE=3D"Arial" LANG=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>Yes, such a statement would be fine, but it should be along with a state=
ment that all interrelationship between namespaces is out of scope.
<BR>
<BR>Mike
<BR></FONT></HTML>

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In a message dated 11/11/2004 6:12:39 PM Eastern Standard Time, 
coreya@nortelnetworks.com writes:


> 2. Is it wise or valid to explicitly allow a proxy to "upgrade" or 
> "downgrade" the  n-p values in a message?
> ---- I'm coming at this from the DSN perspective, but wps and/or ets may 
> allow this kind of thing.  While the DSN/DSRN/(Q735) would not, that 
> does not invalidate the concept.
> 


If one understands the "proxy" to be the entity which provides call 
processing logic and which validates the calling party's authority to use the priority 
level which they signaled in the INVITE (see 
draft-pierce-tsvwg-assured-service-arch-01), then it should be valid for that proxy to "downgrade" the signaled 
value to the level allowed for that user before forwarding the INVITE rather 
than blocking the call. This behavior should not be disallowed by the R-P 
header draft for any namespace.

I can't think of any "upgrading" case, but there is not reason for the R-P 
header draft to disallow it.

Mike Pierce
Artel

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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/11/2004 6:12:39 PM Eastern Standard Time, coreya@nortelnetworks.com=
 writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">2. Is it wise or valid to e=
xplicitly allow a proxy to "upgrade" or=20
<BR>"downgrade" the &nbsp;n-p values in a message?
<BR>---- I'm coming at this from the DSN perspective, but wps and/or ets may=
=20
<BR>allow this kind of thing. &nbsp;While the DSN/DSRN/(Q735) would not, tha=
t=20
<BR>does not invalidate the concept.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>If one understands the "proxy" to be the entity which provides call proc=
essing logic and which validates the calling party's authority to use the pr=
iority level which they signaled in the INVITE (see draft-pierce-tsvwg-assur=
ed-service-arch-01), then it should be valid for that proxy to "downgrade" t=
he signaled value to the level allowed for that user before forwarding the I=
NVITE rather than blocking the call. This behavior should not be disallowed=20=
by the R-P header draft for any namespace.
<BR>
<BR>I can't think of any "upgrading" case, but there is not reason for the R=
-P header draft to disallow it.
<BR>
<BR>Mike Pierce
<BR>Artel</FONT></HTML>

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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
--===============0517376043==--



From sip-bounces@ietf.org  Sat Nov 13 11:24:01 2004
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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (2)
Date: Sat, 13 Nov 2004 11:09:37 -0500
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On Nov 12, 2004, at 7:30 PM, Mpierce1@aol.com wrote:

> In a message dated 11/11/2004 6:12:39 PM Eastern Standard Time,=20
> coreya@nortelnetworks.com writes:
>
>
> 2. Is it wise or valid to explicitly allow a proxy to "upgrade" or
>  "downgrade" the =A0n-p values in a message?
> ---- I'm coming at this from the DSN perspective, but wps and/or ets=20=

> may
>  allow this kind of thing. =A0While the DSN/DSRN/(Q735) would not, =
that
>  does not invalidate the concept.
>
>
>
> If one understands the "proxy" to be the entity which provides call=20
> processing logic and which validates the calling party's authority to=20=

> use the priority level which they signaled in the INVITE (see=20
> draft-pierce-tsvwg-assured-service-arch-01), then it should be valid=20=

> for that proxy to "downgrade" the signaled value to the level allowed=20=

> for that user before forwarding the INVITE rather than blocking the=20
> call. This behavior should not be disallowed by the R-P header draft=20=

> for any namespace.
>
As long as you have the concept of namespaces, with different policies=20=

for different namespaces, it follows logically to allow such policies.=20=

If someone wants a namespace that has a "reject call rather than=20
downgrade" as its policy, why prohibit it? Or are you arguing that the=20=

R-P draft should be silent on upgrade/downgrade/reject other than to=20
point out that the allowed behaviors are part of the namespace policy=20
definition? If that's what you're saying, I'm ok with that.

Dave.


> I can't think of any "upgrading" case, but there is not reason for the=20=

> R-P header draft to disallow it.
>
> Mike Pierce
> Artel_______________________________________________
> 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
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (2)
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In a message dated 11/13/2004 11:09:54 AM Eastern Standard Time, 
oran@cisco.com writes:


> If someone wants a namespace that has a "reject call rather than 
> downgrade" as its policy, why prohibit it? Or are you arguing that the 
> R-P draft should be silent on upgrade/downgrade/reject other than to 
> point out that the allowed behaviors are part of the namespace policy 
> definition? If that's what you're saying, I'm ok with that.
> 


Absolutely. Rejecting a call for this reason is perfectly okay. The R-P 
document should be silent on these types of issues, or, at the most, can mention 
the different types of behavior which "MAY" be used, but not attempt to 
associate each possible behavior with various namespaces.

Mike


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/13/2004 11:09:54 AM Eastern Standard Time, oran@cisco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">If someone wants a namespac=
e that has a "reject call rather than=20
<BR>downgrade" as its policy, why prohibit it? Or are you arguing that the=20
<BR>R-P draft should be silent on upgrade/downgrade/reject other than to=20
<BR>point out that the allowed behaviors are part of the namespace policy=20
<BR>definition? If that's what you're saying, I'm ok with that.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>Absolutely. Rejecting a call for this reason is perfectly okay. The R-P=20=
document should be silent on these types of issues, or, at the most, can men=
tion the different types of behavior which "MAY" be used, but not attempt to=
 associate each possible behavior with various namespaces.
<BR>
<BR>Mike
<BR></FONT></HTML>

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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
--===============0937479078==--



From sip-bounces@ietf.org  Mon Nov 15 10:00:41 2004
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Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (2)
To: David R Oran <oran@cisco.com>
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As I understand it, the proxy would only be allowed to upgrade or downg=
rade
the value if that were the behavior specifically identified for THAT
namespace.

As far as I know, neither ets nor wps have current requirements to do t=
hat.
But the circuit switched versions DO have the concept of a "default
priority value" under certain circumstances.  If that concept were appl=
ied
to the IP side, it MIGHT involve upgrading or downgrading to the defaul=
t
level.

I am comfortable with it being left in as a possible behavior, if and o=
nly
if that behavior is explicitly specified for that namespace.

Janet



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                                                                       =
                                          
                      David R Oran                                     =
                                          
                      <oran                    To:      Mpierce1@aol.co=
m                                         
                      @cisco.com>              cc:      sip@ietf.org   =
                                          
                      Sent by:                 Subject: Re: [Sip] -reso=
urce-priority-05.txt: Comments and        
                      sip-bounces              Recommendations (2)     =
                                          
                                                                       =
                                          
                                                                       =
                                          
                      11/13/2004 11:09                                 =
                                          
                      AM                                               =
                                          
                                                                       =
                                          
                                                                       =
                                          





On Nov 12, 2004, at 7:30 PM, Mpierce1@aol.com wrote:

> In a message dated 11/11/2004 6:12:39 PM Eastern Standard Time,
> coreya@nortelnetworks.com writes:
>
>
> 2. Is it wise or valid to explicitly allow a proxy to "upgrade" or
>  "downgrade" the =A0n-p values in a message?
> ---- I'm coming at this from the DSN perspective, but wps and/or ets
> may
>  allow this kind of thing. =A0While the DSN/DSRN/(Q735) would not, th=
at
>  does not invalidate the concept.
>
>
>
> If one understands the "proxy" to be the entity which provides call
> processing logic and which validates the calling party's authority to=

> use the priority level which they signaled in the INVITE (see
> draft-pierce-tsvwg-assured-service-arch-01), then it should be valid
> for that proxy to "downgrade" the signaled value to the level allowed=

> for that user before forwarding the INVITE rather than blocking the
> call. This behavior should not be disallowed by the R-P header draft
> for any namespace.
>
As long as you have the concept of namespaces, with different policies
for different namespaces, it follows logically to allow such policies.
If someone wants a namespace that has a "reject call rather than
downgrade" as its policy, why prohibit it? Or are you arguing that the
R-P draft should be silent on upgrade/downgrade/reject other than to
point out that the allowed behaviors are part of the namespace policy
definition? If that's what you're saying, I'm ok with that.

Dave.


> I can't think of any "upgrading" case, but there is not reason for th=
e
> R-P header draft to disallow it.
>
> Mike Pierce
> Artel_______________________________________________
> 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
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.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


=



_______________________________________________
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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Date: Mon, 15 Nov 2004 12:52:26 -0600
To: Mpierce1@aol.com, coreya@nortelnetworks.com, sip@ietf.org
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (2)
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specifying which proxy does any header changes is a market requirement - 
not a protocol requirement - therefore will not be included in the document

At 07:30 PM 11/12/2004 -0500, Mpierce1@aol.com wrote:
>In a message dated 11/11/2004 6:12:39 PM Eastern Standard Time, 
>coreya@nortelnetworks.com writes:
>
>
>>2. Is it wise or valid to explicitly allow a proxy to "upgrade" or
>>"downgrade" the  n-p values in a message?
>>---- I'm coming at this from the DSN perspective, but wps and/or ets may
>>allow this kind of thing.  While the DSN/DSRN/(Q735) would not, that
>>does not invalidate the concept.
>
>
>
>If one understands the "proxy" to be the entity which provides call 
>processing logic and which validates the calling party's authority to use 
>the priority level which they signaled in the INVITE (see 
>draft-pierce-tsvwg-assured-service-arch-01), then it should be valid for 
>that proxy to "downgrade" the signaled value to the level allowed for that 
>user before forwarding the INVITE rather than blocking the call. This 
>behavior should not be disallowed by the R-P header draft for any namespace.
>
>I can't think of any "upgrading" case, but there is not reason for the R-P 
>header draft to disallow it.
>
>Mike Pierce
>Artel
>_______________________________________________
>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


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


_______________________________________________
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: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (1)
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Where is "mapping between namespaces" specified in the document now?

I do not believe there is such a sentence/comment/statement, and there 
should not be one, so why was this brought up?

At 03:22 PM 11/12/2004 -0500, Mpierce1@aol.com wrote:
>In a message dated 11/12/2004 11:58:14 AM Eastern Standard Time, 
>An.Nguyen@ncs.gov writes:
>
>
>>Can we just add a sentence to the draft saying mapping between namespaces 
>>is out of scope and it should be addressed separately?
>
>
>Yes, such a statement would be fine, but it should be along with a 
>statement that all interrelationship between namespaces is out of scope.
>
>Mike
>_______________________________________________
>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


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


_______________________________________________
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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	<Mpierce1@aol.com>,
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Subject: RE: [Sip] -resource-priority-05.txt: Comments and  Recommendation
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It's not.  This was one of my questions to you Tuesday, which I shared in my
post.  Given my perspective at the time, I saw this as an alternative to
passing multiple namespace/priority values in a single message.  After
speaking with you, I understood that there may be instances where multiple
values are perfectly acceptable.  This does not rule out such a mapping -
I'd argue that they will have to be defined when implementing this header,
if it will be able to pass across namespace boundaries - but as everyone is
stating, such mappings should not be defined with the header.

Corey W. Alexander

-----Original Message-----
From: James M. Polk [mailto:jmpolk@cisco.com] 
Sent: Monday, November 15, 2004 1:01 PM
To: Mpierce1@aol.com; An.Nguyen@ncs.gov; Alexander, Corey [NGD:X262:EXCH];
sip@ietf.org
Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations
(1)


Where is "mapping between namespaces" specified in the document now?

I do not believe there is such a sentence/comment/statement, and there 
should not be one, so why was this brought up?

At 03:22 PM 11/12/2004 -0500, Mpierce1@aol.com wrote:
>In a message dated 11/12/2004 11:58:14 AM Eastern Standard Time,
>An.Nguyen@ncs.gov writes:
>
>
>>Can we just add a sentence to the draft saying mapping between 
>>namespaces
>>is out of scope and it should be addressed separately?
>
>
>Yes, such a statement would be fine, but it should be along with a
>statement that all interrelationship between namespaces is out of scope.
>
>Mike
>_______________________________________________
>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


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented



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<P><FONT SIZE=3D2>It's not.&nbsp; This was one of my questions to you =
Tuesday, which I shared in my post.&nbsp; Given my perspective at the =
time, I saw this as an alternative to passing multiple =
namespace/priority values in a single message.&nbsp; After speaking =
with you, I understood that there may be instances where multiple =
values are perfectly acceptable.&nbsp; This does not rule out such a =
mapping - I'd argue that they will have to be defined when implementing =
this header, if it will be able to pass across namespace boundaries - =
but as everyone is stating, such mappings should not be defined with =
the header.</FONT></P>

<P><FONT SIZE=3D2>Corey W. Alexander</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: James M. Polk [<A =
HREF=3D"mailto:jmpolk@cisco.com">mailto:jmpolk@cisco.com</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: Monday, November 15, 2004 1:01 PM</FONT>
<BR><FONT SIZE=3D2>To: Mpierce1@aol.com; An.Nguyen@ncs.gov; Alexander, =
Corey [NGD:X262:EXCH]; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: Re: [Sip] -resource-priority-05.txt: =
Comments and Recommendations (1)</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Where is &quot;mapping between namespaces&quot; =
specified in the document now?</FONT>
</P>

<P><FONT SIZE=3D2>I do not believe there is such a =
sentence/comment/statement, and there </FONT>
<BR><FONT SIZE=3D2>should not be one, so why was this brought =
up?</FONT>
</P>

<P><FONT SIZE=3D2>At 03:22 PM 11/12/2004 -0500, Mpierce1@aol.com =
wrote:</FONT>
<BR><FONT SIZE=3D2>&gt;In a message dated 11/12/2004 11:58:14 AM =
Eastern Standard Time,</FONT>
<BR><FONT SIZE=3D2>&gt;An.Nguyen@ncs.gov writes:</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;Can we just add a sentence to the draft =
saying mapping between </FONT>
<BR><FONT SIZE=3D2>&gt;&gt;namespaces</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;is out of scope and it should be addressed =
separately?</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;Yes, such a statement would be fine, but it =
should be along with a</FONT>
<BR><FONT SIZE=3D2>&gt;statement that all interrelationship between =
namespaces is out of scope.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;Mike</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 Use </FONT>
<BR><FONT SIZE=3D2>&gt;sipping@ietf.org for new developments on the =
application of sip</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>cheers,</FONT>
<BR><FONT SIZE=3D2>James</FONT>
</P>

<P><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
*******************</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Truth is not to be argued... it is to =
be presented</FONT>
</P>
<BR>

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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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--===============0527853555==--



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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (1)
Date: Mon, 15 Nov 2004 14:43:34 -0500
To: "James M. Polk" <jmpolk@cisco.com>
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On Nov 15, 2004, at 2:01 PM, James M. Polk wrote:

> Where is "mapping between namespaces" specified in the document now?
>
> I do not believe there is such a sentence/comment/statement, and there 
> should not be one, so why was this brought up?
>
If somebody reads the doc expecting to see something and it isn't 
there, two possibilities obtain:
(a) they don't "get it"
(b) a reasonable person might think that function a natural one to be 
handled, but that's not the approach we took in designing this beastie.

The case at hand seems closer to (b) than (a), so I see no harm in 
saying that namespaces are orthogonal as far as the specification is 
concerned, and while it may be possible to map behavior between 
namespaces, this document doesn't tell you how, and that's intentional.

Dave.

> At 03:22 PM 11/12/2004 -0500, Mpierce1@aol.com wrote:
>> In a message dated 11/12/2004 11:58:14 AM Eastern Standard Time, 
>> An.Nguyen@ncs.gov writes:
>>
>>
>>> Can we just add a sentence to the draft saying mapping between 
>>> namespaces is out of scope and it should be addressed separately?
>>
>>
>> Yes, such a statement would be fine, but it should be along with a 
>> statement that all interrelationship between namespaces is out of 
>> scope.
>>
>> Mike
>> _______________________________________________
>> 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
>
>
> cheers,
> James
>
>                                *******************
>                 Truth is not to be argued... it is to be presented
>
>
> _______________________________________________
> 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
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.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


From sip-bounces@ietf.org  Mon Nov 15 15:39:36 2004
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The .03 draft (and I think also the .04 draft) had a paragraph that said
   "Namespaces do not describe how they relate to other existing
   namespaces, as each namespace is independent of other registrations."

It doesn't seem to be in .05

Maybe it, or something like it, should be put back in.

Janet


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                      David R Oran                                                                               
                      <oran                    To:      "James M. Polk" <jmpolk@cisco.com>                       
                      @cisco.com>              cc:      An.Nguyen@ncs.gov, sip@ietf.org, Mpierce1@aol.com        
                      Sent by:                 Subject: Re: [Sip] -resource-priority-05.txt: Comments and        
                      sip-bounces              Recommendations (1)                                               
                                                                                                                 
                                                                                                                 
                      11/15/2004 02:43                                                                           
                      PM                                                                                         
                                                                                                                 
                                                                                                                 





On Nov 15, 2004, at 2:01 PM, James M. Polk wrote:

> Where is "mapping between namespaces" specified in the document now?
>
> I do not believe there is such a sentence/comment/statement, and there
> should not be one, so why was this brought up?
>
If somebody reads the doc expecting to see something and it isn't
there, two possibilities obtain:
(a) they don't "get it"
(b) a reasonable person might think that function a natural one to be
handled, but that's not the approach we took in designing this beastie.

The case at hand seems closer to (b) than (a), so I see no harm in
saying that namespaces are orthogonal as far as the specification is
concerned, and while it may be possible to map behavior between
namespaces, this document doesn't tell you how, and that's intentional.

Dave.

> At 03:22 PM 11/12/2004 -0500, Mpierce1@aol.com wrote:
>> In a message dated 11/12/2004 11:58:14 AM Eastern Standard Time,
>> An.Nguyen@ncs.gov writes:
>>
>>
>>> Can we just add a sentence to the draft saying mapping between
>>> namespaces is out of scope and it should be addressed separately?
>>
>>
>> Yes, such a statement would be fine, but it should be along with a
>> statement that all interrelationship between namespaces is out of
>> scope.
>>
>> Mike
>> _______________________________________________
>> 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
>
>
> cheers,
> James
>
>                                *******************
>                 Truth is not to be argued... it is to be presented
>
>
> _______________________________________________
> 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
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.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





_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Mon Nov 15 16:35:06 2004
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	Mon, 15 Nov 2004 21:46:58 +0100
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Date: Mon, 15 Nov 2004 21:46:54 +0100
From: Klaus Darilion <klaus.mailinglists@pernau.at>
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To: Christian Stredicke <Christian.Stredicke@snom.de>
Subject: Re: [Sip] PING/PONG
References: <B52FDDEC7CBE9D40B36FE900C9AD78B4176AB8@merenge.intern.snom.de>
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You should also consider the guidelines in the draft
http://www.ietf.org/internet-drafts/draft-ietf-behave-nat-00.txt
which will hopefully soon implemented in NAT routers.

regards,
klaus

Christian Stredicke wrote:

> Hi Spencer!
> 
> You just mentioned another way of keeping TCP connections alive. Thats
> good! 
> 
> So far we have:
> 
> - CRLF (frequenly used today, but not indicated by user agent)
> - STUN (same as CRLF)
> - PING/PONG or whatever (tbd, maybe superflous)
> - OPTIONS (big overhead, but compatible)
> - REGISTER (same as OPTIONS, but even bigger overhead)
> - TCP low-level (how can this be addressed by the application?)
> - No refreshing ("legacy device")
> 
> Lets have a way to indicate these options and let the SBC decide what
> method it likes/supports.
> 
> CS
> 
> 
>>-----Original Message-----
>>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
>>Behalf Of Spencer Dawkins
>>Sent: Wednesday, November 10, 2004 5:01 PM
>>To: sip@ietf.org
>>Subject: Re: [Sip] PING/PONG
>>
>>Call me old-fashioned, but I'm requesting a point of clarification 
>>about PING/PONG.
>>
>>As I should have said in the working group session when we discussed 
>>this, I have no interest in chatting about UDP versus TCP. If you 
>>choose UDP, I assume you did it for a reason, and the discussion at 
>>the working group session makes sense to me.
>>
>>Having said that,
>>
>>
>>>With UDP, the PING will create a new NAT binding and the server can 
>>>update
>>>its mapping for the UA.
>>>
>>>With TCP, if you want to be able to detect that the NAT binding is 
>>>gone on
>>>the order of a transaction timeout, you would need to send a PING 
>>>every 32
>>>seconds or so. Given that NAT bindings for TCP live longer than 
>>>that, I
>>>don't see what the CRLF buys you.
>>
>>it seems to me that if you have chosen to use TCP for some 
>>reason, you 
>>need to at least consider living with TCP behavior on retries.
>>
>>The TCP protocol is extremely persistent, uses exponential backoff 
>>(typically to a maximum of 60-64 seconds per retry), and in most 
>>vanilla implementations will not declare a connection unusable for 
>>something like 10-15 minutes.
>>
>>If a path is lost, for any reason, TCP will use this behavior (the 
>>behavior that allows the Internet to survive the World Wide Web 
>>today).
>>
>>If you are using TCP and you don't like this behavior, and you don't 
>>get a response back when you wanted to get a response back, the 
>>application choices are:
>>
>>- sit on your hands and wait for the TCP connection to fail,
>>
>>- close the TCP connection and open another one, or
>>
>>- close the TCP connection, sit on your hands for some period 
>>of time, 
>>and open another one.
>>
>>The second choice (immediate retry) requires that you reset the TCP 
>>connection (one packet), send a SYN to open a second TCP connection 
>>(one packet), and potentially ACK an incoming SYN/ACK (one packet), 
>>and then resend your application message (could be piggybacked but 
>>most TCP stacks will use a separate packet).
>>
>>Immediately sending up to four new packets when you detect 
>>loss of one 
>>packet, with the possibility that the single-packet loss is due to 
>>network congestion, seems too aggressive.
>>
>>Sitting on your hands and letting TCP retry doesn't seem much worse 
>>than sitting on your hands and then setting up a new TCP connection.
>>
>>Are we assuming that network congestion is not an issue any more?
>>
>>Thanks for any guidance you can provide!
>>
>>Spencer 
>>
>>
>>
>>_______________________________________________
>>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
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Use sipping@ietf.org for new developments on the application of sip


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To: Janet P Gunn <jgunn6@csc.com>, David R Oran <oran@cisco.com>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (1)
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At 03:33 PM 11/15/2004 -0500, Janet P Gunn wrote:




>The .03 draft (and I think also the .04 draft) had a paragraph that said
>    "Namespaces do not describe how they relate to other existing
>    namespaces, as each namespace is independent of other registrations."
>
>It doesn't seem to be in .05

see (the new) section 7.1 about the warnings of multiple namespaces


>Maybe it, or something like it, should be put back in.

see above


>Janet
>
>
>----------------------------------------------------------------------------------------
>
>This is a PRIVATE message. If you are not the intended recipient, please
>delete without copying and kindly advise us by e-mail of the mistake in
>delivery. NOTE: Regardless of content, this e-mail shall not operate to
>bind CSC to any order or other contract unless pursuant to explicit written
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>for such purpose.
>----------------------------------------------------------------------------------------
>
>
>
>
> 
>
>                       David R 
> Oran 
>
>                       <oran                    To:      "James M. Polk" 
> <jmpolk@cisco.com>
>                       @cisco.com>              cc: 
> An.Nguyen@ncs.gov, sip@ietf.org, Mpierce1@aol.com
>                       Sent by:                 Subject: Re: [Sip] 
> -resource-priority-05.txt: Comments and
>                       sip-bounces              Recommendations 
> (1)
> 
>
> 
>
>                       11/15/2004 
> 02:43 
>
>                       PM 
 >
> 
>
> 
>
>
>
>
>
>
>On Nov 15, 2004, at 2:01 PM, James M. Polk wrote:
>
> > Where is "mapping between namespaces" specified in the document now?
> >
> > I do not believe there is such a sentence/comment/statement, and there
> > should not be one, so why was this brought up?
> >
>If somebody reads the doc expecting to see something and it isn't
>there, two possibilities obtain:
>(a) they don't "get it"
>(b) a reasonable person might think that function a natural one to be
>handled, but that's not the approach we took in designing this beastie.
>
>The case at hand seems closer to (b) than (a), so I see no harm in
>saying that namespaces are orthogonal as far as the specification is
>concerned, and while it may be possible to map behavior between
>namespaces, this document doesn't tell you how, and that's intentional.
>
>Dave.
>
> > At 03:22 PM 11/12/2004 -0500, Mpierce1@aol.com wrote:
> >> In a message dated 11/12/2004 11:58:14 AM Eastern Standard Time,
> >> An.Nguyen@ncs.gov writes:
> >>
> >>
> >>> Can we just add a sentence to the draft saying mapping between
> >>> namespaces is out of scope and it should be addressed separately?
> >>
> >>
> >> Yes, such a statement would be fine, but it should be along with a
> >> statement that all interrelationship between namespaces is out of
> >> scope.
> >>
> >> Mike
> >> _______________________________________________
> >> 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
> >
> >
> > cheers,
> > James
> >
> >                                *******************
> >                 Truth is not to be argued... it is to be presented
> >
> >
> > _______________________________________________
> > 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
> >
>David R. Oran
>Cisco Fellow
>Cisco Systems
>7 Ladyslipper Lane
>Acton, MA 01720 USA
>Tel: +1 978 264 2048
>Email: oran@cisco.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


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Mon Nov 15 18:00:06 2004
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Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (1)
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7.1 starts with

"The only rules stipulated here regarding more than one namespace in
   a SIP message are:"

followed by a bulleted list..

May be it could be made stronger by saying, after the list, "Except for
these rules,  namespaces do not describe how they relate to other existing
namespaces."

"Only" should cover it, but if it isn't clear to some readers...

Janet
----------------------------------------------------------------------------------------

This is a PRIVATE message. If you are not the intended recipient, please
delete without copying and kindly advise us by e-mail of the mistake in
delivery. NOTE: Regardless of content, this e-mail shall not operate to
bind CSC to any order or other contract unless pursuant to explicit written
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for such purpose.
----------------------------------------------------------------------------------------




                                                                                                                 
                      "James M. Polk"                                                                            
                      <jmpolk                  To:      Janet P Gunn/FED/CSC@CSC, David R Oran <oran@cisco.com>  
                      @cisco.com>              cc:      An.Nguyen@ncs.gov, sip@ietf.org, sip-bounces@ietf.org,   
                      Sent by:                 Mpierce1@aol.com                                                  
                      sip-bounces              Subject: Re: [Sip] -resource-priority-05.txt: Comments and        
                                               Recommendations (1)                                               
                                                                                                                 
                      11/15/2004 05:20                                                                           
                      PM                                                                                         
                                                                                                                 
                                                                                                                 




At 03:33 PM 11/15/2004 -0500, Janet P Gunn wrote:




>The .03 draft (and I think also the .04 draft) had a paragraph that said
>    "Namespaces do not describe how they relate to other existing
>    namespaces, as each namespace is independent of other registrations."
>
>It doesn't seem to be in .05

see (the new) section 7.1 about the warnings of multiple namespaces


>Maybe it, or something like it, should be put back in.

see above


>Janet
>
>
>----------------------------------------------------------------------------------------

>
>This is a PRIVATE message. If you are not the intended recipient, please
>delete without copying and kindly advise us by e-mail of the mistake in
>delivery. NOTE: Regardless of content, this e-mail shall not operate to
>bind CSC to any order or other contract unless pursuant to explicit
written
>agreement or government initiative expressly permitting the use of e-mail
>for such purpose.
>----------------------------------------------------------------------------------------

>
>
>
>
>
>
>                       David R
> Oran
>
>                       <oran                    To:      "James M. Polk"
> <jmpolk@cisco.com>
>                       @cisco.com>              cc:
> An.Nguyen@ncs.gov, sip@ietf.org, Mpierce1@aol.com
>                       Sent by:                 Subject: Re: [Sip]
> -resource-priority-05.txt: Comments and
>                       sip-bounces              Recommendations
> (1)
>
>
>
>
>                       11/15/2004
> 02:43
>
>                       PM
 >
>
>
>
>
>
>
>
>
>
>On Nov 15, 2004, at 2:01 PM, James M. Polk wrote:
>
> > Where is "mapping between namespaces" specified in the document now?
> >
> > I do not believe there is such a sentence/comment/statement, and there
> > should not be one, so why was this brought up?
> >
>If somebody reads the doc expecting to see something and it isn't
>there, two possibilities obtain:
>(a) they don't "get it"
>(b) a reasonable person might think that function a natural one to be
>handled, but that's not the approach we took in designing this beastie.
>
>The case at hand seems closer to (b) than (a), so I see no harm in
>saying that namespaces are orthogonal as far as the specification is
>concerned, and while it may be possible to map behavior between
>namespaces, this document doesn't tell you how, and that's intentional.
>
>Dave.
>
> > At 03:22 PM 11/12/2004 -0500, Mpierce1@aol.com wrote:
> >> In a message dated 11/12/2004 11:58:14 AM Eastern Standard Time,
> >> An.Nguyen@ncs.gov writes:
> >>
> >>
> >>> Can we just add a sentence to the draft saying mapping between
> >>> namespaces is out of scope and it should be addressed separately?
> >>
> >>
> >> Yes, such a statement would be fine, but it should be along with a
> >> statement that all interrelationship between namespaces is out of
> >> scope.
> >>
> >> Mike
> >> _______________________________________________
> >> 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
> >
> >
> > cheers,
> > James
> >
> >                                *******************
> >                 Truth is not to be argued... it is to be presented
> >
> >
> > _______________________________________________
> > 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
> >
>David R. Oran
>Cisco Fellow
>Cisco Systems
>7 Ladyslipper Lane
>Acton, MA 01720 USA
>Tel: +1 978 264 2048
>Email: oran@cisco.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


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


_______________________________________________
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 sip-bounces@ietf.org  Mon Nov 15 19:07:47 2004
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Date: Mon, 15 Nov 2004 18:58:22 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: [Sip] Identity after reinvite
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Jon,

After the meetings last week a question came to mind re the identity draft.

First, I want to double check on usage in dialogs. The draft doesn't 
say, but I presume that every message in a dialog needs to be signed, 
not just the one that initiates a dialog. If that is a bad assumption, 
it would be good to hear why.

Assuming that, what happens when the initial invite was from Alice to 
Bob, and later a reinvite is sent from Bob to Alice? Initially Bob knew 
that Alice had called, while Alice had no assurance about who she had 
reached with her call. The reinvite is from Bob, so if his identity is 
blessed, then Alice will know she is talking to Bob, but Bob will no 
longer have any assurance that he is still talking to Alice.

This could even manifest itself in the UI presented to Alice and Bob. 
Initially Bob would have a "callerid" display showing Alice as a 
certified caller. But after putting Alice on hold and back off hold the 
display might stop showing Alice, or at least might show the identity as 
suspect. (There may be strong temptations to avoid this via some sort of 
hack.)

I don't see any good solution for this other than solving the connected 
party id problem as well.

	Just checking my understanding,
	Paul


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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip


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To: "'Paul Kyzivat'" <pkyzivat@cisco.com>, Cullen Jennings
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Subject: [Sip] RE: Identity after reinvite
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Well, yes, now one begins to appreciate why we originally coupled the
request identity problem with the response identity problem. :)

In the next revision of the identity draft, which will incorporate the list
comments up to date, I do intend to include some text about dialogs. I agree
that the draft should apply to requests regardless of the dialog context.

Moreover, I don't think the problem you raise below is limited merely to
re-INVITEs within a dialog. I believe it applies equally to BYE requests
sent in the backwards direction in a dialog. Ideally, the originator of the
dialog should have some assurance that requests sent in the backwards
direction come from the same identity that positively responded to the
dialog (i.e., the sender of the 200 and any provisional responses).

In the absence of response identity, though, it is difficult for Alice to
have any such assurance. Equally, as you point out below, Bob can have no
assurance that his BYE or re-INVITE has been fielded by Alice if he is
unable to get a cryptographically-assured 200 response to his own requests.
Alice can't even be certain that she actually connected to Bob, thanks to
retargeting; this is the essence of the connected-ID/response identity
problem, and threatens to drag us back into the morass of retargeting
restrictions and so on that populated the sip-identity-02 draft.

A lack of such a strong binding of backwards-direction BYE requests to the
dialog is a significant limitation; if BYE can be spoofed, then an attacker
can terminate existing dialogs. However, I'm not sure that the situation is
so dire as you suggest below. Without identity, it is trivial for any
attacker to impersonate a dialog-forming request. It is harder, though, for
an attacker to impersonate a response or mid-dialog request; this requires
the attacker to gather dialog state through eavesdropping or some similar
means. There is a very important distinction in sophistication between an
attacker who can merely formulate a plausible dialog-forming request with an
inappropriate From header field versus an attacker who can capture
dialog-forming requests and improvise appropriate responses (with correct
tags, correct Call-ID/CSeq, etc).

Ultimately, request identity can protect against the former but not the
latter. While it is imperfect, I feel that this is an improvement over the
current situation. In cases where retargeting has not occurred, Alice will
see a re-INVITE or BYE from the expected URI (blessed, as you say); in cases
where retargeting has occurred, Alice may receive a warning from her UI that
an unexpected but assured URI has responded. Likewise, Bob will have to
trust, given the signature over the Contact header field in the
dialog-forming INVITE request, that his request in the backwards direction
is forwarded to the appropriate UA instance. In terms of what Bob's UI
should display, well, that really depends on the sophistication of attacks
you are willing to consider plausible. Personally, I don't think Bob's UI
should suggest any uncertainty about the matter.

To improve on all this, we will ultimately need to solve the response
identity problem; however, I maintain that sip-identity as it stands will
have addressed the simplest and most common variety of impersonation, the
one that does not require any eavesdropping whatsoever. Once we feel we have
the right answer to request identity, we can revisit the response identity
problem and try to eliminate the remaining attacks.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Monday, November 15, 2004 3:58 PM
> To: Peterson, Jon; Cullen Jennings; sip@ietf.org
> Subject: Identity after reinvite
> 
> 
> Jon,
> 
> After the meetings last week a question came to mind re the 
> identity draft.
> 
> First, I want to double check on usage in dialogs. The draft doesn't 
> say, but I presume that every message in a dialog needs to be signed, 
> not just the one that initiates a dialog. If that is a bad assumption, 
> it would be good to hear why.
> 
> Assuming that, what happens when the initial invite was from Alice to 
> Bob, and later a reinvite is sent from Bob to Alice? Initially Bob knew 
> that Alice had called, while Alice had no assurance about who she had 
> reached with her call. The reinvite is from Bob, so if his identity is 
> blessed, then Alice will know she is talking to Bob, but Bob will no 
> longer have any assurance that he is still talking to Alice.
> 
> This could even manifest itself in the UI presented to Alice and Bob. 
> Initially Bob would have a "callerid" display showing Alice as a 
> certified caller. But after putting Alice on hold and back off hold the 
> display might stop showing Alice, or at least might show the identity as 
> suspect. (There may be strong temptations to avoid this via some sort of 
> hack.)
> 
> I don't see any good solution for this other than solving the connected 
> party id problem as well.
> 
> 	Just checking my understanding,
> 	Paul
> 

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Peterson, Jon wrote:

> Well, yes, now one begins to appreciate why we originally coupled the
> request identity problem with the response identity problem. :)
> 
> In the next revision of the identity draft, which will incorporate the list
> comments up to date, I do intend to include some text about dialogs. I agree
> that the draft should apply to requests regardless of the dialog context.

I do believe there are issues related to mid-dialog requests, but I'm 
not sure I agree that Paul's characterization is the problem.

The problem is that, due to retargeting, the URI in the To field of the 
INVITE may not identify the user that was eventually reached. In RFC 
3261, requests from the called party to the caller carry the value from 
the original To field in the From. As a result, the requests will have a 
 From field value which does not actually correspond to the originator 
of the reuqest. The identity service will then reject these requests.

The solution, which rfc3261 warns of, is that a UA has to put its actual 
identity in the From field URI, not what it saw in the To field of the 
original request. This will break all compatibility with rfc2543, but 
will work with other rfc3261 compliant endpoints, since the URI is no 
longer used for dialog identification.


> 
> Moreover, I don't think the problem you raise below is limited merely to
> re-INVITEs within a dialog. I believe it applies equally to BYE requests
> sent in the backwards direction in a dialog. Ideally, the originator of the
> dialog should have some assurance that requests sent in the backwards
> direction come from the same identity that positively responded to the
> dialog (i.e., the sender of the 200 and any provisional responses).

Useful, yes, but a different problem. I like the current scope of the 
identity draft - to provide the identity of the sender of a request. 
That is not a panacea for all of our security problems, of course.

> 
> In the absence of response identity, though, it is difficult for Alice to
> have any such assurance. Equally, as you point out below, Bob can have no
> assurance that his BYE or re-INVITE has been fielded by Alice if he is
> unable to get a cryptographically-assured 200 response to his own requests.
> Alice can't even be certain that she actually connected to Bob, thanks to
> retargeting; this is the essence of the connected-ID/response identity
> problem, and threatens to drag us back into the morass of retargeting
> restrictions and so on that populated the sip-identity-02 draft.
> 
> A lack of such a strong binding of backwards-direction BYE requests to the
> dialog is a significant limitation; if BYE can be spoofed, then an attacker
> can terminate existing dialogs.

Doesn't sips prevent that? With sips, you won't be able to identify the 
called party from the 200 OK, but it will guarantee that no one except 
the party you reach can terminate the call.

Indeed, it occurs to me that one can "simulate" response identity by 
having the called party send an UPDATE or nearly any other request in 
the reverse direction immediately upon receipt of the ACK. If you 
structure the From field of that request as I have proposed above, the 
net result provides a form of response identity - it tells you who was 
reached. This doesnt tell you *why* it was reached (as redirection 
would), but it seems valuable nonetheless.


  However, I'm not sure that the situation is
> so dire as you suggest below. Without identity, it is trivial for any
> attacker to impersonate a dialog-forming request. It is harder, though, for
> an attacker to impersonate a response or mid-dialog request; this requires
> the attacker to gather dialog state through eavesdropping or some similar
> means. There is a very important distinction in sophistication between an
> attacker who can merely formulate a plausible dialog-forming request with an
> inappropriate From header field versus an attacker who can capture
> dialog-forming requests and improvise appropriate responses (with correct
> tags, correct Call-ID/CSeq, etc).

sips, of course, prevents this fully.

> 
> Ultimately, request identity can protect against the former but not the
> latter.

But, its not supposed to. Its not supposed to provide all of the 
security functions for sip - just the function which identifies the 
originator of a request.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Identity Update
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Peter,
 
Picking up on your last point (allowing From to change and allowing Identity
to be used for update within a dialog), this would avoid the need to enhance
Identity to cover an identity in a response (at least as far as dialogs are
concerned). As I write this, I see some discussion from Jon and Jonathan on
this same topic. Sounds like a good thing to explore.
 
John (john.elwell@siemens.com)

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Mataga, Peter Andrew (Peter)
Sent: 11 November 2004 06:46
To: sip@ietf.org
Subject: [Sip] Identity Update



There has been a lot of discussion in the past on the method of indicating a
change of identity to a UA, as an explicit user action at the remote UA,
because of transfer within a legacy network or by 3PCC in a B2BUA, or even
because of aliasing, forwarding or redirection+recursion within a target
domain when there is no mechanism to indicate identity in a response. At San
Diego the topic was revisited with 3 options on the table:

      1) Use of INVITE/Replaces as a kind of self-transfer

      2) Changing the From header during a dialog

      3) Inventing a new header

Based on the arguments put forward during that session, there was a hum in
favor of INVITE/Replaces as the only mechanism supported. 

 

On reflection, I don't think all the issues were aired at the time or in the
small amount of subsequent list discussion. The INVITE/Replaces mechanism
certainly has some appealing similarities to other scenarios (particularly
the pure SIP transfer case, of course). I don't know that these are
compelling when the application is to update information about a single
session, and it seems to me there are going to be cases where it is not the
most efficient or convenient way of doing so. Some specific issues:

 

1) How does the SDP in the INVITE/Replaces relate to the media session for
the original dialog? There are more possibilities than normal because the
new dialog has the same participants as the old:

      a) Same SDP session ID and version

      b) Same SDP session ID, new version

      c) New SDP session

I personally would find a) and b) a bit odd because they mean that the media
session migrates from one SIP dialog to another. Not mentioned in the
Replaces spec as far as I can see, though not ruled out either, I suppose.
I'm not sure what everyone involved in the discussions had in mind (or what
implementations might do in cases a) and b) ...).

 

2) I seem to recall claims that INVITE/Replaces would not introduce
significant loss of efficiency compared with UPDATE (or re-INVITE). If case
c) is the only reasonable one above, this may not be so, since QoS and media
encryption negotiation could be necessary. (It might even be the case that
the new request fails call admission, even though the intent is to reuse the
resources already allocated to the original call, because the entity making
the decision does not know this is a replacement.) An in-dialog update
mechanism might be able to avoid this. Indeed, with John Elwell's proposed
amendments to the relevant RFCs, there might be no mention of media session
parameters at all. I think this more lightweight procedure should at least
be possible.

 

3) INVITE/Replaces does not necessarily pass through the same intermediaries
as the original dialog. While this has to be dealt with in other scenarios
(and some in the WG appear to regard this as a Good Thing), applications
that like to "misuse" dialog stickiness will surely find it at least
convenient to be in the path of the post-identity-change session. Presumably
the answer to this is for such an application to be a B2BUA that provides
GRUU+crypto-grid Contacts such that an INV/R from either side will pass
through the application - but such a B2BUA is not transparent to the
mechanism proposed in draft-ietf-sip-identity.

 

4) The hum preference for INVITE/Replaces seemed in large part to be
motivated by difficulties with the >From header. Since there may be other
state information that will need updating in the future, it would be a bad
idea if the adoption of INVITE/Replaces for updating From were to be taken
as a precedent for any other state update, which would not be bound by this
restriction. A consistent and efficient way of updating state information is
required, and INVITE/Replaces doesn't meet the efficiency criterion. The
second hum choice was to introduce another header. This would work, at the
cost of a certain amount of redundancy with From. In particular,
draft-ietf-sip-identity would need enhancing to allow authentication of the
new header as well as (or instead of) the From header.

 

5) All of which leads back to the least-favorite hum choice, which I feel is
the best - to allow >From to change. The motivation in 3261 for keeping the
>From URI immutable was (temporary) 2543-compatibility, and there is pretty
clear language about not depending on that feature. I question whether there
are 2543-only endpoints out there that are capable of deployment in
realistic environments (that require SIPS and GRUU support for
INVITE/Replaces authorization, for example). As pointed out in San Diego,
the worst that would happen would be that a 2543-compliant UA would reject
the re-INVITE or UPDATE with a 481. If the intent is at some point to follow
through on the threat in 3261 to allow From to be changed, why not do it
now, and allow Identity to be used for update within a dialog?

 

Peter.

 

Peter Mataga

Converged Systems Division

Avaya, Inc.

 

 


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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] SIP reinvite
To: "'m. smadi'" <smadi@power.eng.mcmaster.ca>, sip@ietf.org
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This has long been a subject of discussion, most recently the message from
Peter Mataga on 11th November.

John (john.elwell@siemens.com)

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of m.
smadi
Sent: 08 November 2004 23:16
To: sip@ietf.org
Subject: [Sip] SIP reinvite

hello;

my understanding of a SIP reinvite is that it modifies the media 
session.  Is it capable of modifying the address of the send too?  for 
example of A is talking to B, can A send a reinvite to B with C's 
address instead of its own address?

Moe Smadi

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From sip-bounces@ietf.org  Tue Nov 16 09:50:21 2004
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In a message dated 11/11/2004 6:12:39 PM Eastern Standard Time, 
coreya@nortelnetworks.com writes:


> 3. With "preferential treatment" of messages based on R-P content, 
> should not scenarios be considered and listed in which a lower priority 
> message  would need to be handled before a higher priority message?  My 
> example: it would be wise to handle a BYE message with dsn.routine 
> before an INVITE with dsn.flash-override...  At least in certain 
> situations, say when the BYE is sent as a result of a preemption event 
> (which the Reason header should likely indicate).
> ---- Certainly something to consider, especially for DSN.
> 



Yes, this is an interesting idea. However, as in the SS7 world, it is a bad 
idea to intentionally do things that might change the sequence of messages. In 
the example you mention, I would expect that some entity controlling 
preemption would simply send the BYE for the routine call before it sends the INVITE 
for the new flash-override call. If two different entities send these two 
messages, it is a virtually non-existent case that both messages would happen to be 
in the same queue waiting to be sent to even allow some entity to reorder them.

What sometimes confuses this issue is what is done in SS7 (see 
draft-pierce-tsvwg-pref-treat-examples-01). SS7 has a "congestion priority level" associated 
with each signaling message, which is really a discard priority. Some 
(including me) have been confused into thinking that it changed the order to 
transmission, by sending the higher priority messages first. It does not. For the SS7 
messages corresponding to BYE and INVITE and for routine and high priority 
calls, the congestion priority leveI defined for SS7 (and the corresponding SIP 
message) are as follows:

IAM (INVITE) (routine)  0
IAM (INVITE  (priority) 1 or 2
REL (BYE)    (all)      1

This means that, if the signaling system 7 network gets congested, the new 
call setup messages (IAM) for routine calls are the first to be discarded (which 
helps to prevent catastrophic overload).

The draft mentioned about only raised this issue as something to consider to 
provide "preferential treatment" for the various signaling messages, as you 
suggested in your question. But I would only see it as being similar to the SS7 
implementation - determining which messages to discard when you have to 
discard something, not changing the order. It could use AF behavior with 3 drop 
probabilities.

My initial belief is that such a mechanism is not needed, presuming that 
other techniques can ensure the the SIP signaling traffic will never get to the 
congestion point. If not, then I think we need to consider such a mechanism, but 
it's not something that needs to be described in the R-P header draft.

Mike Pierce
Artel


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SAN=
SSERIF" FACE=3D"Arial" LANG=3D"0">In a message dated 11/11/2004 6:12:39 PM E=
astern Standard Time, coreya@nortelnetworks.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">3. With "preferential treat=
ment" of messages based on R-P content,=20
<BR>should not scenarios be considered and listed in which a lower priority=20
<BR>message &nbsp;would need to be handled before a higher priority message?=
 &nbsp;My=20
<BR>example: it would be wise to handle a BYE message with dsn.routine=20
<BR>before an INVITE with dsn.flash-override... &nbsp;At least in certain=20
<BR>situations, say when the BYE is sent as a result of a preemption event=20
<BR>(which the Reason header should likely indicate).
<BR>---- Certainly something to consider, especially for DSN.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>
<BR>Yes, this is an interesting idea. However, as in the SS7 world, it is a=20=
bad idea to intentionally do things that might change the sequence of messag=
es. In the example you mention, I would expect that some entity controlling=20=
preemption would simply send the BYE for the routine call before it sends th=
e INVITE for the new flash-override call. If two different entities send the=
se two messages, it is a virtually non-existent case that both messages woul=
d happen to be in the same queue waiting to be sent to even allow some entit=
y to reorder them.
<BR>
<BR>What sometimes confuses this issue is what is done in SS7 (see draft-pie=
rce-tsvwg-pref-treat-examples-01). SS7 has a "congestion priority level" ass=
ociated with each signaling message, which is really a discard priority. Som=
e (including me) have been confused into thinking that it changed the order=20=
to transmission, by sending the higher priority messages first. It does not.=
 For the SS7 messages corresponding to BYE and INVITE and for routine and hi=
gh priority calls, the congestion priority leveI defined for SS7 (and the co=
rresponding SIP message) are as follows:
<BR>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"FIXED" FACE=3D"Courier New" LANG=
=3D"0">IAM (INVITE) (routine) &nbsp;0
<BR>IAM (INVITE &nbsp;(priority) 1 or 2
<BR>REL (BYE) &nbsp;&nbsp;&nbsp;(all) &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;1
<BR>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">This means that, if the signaling system 7 network gets congested, th=
e new call setup messages (IAM) for routine calls are the first to be discar=
ded (which helps to prevent catastrophic overload).
<BR>
<BR>The draft mentioned about only raised this issue as something to conside=
r to provide "preferential treatment" for the various signaling messages, as=
 you suggested in your question. But I would only see it as being similar to=
 the SS7 implementation - determining which messages to discard when you hav=
e to discard something, not changing the order. It could use AF behavior wit=
h 3 drop probabilities.
<BR>
<BR>My initial belief is that such a mechanism is not needed, presuming that=
 other techniques can ensure the the SIP signaling traffic will never get to=
 the congestion point. If not, then I think we need to consider such a mecha=
nism, but it's not something that needs to be described in the R-P header dr=
aft.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR></FONT></HTML>

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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
--===============0964403507==--



From sip-bounces@ietf.org  Tue Nov 16 10:01:42 2004
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Cc: Cullen Jennings <fluffy@cisco.com>, sip@ietf.org
Subject: [Sip] Re: Identity after reinvite
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Jon,

I generally agree with what you say. I wasn't trying to suggest that we 
hold up identity for connected-party support. I just wanted to get a 
clear understanding of the issues remaining. It seems that they are a 
bit more than just not getting confirmation of who you connected to.

	Paul

Peterson, Jon wrote:
> Well, yes, now one begins to appreciate why we originally coupled the
> request identity problem with the response identity problem. :)
> 
> In the next revision of the identity draft, which will incorporate the list
> comments up to date, I do intend to include some text about dialogs. I agree
> that the draft should apply to requests regardless of the dialog context.
> 
> Moreover, I don't think the problem you raise below is limited merely to
> re-INVITEs within a dialog. I believe it applies equally to BYE requests
> sent in the backwards direction in a dialog. Ideally, the originator of the
> dialog should have some assurance that requests sent in the backwards
> direction come from the same identity that positively responded to the
> dialog (i.e., the sender of the 200 and any provisional responses).
> 
> In the absence of response identity, though, it is difficult for Alice to
> have any such assurance. Equally, as you point out below, Bob can have no
> assurance that his BYE or re-INVITE has been fielded by Alice if he is
> unable to get a cryptographically-assured 200 response to his own requests.
> Alice can't even be certain that she actually connected to Bob, thanks to
> retargeting; this is the essence of the connected-ID/response identity
> problem, and threatens to drag us back into the morass of retargeting
> restrictions and so on that populated the sip-identity-02 draft.
> 
> A lack of such a strong binding of backwards-direction BYE requests to the
> dialog is a significant limitation; if BYE can be spoofed, then an attacker
> can terminate existing dialogs. However, I'm not sure that the situation is
> so dire as you suggest below. Without identity, it is trivial for any
> attacker to impersonate a dialog-forming request. It is harder, though, for
> an attacker to impersonate a response or mid-dialog request; this requires
> the attacker to gather dialog state through eavesdropping or some similar
> means. There is a very important distinction in sophistication between an
> attacker who can merely formulate a plausible dialog-forming request with an
> inappropriate From header field versus an attacker who can capture
> dialog-forming requests and improvise appropriate responses (with correct
> tags, correct Call-ID/CSeq, etc).
> 
> Ultimately, request identity can protect against the former but not the
> latter. While it is imperfect, I feel that this is an improvement over the
> current situation. In cases where retargeting has not occurred, Alice will
> see a re-INVITE or BYE from the expected URI (blessed, as you say); in cases
> where retargeting has occurred, Alice may receive a warning from her UI that
> an unexpected but assured URI has responded. Likewise, Bob will have to
> trust, given the signature over the Contact header field in the
> dialog-forming INVITE request, that his request in the backwards direction
> is forwarded to the appropriate UA instance. In terms of what Bob's UI
> should display, well, that really depends on the sophistication of attacks
> you are willing to consider plausible. Personally, I don't think Bob's UI
> should suggest any uncertainty about the matter.
> 
> To improve on all this, we will ultimately need to solve the response
> identity problem; however, I maintain that sip-identity as it stands will
> have addressed the simplest and most common variety of impersonation, the
> one that does not require any eavesdropping whatsoever. Once we feel we have
> the right answer to request identity, we can revisit the response identity
> problem and try to eliminate the remaining attacks.
> 
> Jon Peterson
> NeuStar, Inc.
> 
> 
>>-----Original Message-----
>>From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>Sent: Monday, November 15, 2004 3:58 PM
>>To: Peterson, Jon; Cullen Jennings; sip@ietf.org
>>Subject: Identity after reinvite
>>
>>
>>Jon,
>>
>>After the meetings last week a question came to mind re the 
>>identity draft.
>>
>>First, I want to double check on usage in dialogs. The draft doesn't 
>>say, but I presume that every message in a dialog needs to be signed, 
>>not just the one that initiates a dialog. If that is a bad assumption, 
>>it would be good to hear why.
>>
>>Assuming that, what happens when the initial invite was from Alice to 
>>Bob, and later a reinvite is sent from Bob to Alice? Initially Bob knew 
>>that Alice had called, while Alice had no assurance about who she had 
>>reached with her call. The reinvite is from Bob, so if his identity is 
>>blessed, then Alice will know she is talking to Bob, but Bob will no 
>>longer have any assurance that he is still talking to Alice.
>>
>>This could even manifest itself in the UI presented to Alice and Bob. 
>>Initially Bob would have a "callerid" display showing Alice as a 
>>certified caller. But after putting Alice on hold and back off hold the 
>>display might stop showing Alice, or at least might show the identity as 
>>suspect. (There may be strong temptations to avoid this via some sort of 
>>hack.)
>>
>>I don't see any good solution for this other than solving the connected 
>>party id problem as well.
>>
>>	Just checking my understanding,
>>	Paul
>>
> 
> 


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Subject: Re: [Sip] RE: Identity after reinvite
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Jonathan Rosenberg wrote:
> 
> 
> Peterson, Jon wrote:
> 
>> Well, yes, now one begins to appreciate why we originally coupled the
>> request identity problem with the response identity problem. :)
>>
>> In the next revision of the identity draft, which will incorporate the 
>> list
>> comments up to date, I do intend to include some text about dialogs. I 
>> agree
>> that the draft should apply to requests regardless of the dialog context.
> 
> 
> I do believe there are issues related to mid-dialog requests, but I'm 
> not sure I agree that Paul's characterization is the problem.
> 
> The problem is that, due to retargeting, the URI in the To field of the 
> INVITE may not identify the user that was eventually reached. In RFC 
> 3261, requests from the called party to the caller carry the value from 
> the original To field in the From. As a result, the requests will have a 
> From field value which does not actually correspond to the originator of 
> the reuqest.

Very good point!

 > The identity service will then reject these requests.

Hmm. Will the request be rejected, or will it just not get a 
authentication that it is valid? I suppose that is local policy. It may 
be necessary to permit this, without authentication, just to avoid 
breaking these common scenarios.

> The solution, which rfc3261 warns of, is that a UA has to put its actual 
> identity in the From field URI, not what it saw in the To field of the 
> original request.

Yes, this would be a good solution, if we can sort out the compatibility 
issues getting to it.

 > This will break all compatibility with rfc2543, but
> will work with other rfc3261 compliant endpoints, since the URI is no 
> longer used for dialog identification.

If we were going to switch to this, then it could begin with the first 
response to the invite. That would be the first step to really solving 
the connected-partiy-id problem.

>> Moreover, I don't think the problem you raise below is limited merely to
>> re-INVITEs within a dialog. I believe it applies equally to BYE requests
>> sent in the backwards direction in a dialog.

Yes, that is true.

 >> Ideally, the originator
>>  of the
>> dialog should have some assurance that requests sent in the backwards
>> direction come from the same identity that positively responded to the
>> dialog (i.e., the sender of the 200 and any provisional responses).
> 
> Useful, yes, but a different problem. I like the current scope of the 
> identity draft - to provide the identity of the sender of a request. 
> That is not a panacea for all of our security problems, of course.
> 
>> In the absence of response identity, though, it is difficult for Alice to
>> have any such assurance. Equally, as you point out below, Bob can have no
>> assurance that his BYE or re-INVITE has been fielded by Alice if he is
>> unable to get a cryptographically-assured 200 response to his own 
>> requests.
>> Alice can't even be certain that she actually connected to Bob, thanks to
>> retargeting; this is the essence of the connected-ID/response identity
>> problem, and threatens to drag us back into the morass of retargeting
>> restrictions and so on that populated the sip-identity-02 draft.
>>
>> A lack of such a strong binding of backwards-direction BYE requests to 
>> the
>> dialog is a significant limitation; if BYE can be spoofed, then an 
>> attacker
>> can terminate existing dialogs.

This of course is considered a feature by some, who want a proxy to 
morph into a B2BUA just to send a BYE.

> Doesn't sips prevent that? With sips, you won't be able to identify the 
> called party from the 200 OK, but it will guarantee that no one except 
> the party you reach can terminate the call.

Well, if there are multiple hops that isn't proof against every possible 
attack. But it certainly makes it harder. And the kinds of attacks that 
would work here could probably just as well compromise the 
authentication server itself.

> Indeed, it occurs to me that one can "simulate" response identity by 
> having the called party send an UPDATE or nearly any other request in 
> the reverse direction immediately upon receipt of the ACK. If you 
> structure the From field of that request as I have proposed above, the 
> net result provides a form of response identity - it tells you who was 
> reached. This doesnt tell you *why* it was reached (as redirection 
> would), but it seems valuable nonetheless.

As I mention above, if the To were changed in responses, then the caller 
would know right away, without the extra request. But it wouldn't be 
signed. However then the way forward would be clear - have a callee 
suthentiation server sign the To header of responses.

>  However, I'm not sure that the situation is
> 
>> so dire as you suggest below. Without identity, it is trivial for any
>> attacker to impersonate a dialog-forming request. It is harder, 
>> though, for
>> an attacker to impersonate a response or mid-dialog request; this 
>> requires
>> the attacker to gather dialog state through eavesdropping or some similar
>> means. There is a very important distinction in sophistication between an
>> attacker who can merely formulate a plausible dialog-forming request 
>> with an
>> inappropriate From header field versus an attacker who can capture
>> dialog-forming requests and improvise appropriate responses (with correct
>> tags, correct Call-ID/CSeq, etc).
> 
> sips, of course, prevents this fully.
> 
>> Ultimately, request identity can protect against the former but not the
>> latter.
> 
> But, its not supposed to. Its not supposed to provide all of the 
> security functions for sip - just the function which identifies the 
> originator of a request.

Agreed. But is helpful to put it in its place in the bigger context.

	Paul


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From sip-bounces@ietf.org  Tue Nov 16 12:04:18 2004
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Subject: Re: [Sip] mib-08
From: Cullen Jennings <fluffy@cisco.com>
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It's probably better not to ask exactly how much implementation there is
going to be off this :-)


On 11/9/04 11:36 AM, "Steve Langstaff" <steve.langstaff@citel.com> wrote:

> KL: "we need to move the I-D forward so we can get a RFC that can get some
> implementation experience."
> 
> I thought that the I-D phase was the time to get the implementation
> experience. Is it your intention that the RFC will be "Experimental" or
> "Informational" before getting this implementation experience, and then become
> a standard afterwards?
> 
> --
> Steve Langstaff.
> 
> 
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
> Kevin Lingle
> Sent: 08 November 2004 17:26
> To: Cullen Jennings
> Cc: sip@ietf.org
> Subject: Re: [Sip] mib-08
> [snip]
> 
> we need to move the
> I-D forward so we can get a RFC that can get some implementation
> experience.
> 
> kevin



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From sip-bounces@ietf.org  Tue Nov 16 12:08:32 2004
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Subject: Re: [Sip] mib-08
From: Cullen Jennings <fluffy@cisco.com>
To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>,
        Kevin Lingle <klingle@cisco.com>
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Completely agree with what you guys are saying and we should not be
expanding scope - it would just be good to get a few people to actually read
it to make sure it makes sense (in a WGLC is fine).


On 11/8/04 1:16 PM, "Romascanu, Dan (Dan)" <dromasca@avaya.com> wrote:

> I agree with Kevin. At this point in time we should not change the scope of
> the SIP MIB work, but rather focus on completing this phase of the work, in
> the framework already agreed. This does not mean that more eyes to read and
> comment on the MIB would do harm, quite the contrary - but changes should
> focus on fixing problems, not introducing new items.
> 
> After this phase is completed, the WG may be open to discussing possible
> extensions which can be introduced in supplementary SIP MIB modules.
> 
> Regards,
> 
> Dan
> 
> 
> 
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On
>> Behalf Of Kevin Lingle
>> Sent: 08 November, 2004 7:26 PM
>> To: Cullen Jennings
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] mib-08
>> 
>> 
>> hi cullen,
>> 
>> we had a relatively small number of outstanding issues to
>> still resolve and were trying to get a -09 rev out a couple
>> of weeks back.  we didn't make that target because while working
>> the outstanding issues, we also tried to accommodate a request
>> to enhance user agent information.
>> 
>> i don't, think we should try and include that in the initial
>> offering of the user agent mib module.  i think jean-francois
>> is possibly in agreement with that now.  he was going to take
>> a crack at the issue but didn't have time yet.  thus we haven't
>> published rev -09 but hope to soon.  i don't have any time to
>> devote to the new issue of additional user agent information.
>> so i'd like to leave that for the future.
>> 
>> the delta's between -08 and -09 shouldn't be vast.... relative
>> to the deltas between -07 and -08 ;)   i think the "bunch of people"
>> you refer to should be the wglc reviewers.  i don't mind others
>> also reviewing, but i don't want this to become yet another wglc on
>> this draft (we've had two in as many years!).  we need to move the
>> I-D forward so we can get a RFC that can get some implementation
>> experience.
>> 
>> kevin
>> 
>> Cullen Jennings wrote:
>>> I tired to read this whole thing again and it made my head
>> want to explode -
>>> I think getting a bunch of people to read it again would be
>> really good.
>>> There has been some nice work done on it since the previous rev.
>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> 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 sip-bounces@ietf.org  Tue Nov 16 12:33:23 2004
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Subject: [Sip] Boss-Secretary functionalities over SIP
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Hi all,
I am looking for an overview on SIP of the classic PABX Boss-Secretary functionality from the high level call flow perspective.
Does anyone know where I could find this ? 
thanks in advance and kind regards

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Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (1)
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In a message dated 11/15/2004 11:39:06 PM W. Europe Standard Time, 
jmpolk@cisco.com writes:


> >The .03 draft (and I think also the .04 draft) had a paragraph that said
> >    "Namespaces do not describe how they relate to other existing
> >    namespaces, as each namespace is independent of other registrations."
> >
> >It doesn't seem to be in .05
> 
> see (the new) section 7.1 about the warnings of multiple namespaces
> 



I don't read anything in 7.1 as saying what the previous versions said. What 
I think is needed is a very clear statement something like "neither this 
document nor namespaces specify the interrelationship or interworking between 
namespaces, such as mapping from one namespace to another at domain boundaries". 
That was my understanding of the previous agreement.

As a result, the first bullet in Section 7.1, which states "There MUST be one 
order of priorities a SIP element has to process to", while talking about 
multiple namespaces in a message, seems to be wrong (or misunderstood), since it 
seems to require something about the relationship. In fact the example 
following this bullet list is trying to say something about the "acceptable" ways 
that the SIP element may process multiple namespaces. The part about what is "not 
acceptable" is not needed. The examples of incorrect order within each 
individual namespace are not allowed even without multiple namespaces.

Since the document should say that it does not address interworking between 
namespaces, it should not contain this material.

I don't know what the second bullet is trying to say.

The third bullet is obvious, I thought, according to the rules of SIP. If 
not, this statement could be worded better.

Section 7.1 should be deleted completely and replaced with a single statement 
something like I proposed above.

( I guess this will start a long series of e-mails on what I find wrong in 
-05.)

Mike Pierce


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/15/2004 11:39:06 PM W. Europe Standard Time, jmpolk@cisco.com write=
s:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">&gt;The .03 draft (and I th=
ink also the .04 draft) had a paragraph that said
<BR>&gt; &nbsp;&nbsp;&nbsp;"Namespaces do not describe how they relate to ot=
her existing
<BR>&gt; &nbsp;&nbsp;&nbsp;namespaces, as each namespace is independent of o=
ther registrations."
<BR>&gt;
<BR>&gt;It doesn't seem to be in .05
<BR>
<BR>see (the new) section 7.1 about the warnings of multiple namespaces
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>
<BR>I don't read anything in 7.1 as saying what the previous versions said.=20=
What I think is needed is a very clear statement something like "neither thi=
s document nor namespaces specify the interrelationship or interworking betw=
een namespaces, such as mapping from one namespace to another at domain boun=
daries". That was my understanding of the previous agreement.
<BR>
<BR>As a result, the first bullet in Section 7.1, which states "There MUST b=
e one order of priorities a SIP element has to process to", while talking ab=
out multiple namespaces in a message, seems to be wrong (or misunderstood),=20=
since it seems to require something about the relationship. In fact the exam=
ple following this bullet list is trying to say something about the "accepta=
ble" ways that the SIP element may process multiple namespaces. The part abo=
ut what is "not acceptable" is not needed. The examples of incorrect order w=
ithin each individual namespace are not allowed even without multiple namesp=
aces.
<BR>
<BR>Since the document should say that it does not address interworking betw=
een namespaces, it should not contain this material.
<BR>
<BR>I don't know what the second bullet is trying to say.
<BR>
<BR>The third bullet is obvious, I thought, according to the rules of SIP. I=
f not, this statement could be worded better.
<BR>
<BR>Section 7.1 should be deleted completely and replaced with a single stat=
ement something like I proposed above.
<BR>
<BR>( I guess this will start a long series of e-mails on what I find wrong=20=
in -05.)
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

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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
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Date: Tue, 16 Nov 2004 09:29:38 -0800
Subject: Re: [Sip] RE: Identity after reinvite
From: Cullen Jennings <fluffy@cisco.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>,
        Jon Peterson <jon.peterson@neustar.biz>
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On 11/16/04 12:05 AM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:

> The solution, which rfc3261 warns of, is that a UA has to put its actual
> identity in the From field URI, not what it saw in the To field of the
> original request. This will break all compatibility with rfc2543, but
> will work with other rfc3261 compliant endpoints, since the URI is no
> longer used for dialog identification.

At this point, I really see two paths forward for the connected party
problem.

1) We do as suggested above and allow everything in the From and To other
than the tags to be changed in a Re-INVITE or UPDATE.

2) We create a new thing that is like invite with replaces but add a new tag
that is called say "subsumes". So if A and B have a dialog, or early dialog,
and one of them wants to change a From. The UA sends an INVITE for a new
dialog, indicates it replaces the old dialog, and indicates that it subsumes
it. The subsumes flag would mean that any existing media session, qos
reservations, etc did not change and got moved to the new dialog. 



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Tue Nov 16 12:54:03 2004
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Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (1)
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05 is being rewritten as 06.

The point about independance of namespaces has been made.

Let's see what comes out in 06 before spending time on dissecting the fine
points of 05, which will almost certainly be OBE.

Janet


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                      Mpierce1                                                                                   
                      @aol.com                 To:      jmpolk@cisco.com, Janet P Gunn/FED/CSC@CSC,              
                                               oran@cisco.com                                                    
                      11/16/2004 12:17         cc:      An.Nguyen@ncs.gov, sip@ietf.org, sip-bounces@ietf.org    
                      PM                       Subject: Re: [Sip] -resource-priority-05.txt: Comments and        
                                               Recommendations (1)                                               
                                                                                                                 




In a message dated 11/15/2004 11:39:06 PM W. Europe Standard Time,
jmpolk@cisco.com writes:


 >The .03 draft (and I think also the .04 draft) had a paragraph that said
 >    "Namespaces do not describe how they relate to other existing
 >    namespaces, as each namespace is independent of other registrations."

 >
 >It doesn't seem to be in .05

 see (the new) section 7.1 about the warnings of multiple namespaces




I don't read anything in 7.1 as saying what the previous versions said.
What I think is needed is a very clear statement something like "neither
this document nor namespaces specify the interrelationship or interworking
between namespaces, such as mapping from one namespace to another at domain
boundaries". That was my understanding of the previous agreement.

As a result, the first bullet in Section 7.1, which states "There MUST be
one order of priorities a SIP element has to process to", while talking
about multiple namespaces in a message, seems to be wrong (or
misunderstood), since it seems to require something about the relationship.
In fact the example following this bullet list is trying to say something
about the "acceptable" ways that the SIP element may process multiple
namespaces. The part about what is "not acceptable" is not needed. The
examples of incorrect order within each individual namespace are not
allowed even without multiple namespaces.

Since the document should say that it does not address interworking between
namespaces, it should not contain this material.

I don't know what the second bullet is trying to say.

The third bullet is obvious, I thought, according to the rules of SIP. If
not, this statement could be worded better.

Section 7.1 should be deleted completely and replaced with a single
statement something like I proposed above.

( I guess this will start a long series of e-mails on what I find wrong in
-05.)

Mike Pierce




_______________________________________________
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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Cullen Jennings wrote:
> On 11/16/04 12:05 AM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:
> 
>>The solution, which rfc3261 warns of, is that a UA has to put its actual
>>identity in the From field URI, not what it saw in the To field of the
>>original request. This will break all compatibility with rfc2543, but
>>will work with other rfc3261 compliant endpoints, since the URI is no
>>longer used for dialog identification.
> 
> At this point, I really see two paths forward for the connected party
> problem.
> 
> 1) We do as suggested above and allow everything in the From and To other
> than the tags to be changed in a Re-INVITE or UPDATE.
> 
> 2) We create a new thing that is like invite with replaces but add a new tag
> that is called say "subsumes". So if A and B have a dialog, or early dialog,
> and one of them wants to change a From. The UA sends an INVITE for a new
> dialog, indicates it replaces the old dialog, and indicates that it subsumes
> it. The subsumes flag would mean that any existing media session, qos
> reservations, etc did not change and got moved to the new dialog. 

I vote for biting the bullet and changing From/To. It will initially 
cause some pain, but it will solve a number of problems in a clean way.

We could define an option to be announced in Supported that indicates 
ability to do this. Could end up with:

	INVITE ...
	To: sip:bob@biloxi.com
	From: sip:alice@atlanta.com;tag=alice
	Supported: to-from-change
	Identity: aaaaaaaa (signed over From)
	Identity-Info: https://atlanta.com/cert
	...

	180 ...
	To: sip:charlie@chicago.com;tag=charlie
	From: sip:alice@atlanta.com;tag=alice
	Supported: to-from-change
	Identity: cccccccc (signed over To)
	Identity-Info: https://chicago.com/cert
	...

	200 ...
	To: sip:charlie@chicago.com;tag=charlie
	From: sip:alice@atlanta.com;tag=alice
	Supported: to-from-change
	Identity: cccccccc (signed over To)
	Identity-Info: https://chicago.com/cert
	...

	ACK ...
	To: sip:charlie@chicago.com;tag=charlie
	From: sip:alice@atlanta.com;tag=alice
	Supported: to-from-change
	Identity: aaaaaaaa (signed over From)
	Identity-Info: https://atlanta.com/cert
	...

	UPDATE ...
	To: sip:alice@atlanta.com;tag=alice
	From: sip:charlie@chicago.com;tag=charlie
	Supported: to-from-change
	Identity: cccccccc (signed over From)
	Identity-Info: https://chicago.com/cert
	...

	200 ...
	To: sip:alice@atlanta.com;tag=alice
	From: sip:charlie@chicago.com;tag=charlie
	Supported: to-from-change
	Identity: aaaaaaaa (signed over To)
	Identity-Info: https://atlanta.com/cert
	...

(Above shown as they look to receiving UA.)

	Paul


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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] RE: Identity after reinvite
Date: Tue, 16 Nov 2004 13:33:21 -0500
To: sip <sip@ietf.org>
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I have a strong preference for (1). I intensely dislike accumulated 
hacks and entropy in a protocol. Backward compatibility is an important 
property, but 2543 compatibility is fast becoming a a concrete block 
chained to our legs.

We should declare a day that 2543 SIP implementations don't have to 
work right any more. I suggest we make that when 3261 goes to draft 
standard. We can couple that to decreasing entropy in another way - 
don't make more entropic stuff proposed standard until SIP itself gets 
to draft. Then we'll see what existing junk gets pulled from SIP 
because nobody implements it and be moving forward from a stable common 
base.

Donning flame-retardant BVDs...

Dave.

On Nov 16, 2004, at 12:29 PM, Cullen Jennings wrote:

> On 11/16/04 12:05 AM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:
>
>> The solution, which rfc3261 warns of, is that a UA has to put its 
>> actual
>> identity in the From field URI, not what it saw in the To field of the
>> original request. This will break all compatibility with rfc2543, but
>> will work with other rfc3261 compliant endpoints, since the URI is no
>> longer used for dialog identification.
>
> At this point, I really see two paths forward for the connected party
> problem.
>
> 1) We do as suggested above and allow everything in the From and To 
> other
> than the tags to be changed in a Re-INVITE or UPDATE.
>
> 2) We create a new thing that is like invite with replaces but add a 
> new tag
> that is called say "subsumes". So if A and B have a dialog, or early 
> dialog,
> and one of them wants to change a From. The UA sends an INVITE for a 
> new
> dialog, indicates it replaces the old dialog, and indicates that it 
> subsumes
> it. The subsumes flag would mean that any existing media session, qos
> reservations, etc did not change and got moved to the new dialog.
>
>
>
> _______________________________________________
> 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
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


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comments in-line

At 12:17 PM 11/16/2004 -0500, Mpierce1@aol.com wrote:
>In a message dated 11/15/2004 11:39:06 PM W. Europe Standard Time, 
>jmpolk@cisco.com writes:
>
>
>> >The .03 draft (and I think also the .04 draft) had a paragraph that said
>> >    "Namespaces do not describe how they relate to other existing
>> >    namespaces, as each namespace is independent of other registrations."
>> >
>> >It doesn't seem to be in .05
>>
>>see (the new) section 7.1 about the warnings of multiple namespaces
>
>I don't read anything in 7.1 as saying what the previous versions said.

correct

I removed any comment about dependency because there may be a dependency in 
a domain - based on local policy, and this document should not make any 
comment that could contradict local policy.

>What I think is needed is a very clear statement something like "neither 
>this document nor namespaces specify the interrelationship or interworking 
>between namespaces, such as mapping from one namespace to another at 
>domain boundaries".

What you have above is unnecessary. If the document does not state a 
dependency, there is not one defined.

>That was my understanding of the previous agreement.

what previous agreement?


>As a result, the first bullet in Section 7.1, which states "There MUST be 
>one order of priorities a SIP element has to process to", while talking 
>about multiple namespaces in a message, seems to be wrong (or misunderstood),

It is not wrong, and I'm sorry if it is misunderstood.

The section is about *ordering* of namespaces, and gives clear examples of 
ordering that is in line with this specification (to be) and clear examples 
of ordering that is not allowed in this specification (to be).

>  since it seems to require something about the relationship.

no it does not. It either does, or it does not - there is no "seems".

>In fact the example following this bullet list is trying to say something 
>about the "acceptable" ways that the SIP element may process multiple 
>namespaces.

That it does, as it is a violation to reorder priority values just because 
another namespace is introduced. It primarily gives examples of how 
multiple namespaces can be interleaved in acceptable and unacceptable orders.

>The part about what is "not acceptable" is not needed.

I do not agree

>The examples of incorrect order within each individual namespace are not 
>allowed even without multiple namespaces.

Well, how many coders have you spoken or emailed with that understood this 
from the previous version(s)? I'll bet none. I have fielded many such 
communications - and in each given them an explanation just like what is in 
7.1 and it becomes clear fairly quickly what was meant in previous versions 
(that was not explained well enough).


>Since the document should say that it does not address interworking 
>between namespaces, it should not contain this material.

giving examples to clarify a known area of confusion is a good thing - and 
will stay in the document.


>I don't know what the second bullet is trying to say.

that there can be more than one namespace in a message, and have 1 or more 
of these additional namespaces ignored


>The third bullet is obvious, I thought, according to the rules of SIP. If 
>not, this statement could be worded better.

Again, fielded a lot of questions about this, and put it in to clarify to 
coders that the order of RP headers in a message does not determine which 
matter. I reworded it already for the next version.


>Section 7.1 should be deleted completely and replaced with a single 
>statement something like I proposed above.

it will not be deleted, though it might be modified to add or clarify meaning


>( I guess this will start a long series of e-mails on what I find wrong in 
>-05.)

You, have an opinion?

tell me it isn't so   ;-)


>Mike Pierce


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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David R Oran wrote:

> I have a strong preference for (1). I intensely dislike accumulated 
> hacks and entropy in a protocol. Backward compatibility is an important 
> property, but 2543 compatibility is fast becoming a a concrete block 
> chained to our legs.
> 
> We should declare a day that 2543 SIP implementations don't have to work 
> right any more. I suggest we make that when 3261 goes to draft standard. 

Cough... cough... [clutching chest as heart attack ensues]

Draft standard? After the pain we went through to get rfc3261 issued I 
can't imagine doing that again...

Practically speaking, based on the current rules for normative 
references, we are never going to be able to get to draft because 
RFC3261 has too many normative dependencies on protocols that are not at 
draft standard.

I agree we should take approach (1). We could spec it in a backwards 
compatible way, too. If you change the From field in a request you send 
mid-dialog, and you get a 481, you know that something along the way got 
confused about the dialog identifiers. You can then retry, using the 
 From URI from the original INVITE, and include some kind of flag to 
signal to the identity service not to try and insert the Identity header.

Paul wrote:
> We could define an option to be announced in Supported that indicates ability to do this. Could end up with:
> 
>     INVITE ...
>     To: sip:bob@biloxi.com
>     From: sip:alice@atlanta.com;tag=alice
>     Supported: to-from-change
>     Identity: aaaaaaaa (signed over From)
>     Identity-Info: https://atlanta.com/cert
>     ...
> 
>     180 ...
>     To: sip:charlie@chicago.com;tag=charlie
>     From: sip:alice@atlanta.com;tag=alice
>     Supported: to-from-change
>     Identity: cccccccc (signed over To)
>     Identity-Info: https://chicago.com/cert
>     ...
> 
>     200 ...
>     To: sip:charlie@chicago.com;tag=charlie
>     From: sip:alice@atlanta.com;tag=alice
>     Supported: to-from-change
>     Identity: cccccccc (signed over To)
>     Identity-Info: https://chicago.com/cert
>     ...
> 
>     ACK ...
>     To: sip:charlie@chicago.com;tag=charlie
>     From: sip:alice@atlanta.com;tag=alice
>     Supported: to-from-change
>     Identity: aaaaaaaa (signed over From)
>     Identity-Info: https://atlanta.com/cert
>     ...
> 
>     UPDATE ...
>     To: sip:alice@atlanta.com;tag=alice
>     From: sip:charlie@chicago.com;tag=charlie
>     Supported: to-from-change
>     Identity: cccccccc (signed over From)
>     Identity-Info: https://chicago.com/cert
>     ...
> 
>     200 ...
>     To: sip:alice@atlanta.com;tag=alice
>     From: sip:charlie@chicago.com;tag=charlie
>     Supported: to-from-change
>     Identity: aaaaaaaa (signed over To)
>     Identity-Info: https://atlanta.com/cert
>     ...
> 
> (Above shown as they look to receiving UA.) 

This helps only if you assume that the problem lies in the UA; I would 
be worred about other nasties along the way.

In any case, I don't see a need for a Supported header field. The 
feature is supposed to work with any rfc3261 client out there.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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To: Christian Stredicke <Christian.Stredicke@snom.de>
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inline.

Christian Stredicke wrote:

> Hi Spencer!
> 
> You just mentioned another way of keeping TCP connections alive. Thats
> good! 
> 
> So far we have:
> 
> - CRLF (frequenly used today, but not indicated by user agent)
> - STUN (same as CRLF)

It is not the same. STUN solicits a response, which allows the client to 
determine if the NAT binding has changed, and if the server is still alive.

> - PING/PONG or whatever (tbd, maybe superflous)

I am very much against SIP-based keepalives. This turns into a huge 
scalability nightmare. It ends up being the dominant processing cost for 
proxies along with the dominant percentage of the signaling bandwidth, 
and it buys you nothing.


> - OPTIONS (big overhead, but compatible)
> - REGISTER (same as OPTIONS, but even bigger overhead)
> - TCP low-level (how can this be addressed by the application?)
> - No refreshing ("legacy device")
> 
> Lets have a way to indicate these options and let the SBC decide what
> method it likes/supports.

Generally speaking, having multiple choices reduces interoperability. I 
think we should pick a solution, and only describe multiple ones if 
there is a compelling reason why multiple ones need to exist.

Its worth pointing out that there are several functions these various 
keepalvies provide:

1. they keep nat bindings fresh
2. they allow the client to detect if its nat binding as seen by the 
server has changed
3. they allow the client to determine if the server is still alive and 
responding to messages
4. they allow the client to determine if the nat has dropped its binding 
altogether, so that packets are just going into the ether

I *think* Bob is proposing another usage:

5. they allow the server to determine that the nat binding for the 
client has changed, so it can update its registration to reflect the new 
address


Using the keepalive to update the binding on the server introduces 
security risks. If the keepalive is not authenticated, an attacker can 
inject the keepalives with a falsified source IP address, and as a 
result force incoming calls to be directed towards it. This is not 
acceptable. Thus, any request which causes the server to update any kind 
  of bindings MUST be authenticated. I would rather not require the 
keepalives to be authenticated, as this substantially increases the cost 
of processing them. If you use unauthenticated STUN packets, the client 
detects that its nat binding has changed, and then it can re-register 
with a proper REGISTER, which can be authenticated and that can be used 
to update the binding, thus avoiding the attack.

It is possible to use different mechanisms to address each of the 4 
different issues above. An empty UDP packet or a CRLF on the TCP 
connection can address the binding keepalive issue but not any of the 
liveness checks. I strongly prefer a single solution that addresses all 
of these.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org  Tue Nov 16 15:17:14 2004
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From: Jonathan Rosenberg <jdrosen@cisco.com>
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Subject: [Sip] target topology in connect-reuse
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During the sip meeting, we had a lot of discussion about the 
interactions between connect-reuse and the RFC 3263 mechanisms for load 
balancing. The current mechanism in connect-reuse will, I believe, 
create a single connection between a proxy in one domain and one of the 
servers in another domain.

I think the objective of the connect-reuse mechanism is to produce a 
topology that looks like this:

     Please view in a fixed-width font such as
                     Courier.






        +---------+
        |         |
        |         |
        | Proxy   | ---
        |         |    ------          +---------+
        |         |\         -----     |         |
        +---------+ \\            ---  |         |
                      \                | Proxy   |
                       \            -- |         |
                        \\      ----   |         |
        +---------+       \  ---     / +---------+
        |         |       -\\       /
        |         |   ----   \    //
        | Proxy   | --        \  /
        |         |            \X
        |         | --         / \
        +---------+   ----    /   \
                          ---/     \\  +---------+
                           / ----    \ |         |
                          /      --    |         |
                         /         --  | Proxy   |
        +---------+     /       ---    |         |
        |         |   //    ----       |         |
        |         |  /   ---           +---------+
        | Proxy   | / ---
        |         | --
        |         |
        +---------+

That is, between each proxy in domain A and each proxy in domain B, 
there is a single connection used for traffic in each direction. The 
transaction load is distributed across those connections proportiaonlly 
to the weight factors in the SRV records.

This effect can be achieved if a connection is aliased to to an IP 
address, so that the connection reuse takes place *after* the DNS 
operations have occurred.

This means that, when one proxy (A) opens a connection to another (B), A 
has to tell B that, when B wants to send requests to a particular IP 
address, it can use this connection instead. This needs to be done 
securely.

I'd suggest the following. A opens a TCP connection to a server in B 
(based on the DNS lookup of B's domain). There is a TLS exhcange. A 
indicates to B an IP address for which it is an alias. B takes the 
domain from the subjectAltName in the cert presented by A, looks up the 
domain in DNS, and verifies that the offered IP address is listed in the 
DNS record. If so, it is accepted as an alias.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From sip-bounces@ietf.org  Tue Nov 16 15:27:47 2004
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Subject: Re: [Sip] RE: Identity after reinvite
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Jonathan Rosenberg wrote:
> 
> 
> David R Oran wrote:
> 
>> I have a strong preference for (1). I intensely dislike accumulated 
>> hacks and entropy in a protocol. Backward compatibility is an 
>> important property, but 2543 compatibility is fast becoming a a 
>> concrete block chained to our legs.
>>
>> We should declare a day that 2543 SIP implementations don't have to 
>> work right any more. I suggest we make that when 3261 goes to draft 
>> standard. 
> 
> Cough... cough... [clutching chest as heart attack ensues]
> 
> Draft standard? After the pain we went through to get rfc3261 issued I 
> can't imagine doing that again...
> 
> Practically speaking, based on the current rules for normative 
> references, we are never going to be able to get to draft because 
> RFC3261 has too many normative dependencies on protocols that are not at 
> draft standard.
> 
> I agree we should take approach (1). We could spec it in a backwards 
> compatible way, too. If you change the From field in a request you send 
> mid-dialog, and you get a 481, you know that something along the way got 
> confused about the dialog identifiers. You can then retry, using the 
>  From URI from the original INVITE, and include some kind of flag to 
> signal to the identity service not to try and insert the Identity header.

The situation is a little more difficult to detect if you change the To 
in the response to the INVITE, which I was suggesting. Error responses 
don't help in this case. The to-from-change option would however help to 
know you could do this.

But as you point out below, that doesn't help if the proxies on the path 
get upset. To get around that problem we could use Proxy-Require, but 
that is pretty heavy handed. Or we could use some more cooperative 
option negotiation, where all the proxies must concur before the UAS can 
use the feature.

> Paul wrote:
> 
>> We could define an option to be announced in Supported that indicates 
>> ability to do this. Could end up with:
>>
>>     INVITE ...
>>     To: sip:bob@biloxi.com
>>     From: sip:alice@atlanta.com;tag=alice
>>     Supported: to-from-change
>>     Identity: aaaaaaaa (signed over From)
>>     Identity-Info: https://atlanta.com/cert
>>     ...
>>
>>     180 ...
>>     To: sip:charlie@chicago.com;tag=charlie
>>     From: sip:alice@atlanta.com;tag=alice
>>     Supported: to-from-change
>>     Identity: cccccccc (signed over To)
>>     Identity-Info: https://chicago.com/cert
>>     ...
>>
>>     200 ...
>>     To: sip:charlie@chicago.com;tag=charlie
>>     From: sip:alice@atlanta.com;tag=alice
>>     Supported: to-from-change
>>     Identity: cccccccc (signed over To)
>>     Identity-Info: https://chicago.com/cert
>>     ...
>>
>>     ACK ...
>>     To: sip:charlie@chicago.com;tag=charlie
>>     From: sip:alice@atlanta.com;tag=alice
>>     Supported: to-from-change
>>     Identity: aaaaaaaa (signed over From)
>>     Identity-Info: https://atlanta.com/cert
>>     ...
>>
>>     UPDATE ...
>>     To: sip:alice@atlanta.com;tag=alice
>>     From: sip:charlie@chicago.com;tag=charlie
>>     Supported: to-from-change
>>     Identity: cccccccc (signed over From)
>>     Identity-Info: https://chicago.com/cert
>>     ...
>>
>>     200 ...
>>     To: sip:alice@atlanta.com;tag=alice
>>     From: sip:charlie@chicago.com;tag=charlie
>>     Supported: to-from-change
>>     Identity: aaaaaaaa (signed over To)
>>     Identity-Info: https://atlanta.com/cert
>>     ...
>>
>> (Above shown as they look to receiving UA.) 
> 
> 
> This helps

By "this" I assume you mean the to-from-change option?

 > only if you assume that the problem lies in the UA; I would
> be worred about other nasties along the way.

For changing on subsequent requests within the dialog, what nasties are 
you thinking of? I think only a call stateful proxy would notice, and it 
seems there aren't many of those. Of course there are B2BUAs, but they 
at least *ought* to be UAs and behave as such.

Maybe there would be problems with SBCs, and with "transparent" B2BUAs 
that normally act as proxy but morph into UAs on occasion. But I think 
these nasties will have other problems with Identity too. I suppose we 
must however confront these.

> In any case, I don't see a need for a Supported header field. The 
> feature is supposed to work with any rfc3261 client out there.

Other than just hoping for the best, why would we expect existing 3261 
clients to work with this change?

	Paul


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>,
        Jonathan Rosenberg
	<jdrosen@cisco.com>
Subject: RE: [Sip] RE: Identity after reinvite
Date: Tue, 16 Nov 2004 16:46:19 -0500
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> > The problem is that, due to retargeting, the URI in the To field of the 
> > INVITE may not identify the user that was eventually reached. In RFC 
> > 3261, requests from the called party to the caller carry the value from 
> > the original To field in the From. As a result, the requests will have a

> > From field value which does not actually correspond to the originator of

> > the reuqest.
> 
> Very good point!

This is a more significant problem with mid-dialog requests and the request
identity system, agreed.

>  > The identity service will then reject these requests.
> 
> Hmm. Will the request be rejected, or will it just not get a 
> authentication that it is valid? I suppose that is local policy. It may 
> be necessary to permit this, without authentication, just to avoid 
> breaking these common scenarios.

Yes, this is the crux of the matter. The text in Section 6 of
sip-identity-03 says the following:

   If the identity field contains a SIP or SIPS URI,
   the authentication service MUST extract the hostname portion of the
   identity field and compare it to the domain(s) for which it is
   responsible...
   If the authentication service is not responsible for the identity in
   question, it MAY handle the request normally, but it MUST NOT add an
   Identity header; see below for more information on authentication
   service handling of an existing Identity header.

So it isn't that the auth service will reject the request; it MAY just
forward the request normally. What will happen in that instance? Well, if
the auth service role is instantiated by a proxy server, it will forward the
request in accordance with the URI without adding an Identity header; it
won't challenge the request, or what have you.

> > The solution, which rfc3261 warns of, is that a UA has to put its actual

> > identity in the From field URI, not what it saw in the To field of the 
> > original request.
> 
> Yes, this would be a good solution, if we can sort out the compatibility 
> issues getting to it.

I am extremely wary of predicating request identity on solving the
connected-party problem, in part because I believe doing so will dictate a
direction for the response identity problem. If we go down this path here
(changing the From header field value), we will be forced to do the same for
response identity. I think this is the wrong approach, as sip-identity-02
suggested; I think we are sacrificing certain valuable security properties
for the long term if we go down this path.

I have a number of arguments about this, but here are just three:

- Identity will succeed in the marketplace if it can be used
opportunistically. Requiring UAs to change the From header field in this
fashion forces them to become identity-aware, which is an impediment to
adoption. Basically, we force UAs to be either backwards-compliant with
RFC2543 or compliant with Identity. Tough choice, especially given that the
UA might not be aware of its need to be either of those things.

- I also question whether or not the callee UA will even know which identity
it needs to stick in the From header of requests in the backwards direction:
after all, if the UA is registered under multiple AoRs, what exactly about
the dialog-forming request will tell it definitively which identity it is
supposed to use? While in some cases, this could be inferred from the
Request-URI of the dialog-forming requests, it won't be sufficient in all
cases, I think. The previous hop (last proxy server to handle the request)
might also be a hint. Now, it could be that there's a good answer to this
question, but if so, we need to know that answer.

- Retargeting is inherently insecure, and no amount of wiggling with the
sip-identity mechanism will repair this. If Alice sent a request to Bob, and
a new request in the backwards direction comes from Edgar, I think there is
-no- circumstance in which Alice should not view this as a potential
security risk, regardless of the presence or absence of an Identity header,
SIPS, or what have you. Trying to explain this aware flies in the face of
common sense, and will be incomprehensible to actual users who will need to
make security decisions based on the evidence collected from the protocol by
their user agent.

> > This will break all compatibility with rfc2543, but
> > will work with other rfc3261 compliant endpoints, since the URI is no 
> > longer used for dialog identification.
> 
> If we were going to switch to this, then it could begin with the first 
> response to the invite. That would be the first step to really solving 
> the connected-partiy-id problem.

Not just 'could begin' but 'should begin'. If the identity you see in the
response to the INVITE does not equal the identity you see in requests in
the backwards direction, I believe that is extremely problematic. Yes, this
is what I mean by solving the response identity problem.

> > Doesn't sips prevent that? With sips, you won't be able to identify the 
> > called party from the 200 OK, but it will guarantee that no one except 
> > the party you reach can terminate the call.
> 
> Well, if there are multiple hops that isn't proof against every possible 
> attack. But it certainly makes it harder. And the kinds of attacks that 
> would work here could probably just as well compromise the 
> authentication server itself.
> 

Don't forget, for example, that SIPS does not apply after the domain
indicated by the Request-URI of the request, and so on. SIPS is not a
powerful mechanism. Until we all acknowledge that we were mistaken in our
formulation in RFC3261, and that SIPS, when present, signifies unambiguously
that TLS should be used end-to-end, I do not think SIPS can be said to solve
any of our problems. Once we have the connect-reuse mechanism nailed down
firmly, I think it will be possible to shift to that understanding of SIPS,
if we want to.

> > Indeed, it occurs to me that one can "simulate" response identity by 
> > having the called party send an UPDATE or nearly any other request in 
> > the reverse direction immediately upon receipt of the ACK. If you 
> > structure the From field of that request as I have proposed above, the 
> > net result provides a form of response identity - it tells you who was 
> > reached. This doesnt tell you *why* it was reached (as redirection 
> > would), but it seems valuable nonetheless.
> 
> As I mention above, if the To were changed in responses, then the caller 
> would know right away, without the extra request. But it wouldn't be 
> signed. However then the way forward would be clear - have a callee 
> suthentiation server sign the To header of responses.
> 

Yet more justification for the idea that this solution must share its core
with our longer-term response identity mechanism.

> >> There is a very important distinction in sophistication between an
> >> attacker who can merely formulate a plausible dialog-forming request
with an
> >> inappropriate From header field versus an attacker who can capture
> >> dialog-forming requests and improvise appropriate responses (with
correct
> >> tags, correct Call-ID/CSeq, etc).
> > 
> > sips, of course, prevents this fully.

SIPS would be sufficient if it were not crippled by restrictions about the
final administrative domain, and the ambiguity about finality in the context
of retargeting.

Jon Peterson
NeuStar, Inc.

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From sip-bounces@ietf.org  Tue Nov 16 17:47:21 2004
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
        Jonathan Rosenberg
	<jdrosen@cisco.com>
Subject: third alternative (was RE: [Sip] RE: Identity after reinvite)
Date: Tue, 16 Nov 2004 17:07:41 -0500
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I would like to suggest a third alternative:

3) Change nothing about in To/From. Requests in the backwards direction in a
dialog may go through the authentication service of the domain indicated in
the host portion of the From header of the request. If that is not possible,
do not supply request identity for such messages.

While this seem like a non-starter in retargeting cases, requests get
retargeted for a reason. If the dialog-forming request was appropriately
delivered to the domain indicated in the Request-URI (biloxi.example.com),
and that domain chose to forward the request to another domain
(chicago.example.com), it did so because some entity was authorized to add
the chicago URI as a contact for the biloxi URI. In other words, this
approach views retargeting as if the callee's UA were actually registered at
biloxi "through" chicago, and that when the UA receives a dialog-forming
request, it is the To header that determines who should assert identity for
new requests in the backwards direction. In cases where the user contacted
at chicago does not possess credentials to authenticate themselves to
biloxi, I think it would be fair to say that requests in the backwards
direction should not get an Identity header. To go any further, I think,
requires us to solve the response identity problem. This approach is, I
think, forward-compatible with any reasonable solution to the response
identity problem, though.

The primary challenge of this approach is insuring that requests in the
backwards direction will actually go through the authentication service of
biloxi.com, since in many cases they ordinarily might not. If biloxi.com
Record-Routes its auth service in the dialog-forming request, this would
provide the right assurance (and seems plausible from a deployment
perspective); however, if chicago.com also Record-Routes itself, for
example, then the callee will not be able to form a direct TLS connection to
biloxi.com. It is also possible that the callee's UA could force a Route
header to connect to biloxi.com first, but this requires the callee's UA to
be identity-aware, which is undesirable when sip-identity is applied
opportunistically.

But it isn't immediately clear how this is any different from normal cases
of request identity outside the scope of a dialog. There are any number of
possible reasons why the calling UA might not be able to connect directly
via TLS to its auth service (including interposing intermediaries and/or the
inability of the UA to force a Route through their auth service), and the
draft currently just says that we shouldn't provide Identity when we cannot
get the needed relationship between the UA and auth service. I've gotten a
lot of feedback (from Paul, Kumiko and others) that we should try to weaken
the requirement for direct TLS, and I intend to add some text to the next
version about some specific ways in which it is still acceptable to provide
identity even if a direct TLS connection cannot be formed. Certain forms of
weakness there could be beneficial to this approach.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Tuesday, November 16, 2004 9:30 AM
> To: Jonathan Rosenberg; Jon Peterson
> Cc: Paul H Kyzivat; sip@ietf.org
> Subject: Re: [Sip] RE: Identity after reinvite
> 
> 
> On 11/16/04 12:05 AM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:
> 
> > The solution, which rfc3261 warns of, is that a UA has to 
> put its actual
> > identity in the From field URI, not what it saw in the To 
> field of the
> > original request. This will break all compatibility with 
> rfc2543, but
> > will work with other rfc3261 compliant endpoints, since the 
> URI is no
> > longer used for dialog identification.
> 
> At this point, I really see two paths forward for the connected party
> problem.
> 
> 1) We do as suggested above and allow everything in the From 
> and To other
> than the tags to be changed in a Re-INVITE or UPDATE.
> 
> 2) We create a new thing that is like invite with replaces 
> but add a new tag
> that is called say "subsumes". So if A and B have a dialog, 
> or early dialog,
> and one of them wants to change a From. The UA sends an 
> INVITE for a new
> dialog, indicates it replaces the old dialog, and indicates 
> that it subsumes
> it. The subsumes flag would mean that any existing media session, qos
> reservations, etc did not change and got moved to the new dialog. 
> 
> 

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To: "Peterson, Jon" <jon.peterson@neustar.biz>
Subject: Re: [Sip] RE: Identity after reinvite
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Jon,

One of your key points below seems to be to resist tying any of these 
remedies to the identity draft itself. I have no problem with that - 
despite the remaining problems the identity feature does add value and 
need not wait for solution of the other problems. I didn't intend to 
imply otherwise.

More comments below.

	Paul

Peterson, Jon wrote:
>>>The problem is that, due to retargeting, the URI in the To field of the 
>>>INVITE may not identify the user that was eventually reached. In RFC 
>>>3261, requests from the called party to the caller carry the value from 
>>>the original To field in the From. As a result, the requests will have a
> 
>>>From field value which does not actually correspond to the originator of
>>>the reuqest.
>>
>>Very good point!
> 
> This is a more significant problem with mid-dialog requests and the request
> identity system, agreed.
> 
>> > The identity service will then reject these requests.
>>
>>Hmm. Will the request be rejected, or will it just not get a 
>>authentication that it is valid? I suppose that is local policy. It may 
>>be necessary to permit this, without authentication, just to avoid 
>>breaking these common scenarios.
> 
> Yes, this is the crux of the matter. The text in Section 6 of
> sip-identity-03 says the following:
> 
>    If the identity field contains a SIP or SIPS URI,
>    the authentication service MUST extract the hostname portion of the
>    identity field and compare it to the domain(s) for which it is
>    responsible...
>    If the authentication service is not responsible for the identity in
>    question, it MAY handle the request normally, but it MUST NOT add an
>    Identity header; see below for more information on authentication
>    service handling of an existing Identity header.
> 
> So it isn't that the auth service will reject the request; it MAY just
> forward the request normally. What will happen in that instance? Well, if
> the auth service role is instantiated by a proxy server, it will forward the
> request in accordance with the URI without adding an Identity header; it
> won't challenge the request, or what have you.

Good. You confirm what I was assuming.

>>>The solution, which rfc3261 warns of, is that a UA has to put its actual
>>>identity in the From field URI, not what it saw in the To field of the 
>>>original request.
>>
>>Yes, this would be a good solution, if we can sort out the compatibility 
>>issues getting to it.
> 
> I am extremely wary of predicating request identity on solving the
> connected-party problem, in part because I believe doing so will dictate a
> direction for the response identity problem. If we go down this path here
> (changing the From header field value), we will be forced to do the same for
> response identity. I think this is the wrong approach, as sip-identity-02
> suggested; I think we are sacrificing certain valuable security properties
> for the long term if we go down this path.

I agree the basic request identity problem is separable from the others. 
But it does seem that connected-party, response-identity, and reverse 
direction requests in dialog are all entangled, and that changing 
to/from may be a single solution good for all of them.

> I have a number of arguments about this, but here are just three:
> 
> - Identity will succeed in the marketplace if it can be used
> opportunistically. Requiring UAs to change the From header field in this
> fashion forces them to become identity-aware, which is an impediment to
> adoption. Basically, we force UAs to be either backwards-compliant with
> RFC2543 or compliant with Identity. Tough choice, especially given that the
> UA might not be aware of its need to be either of those things.

Agree.

> - I also question whether or not the callee UA will even know which identity
> it needs to stick in the From header of requests in the backwards direction:
> after all, if the UA is registered under multiple AoRs, what exactly about
> the dialog-forming request will tell it definitively which identity it is
> supposed to use? While in some cases, this could be inferred from the
> Request-URI of the dialog-forming requests, it won't be sufficient in all
> cases, I think.  The previous hop (last proxy server to handle the request)
> might also be a hint. Now, it could be that there's a good answer to this
> question, but if so, we need to know that answer.

There are many reasons why a UA with multiple registrations needs to 
know which one was the source of an incoming call. And the UA can easily 
achieve this by using distinct contacts in the registrations. I don't 
think we need to lose sleep over those that choose not to do so.

> - Retargeting is inherently insecure, and no amount of wiggling with the
> sip-identity mechanism will repair this. If Alice sent a request to Bob, and
> a new request in the backwards direction comes from Edgar, I think there is
> -no- circumstance in which Alice should not view this as a potential
> security risk, regardless of the presence or absence of an Identity header,
> SIPS, or what have you. Trying to explain this aware flies in the face of
> common sense, and will be incomprehensible to actual users who will need to
> make security decisions based on the evidence collected from the protocol by
> their user agent.

Of course it is a potential security risk, which is why it is good to 
report that it happened. (A good UA will hopefully note that the 
connected party is not the called party and make a big fuss at its UI.) 
But it is still good to know who in fact you ended up talking to.

This is however not in any way a surprising event. It happens all the 
time for a variety of reasons. Usually it is detected because the voice 
you hear isn't the one you expected to hear. The fact that we can do 
better than that is itself of value. But I don't think we need to 
deprecate the call any further. While it would be good to know more 
about why the retargetting occured, in most cases people will do fine 
without that.

>>>This will break all compatibility with rfc2543, but
>>>will work with other rfc3261 compliant endpoints, since the URI is no 
>>>longer used for dialog identification.
>>
>>If we were going to switch to this, then it could begin with the first 
>>response to the invite. That would be the first step to really solving 
>>the connected-partiy-id problem.
> 
> Not just 'could begin' but 'should begin'. If the identity you see in the
> response to the INVITE does not equal the identity you see in requests in
> the backwards direction, I believe that is extremely problematic. Yes, this
> is what I mean by solving the response identity problem.

The 'could' was a response to Jonathan, who didn't seem to have that in 
mind.

Re having the identity in a subsequent request differ from the one in 
the response to the invite: I'm not sure it is so problematic. The 
identity can change in mid call, and that will then probably first 
manifest itself in the first message sent following the identity change. 
This could happen if one party is a B2BUA or gateway.

Perhaps it would be more problematic if the change was first seen in 
some random message (e.g. INFO or OPTIONS) rather than in a message 
intended to change dialog state (e.g. INVITE or UPDATE). But it is only 
problematic if we enact some rule that says changes in identity must be 
formally announced.

>>>Doesn't sips prevent that? With sips, you won't be able to identify the 
>>>called party from the 200 OK, but it will guarantee that no one except 
>>>the party you reach can terminate the call.
>>
>>Well, if there are multiple hops that isn't proof against every possible 
>>attack. But it certainly makes it harder. And the kinds of attacks that 
>>would work here could probably just as well compromise the 
>>authentication server itself.
> 
> Don't forget, for example, that SIPS does not apply after the domain
> indicated by the Request-URI of the request, and so on. SIPS is not a
> powerful mechanism. Until we all acknowledge that we were mistaken in our
> formulation in RFC3261, and that SIPS, when present, signifies unambiguously
> that TLS should be used end-to-end, I do not think SIPS can be said to solve
> any of our problems. Once we have the connect-reuse mechanism nailed down
> firmly, I think it will be possible to shift to that understanding of SIPS,
> if we want to.
> 
>>>Indeed, it occurs to me that one can "simulate" response identity by 
>>>having the called party send an UPDATE or nearly any other request in 
>>>the reverse direction immediately upon receipt of the ACK. If you 
>>>structure the From field of that request as I have proposed above, the 
>>>net result provides a form of response identity - it tells you who was 
>>>reached. This doesnt tell you *why* it was reached (as redirection 
>>>would), but it seems valuable nonetheless.
>>
>>As I mention above, if the To were changed in responses, then the caller 
>>would know right away, without the extra request. But it wouldn't be 
>>signed. However then the way forward would be clear - have a callee 
>>suthentiation server sign the To header of responses.
> 
> Yet more justification for the idea that this solution must share its core
> with our longer-term response identity mechanism.
> 
>>>>There is a very important distinction in sophistication between an
>>>>attacker who can merely formulate a plausible dialog-forming request
>>>>with an
>>>>inappropriate From header field versus an attacker who can capture
>>>>dialog-forming requests and improvise appropriate responses (with
>>>>correct
>>>>tags, correct Call-ID/CSeq, etc).
>>>
>>>sips, of course, prevents this fully.
> 
> SIPS would be sufficient if it were not crippled by restrictions about the
> final administrative domain, and the ambiguity about finality in the context
> of retargeting.
> 
> Jon Peterson
> NeuStar, Inc.


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
Subject: RE: [Sip] RE: Identity after reinvite
Date: Tue, 16 Nov 2004 19:35:34 -0500
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Paul,

I think we're mostly agreed here, but a couple of small points.

> > - I also question whether or not the callee UA will even know which
identity
> > it needs to stick in the From header of requests in the backwards
direction:
> > after all, if the UA is registered under multiple AoRs, what exactly
about
> > the dialog-forming request will tell it definitively which identity it
is
> > supposed to use? While in some cases, this could be inferred from the
> > Request-URI of the dialog-forming requests, it won't be sufficient in
all
> > cases, I think.  The previous hop (last proxy server to handle the
request)
> > might also be a hint. Now, it could be that there's a good answer to
this
> > question, but if so, we need to know that answer.
> 
> There are many reasons why a UA with multiple registrations needs to 
> know which one was the source of an incoming call. And the UA can easily 
> achieve this by using distinct contacts in the registrations. I don't 
> think we need to lose sleep over those that choose not to do so.

I agree that what you say above is the right overall strategy for SIP
deployments, but I also believe that retargeting would not occur in the case
you describe above. Retargeting occurs when some circumstance like this
prevails: My AoR is jon@biloxi.example.com, and for some reason my current
UA can only register at, say, chicago.example.com, and accordingly I am
forced to put a contact in biloxi.example.com for jon@chicago.example.com.
In that case, I can't differentiate whether a request I receive had
originally been sent through biloxi.example.com or chicago.example.com from
anything other than the To header. In that case, it's not clear what the
"actual" identity of the UA would be, other than just what it sees in the To
header. This is significant, given the original proposal on which I was
commenting:

>>>The solution, which rfc3261 warns of, is that a UA has to put its actual
>>>identity in the From field URI, not what it saw in the To field of the 
>>>original request.

My point is that this is less useful, if not circular, in cases of
retargeting, and the retargeting cases are the ones we are trying to solve.
There's no reason, just because of some transitive registrations, that the
chicago URI is any more "actual" than the biloxi URI; I see an equally good
argument that the biloxi URI is my actual AoR. UAs and their users don't
have unique "actual" identities in SIP.

> > - Retargeting is inherently insecure, and no amount of wiggling with the
> > sip-identity mechanism will repair this. If Alice sent a request to Bob,
and
> > a new request in the backwards direction comes from Edgar, I think there
is
> > -no- circumstance in which Alice should not view this as a potential
> > security risk, regardless of the presence or absence of an Identity
header,
> > SIPS, or what have you. Trying to explain this aware flies in the face
of
> > common sense, and will be incomprehensible to actual users who will need
to
> > make security decisions based on the evidence collected from the
protocol by
> > their user agent.
> 
> Of course it is a potential security risk, which is why it is good to 
> report that it happened. (A good UA will hopefully note that the 
> connected party is not the called party and make a big fuss at its UI.) 
> But it is still good to know who in fact you ended up talking to.

Speaking strictly to request identity for a moment, it is of course possible
that a dialog will pass without any requests being sent in the backwards
direction. So certainly we can't rely on request identity to tell you who
you ended up talking to. If connected-party is a property we want, we should
be getting it from responses - well, more to the point, I think we should be
getting it by eliminating retargeting in favor of redirection, but let's not
go there now. :)

> This is however not in any way a surprising event. It happens all the 
> time for a variety of reasons. Usually it is detected because the voice 
> you hear isn't the one you expected to hear. The fact that we can do 
> better than that is itself of value. But I don't think we need to 
> deprecate the call any further. While it would be good to know more 
> about why the retargetting occured, in most cases people will do fine 
> without that.

I have never found this line of argument very persuasive, but perhaps that's
because I'm less concerned about success cases than failure cases. It's
useful to look at this for a moment as a response identity problem: if the
response you receive does not lead to you hearing a voice (i.e., it's a 604,
or a 301, or something), then you want to know at a protocol level who sent
that response. The reason why we need to do better than the PSTN is not
because the success case is very different from a security perspective
(after all, in the PSTN we just throw calls over the wall and have no
insight into retargeting/security whatsoever, and rely on human voice
identitification as you suggest), it's because the failure case is very
different from a security perspective (because the untrusted Internet is
delivering us these responses rather than a private telephony signaling
network).

Now, all that said, I don't think this is limited to response identity, in
the sense that new requests in the backwards direction (like BYE) have a
serious denial-of-service potential comparable to the use of something like
604 in response identity. So I think there is a clear motivation to do
better than the PSTN. In order to make a reasonable authorization decision
about handling a BYE request in the backwards direction, Alice needs to be
able to anticipate who should be authorized to send such a request. My gut
tells me that Bob should be the person so authorized, and that anyone else
sending such requests should be extremely suspect, if not discarded out of
hand without human oversight.

> >>>This will break all compatibility with rfc2543, but
> >>>will work with other rfc3261 compliant endpoints, since the URI is no 
> >>>longer used for dialog identification.
> >>
> >>If we were going to switch to this, then it could begin with the first 
> >>response to the invite. That would be the first step to really solving 
> >>the connected-partiy-id problem.
> > 
> > Not just 'could begin' but 'should begin'. If the identity you see in
the
> > response to the INVITE does not equal the identity you see in requests
in
> > the backwards direction, I believe that is extremely problematic. Yes,
this
> > is what I mean by solving the response identity problem.
> 
> The 'could' was a response to Jonathan, who didn't seem to have that in 
> mind.
> 
> Re having the identity in a subsequent request differ from the one in 
> the response to the invite: I'm not sure it is so problematic. The 
> identity can change in mid call, and that will then probably first 
> manifest itself in the first message sent following the identity change. 
> This could happen if one party is a B2BUA or gateway. 

So far, I haven't incorporated the idea that the identity can change
mid-call into my requirements space. I am extremely skeptical that this is
an achievable requirement if we cast it as something that the originating
user and the original target don't explicitly sanction. It brings Alice back
to the case of wondering if this guy Edgar can actually tell her to tear
down her call with Bob or not. I really don't see how we can expect users to
understand this.

Now, with some kind of explicit Replaces-style operation that Alice must
authorize, originating from Bob, telling her that Edgar is now in charge, I
believe Alice can make a reasonable decision, and an identity can change
mid-call. But the idea that she is just going to start receiving BYE
messages from some guy named Edgar and just assume that he's in charge;
well, that's tough for me to swallow. Hence I support roughly the strawman
you sketch in the following sentence, and that's why I see this idea of
dissonance between the identities in requests and responses as a problem.

> Perhaps it would be more problematic if the change was first seen in 
> some random message (e.g. INFO or OPTIONS) rather than in a message 
> intended to change dialog state (e.g. INVITE or UPDATE). But it is only 
> problematic if we enact some rule that says changes in identity must be 
> formally announced.
> 

Jon Peterson
NeuStar, Inc.

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From sip-bounces@ietf.org  Tue Nov 16 20:07:20 2004
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
        "'Jonathan Rosenberg'"
	<jdrosen@cisco.com>
Subject: third alternative (was RE: [Sip] RE: Identity after reinvite)
Date: Tue, 16 Nov 2004 19:58:29 -0500
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I would like to suggest a third alternative:

3) Change nothing about in To/From. Requests in the backwards direction in a
dialog may go through the authentication service of the domain indicated in
the host portion of the From header of the request. If that is not possible,
do not supply request identity for such messages.

While this seem like a non-starter in retargeting cases, requests get
retargeted for a reason. If the dialog-forming request was appropriately
delivered to the domain indicated in the Request-URI (biloxi.example.com),
and that domain chose to forward the request to another domain
(chicago.example.com), it did so because some entity was authorized to add
the chicago URI as a contact for the biloxi URI. In other words, this
approach views retargeting as if the callee's UA were actually registered at
biloxi "through" chicago, and that when the UA receives a dialog-forming
request, it is the To header that determines who should assert identity for
new requests in the backwards direction. In cases where the user contacted
at chicago does not possess credentials to authenticate themselves to
biloxi, I think it would be fair to say that requests in the backwards
direction should not get an Identity header. To go any further, I think,
requires us to solve the response identity problem. This approach is, I
think, forward-compatible with any reasonable solution to the response
identity problem, though.

The primary challenge of this approach is insuring that requests in the
backwards direction will actually go through the authentication service of
biloxi.com, since in many cases they ordinarily might not. If biloxi.com
Record-Routes its auth service in the dialog-forming request, this would
provide the right assurance (and seems plausible from a deployment
perspective); however, if chicago.com also Record-Routes itself, for
example, then the callee will not be able to form a direct TLS connection to
biloxi.com. It is also possible that the callee's UA could force a Route
header to connect to biloxi.com first, but this requires the callee's UA to
be identity-aware, which is undesirable when sip-identity is applied
opportunistically.

But it isn't immediately clear how this is any different from normal cases
of request identity outside the scope of a dialog. There are any number of
possible reasons why the calling UA might not be able to connect directly
via TLS to its auth service (including interposing intermediaries and/or the
inability of the UA to force a Route through their auth service), and the
draft currently just says that we shouldn't provide Identity when we cannot
get the needed relationship between the UA and auth service. I've gotten a
lot of feedback (from Paul, Kumiko and others) that we should try to weaken
the requirement for direct TLS, and I intend to add some text to the next
version about some specific ways in which it is still acceptable to provide
identity even if a direct TLS connection cannot be formed. Certain forms of
weakness there could be beneficial to this approach.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Tuesday, November 16, 2004 9:30 AM
> To: Jonathan Rosenberg; Jon Peterson
> Cc: Paul H Kyzivat; sip@ietf.org
> Subject: Re: [Sip] RE: Identity after reinvite
> 
> 
> On 11/16/04 12:05 AM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:
> 
> > The solution, which rfc3261 warns of, is that a UA has to 
> put its actual
> > identity in the From field URI, not what it saw in the To 
> field of the
> > original request. This will break all compatibility with 
> rfc2543, but
> > will work with other rfc3261 compliant endpoints, since the 
> URI is no
> > longer used for dialog identification.
> 
> At this point, I really see two paths forward for the connected party
> problem.
> 
> 1) We do as suggested above and allow everything in the From 
> and To other
> than the tags to be changed in a Re-INVITE or UPDATE.
> 
> 2) We create a new thing that is like invite with replaces 
> but add a new tag
> that is called say "subsumes". So if A and B have a dialog, 
> or early dialog,
> and one of them wants to change a From. The UA sends an 
> INVITE for a new
> dialog, indicates it replaces the old dialog, and indicates 
> that it subsumes
> it. The subsumes flag would mean that any existing media session, qos
> reservations, etc did not change and got moved to the new dialog. 
> 
> 

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Subject: [Sip] SIP open source for Win CE platform
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I would like to implement VoIP(SIP based) on Win CE platform. Any open
source available? which commercial product is more suitable for this
embedded platform?

Thanks a lot,

Bob

_______________________________________________
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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 sip-bounces@ietf.org  Wed Nov 17 02:35:50 2004
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From: "F.S.Salloum" <ssal@intracom.gr>
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Subject: RE: [Sip] SIP open source for Win CE platform
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Hi, 

I have tested, Microsoft Potrait
http://research.microsoft.com/~jiangli/portrait/ and 
SJPhone http://www.sjlabs.com/ 
which work fine with SIP Express Router. www.iptel.org and Asterisk
www.asterisk.org over WLAN 

I'm also trying to make a simple SIP client for WinCE based on open source 
stuff, which will be based on libOSIP and jrtplib modified to work on WinCE
Embedded VC4++, or if this doesn't work with Java and JMF (mobile edition)

I will let you know when I will have the first version..

Regards
Sotiris


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Bo Liu
Sent: Wednesday, November 17, 2004 7:59 AM
To: sip@ietf.org
Subject: [Sip] SIP open source for Win CE platform

I would like to implement VoIP(SIP based) on Win CE platform. Any open
source available? which commercial product is more suitable for this
embedded platform?

Thanks a lot,

Bob

_______________________________________________
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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Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Wed Nov 17 03:21:38 2004
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Based on the contents of previous versions of this draft, and comments I and 
others have made, I believe that the following are essential for the next 
version of this draft.

The only normative information included per namespace should be:

- name of the namespace
- names of the priority levels
- order of the priority values
- default priority level (may be used if the receiving end does not recognize 
a valid priority value)

While the draft could detail various procedures which might be applied, such 
as "modes", it should not associate any specific procedure with any specific 
namespace.

There should not be any other normative, mandatory behavior associated with 
the use of the Resoure-priority header, such as specific authorization 
procedures.

In general, the draft can be made much shorter by deletion of irrelevant 
material. For example, Section 7 describes procedures for "move to the head of the 
line" scenarios for processing of messages in SIP servers or UAs. It has been 
proposed in draft-pierce-tsvwg-pref-treat-examples since April 2002 (and 
never commented on) that there is no need to do such a thing. In fact, it is not a 
good idea to intentionally change the order of messages..

Mike Pierce


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>Based on the=20=
contents of previous versions of this draft, and comments I and others have=20=
made, I believe that the following are essential for the next version of thi=
s draft.
<BR>
<BR>The only normative information included per namespace should be:
<BR>
<BR>- name of the namespace
<BR>- names of the priority levels
<BR>- order of the priority values
<BR>- default priority level (may be used if the receiving end does not reco=
gnize a valid priority value)
<BR>
<BR>While the draft could detail various procedures which might be applied,=20=
such as "modes", it should not associate any specific procedure with any spe=
cific namespace.
<BR>
<BR>There should not be any other normative, mandatory behavior associated w=
ith the use of the Resoure-priority header, such as specific authorization p=
rocedures.
<BR>
<BR>In general, the draft can be made much shorter by deletion of irrelevant=
 material. For example, Section 7 describes procedures for "move to the head=
 of the line" scenarios for processing of messages in SIP servers or UAs. It=
 has been proposed in draft-pierce-tsvwg-pref-treat-examples since April 200=
2 (and never commented on) that there is no need to do such a thing. In fact=
, it is not a good idea to intentionally change the order of messages..
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

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_______________________________________________
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--===============0100863864==--



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From: "Nguyen, An" <An.Nguyen@ncs.gov>
To: "'Mpierce1@aol.com'" <Mpierce1@aol.com>, coreya@nortelnetworks.com,
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Subject: RE: [Sip] -resource-priority-05.txt: Comments and Recommendations (1)
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Mike,
 
I agree with you that it would be overly complex to address all possible
mappings between namespaces in this draft. Mappings between namespaces
issues should be specifically addressed in different documents. Can we just
add a sentence to the draft saying mapping between namespaces is out of
scope and it should be addressed separately? Otherwise, we would never get
this draft published.
 
An

[Nguyen, An]  From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: Friday, November 12, 2004 9:54 AM
To: coreya@nortelnetworks.com; sip@ietf.org
Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations
(1)



Corey, 

Thanks for the great analysis of the situation and the suggestions. I agree
almost completely with what you suggested. I'll follow this up with a few
minor points on your suggestions. But first, some comments on your
questions, which might help understand why this approach was taken. 

In a message dated 11/11/2004 6:12:39 PM Eastern Standard Time,
coreya@nortelnetworks.com writes: 




1. Why support multiple namespace/priority (n-p) values in a single 
message, in one or more R-P headers?  And if supported, should not the 
n-p values in the message equate to the same priority level (for 
example, dsn.flash and q735.1 would be equal)?  Would it not be better 
to support only one n-p value per message, and recommend that a 
proxy/gateway between networks supporting two different namespaces 
handle mapping n-p values? 
---- There may be/are networks which would support multiple levels.  The 
draft is not dictating that there must be multiple levels. 
---- As for having the n-p levels be "equal", James cited a GETS call 
with a specific precedence which becomes a Routine call on the DSN. 
---- Having a proxy/gateway mapped between namespaces is an viable 
alternative. 





While it is a little hard to describe specific cases for the use of multiple
namespaces, I believe it could be used, such as a GETS call originated in a
military network that normally uses only the dsn namespace. The caller,
knowing that the call will likely go outside the dsn, might be able to
specify both dsn=flash and ETS=3. It is not possible to say that they
"equal", since they are two separately defined lists of priorities which
have no direct relationship. In one application, only the dsn namespace
would have an effect while the call remains within the private network (DOD)
for which it was defined. When the call gets to a PSTN gateway to jump into
the "public" network, the dsn namespace/priority is discarded, since the
"public" network does not support it. At the GW, the ETS namespace/priority
is converted to the appropriate PSTN signaling which that netwrok supports.
Note that in this case, the PSTN signaling does not have any further
signaling of this value beyond the initial call setup message, which is why
the SIP signaling would not include this R-P header in anything beyond the
INVITE. 

When an ETS call from the public network enters the dsn, it could be mapped
to Routine or to some higher priority level if the PSTN signaling provided
that information. That will be determine by the administrator of the network
and what the GW is told to do. It certainly can't be defined in the R-P
header draft. 

Another application is a gateway between two networks which each use the R-P
header but with different definitions. Lets say between the US DOD network
with 5 values and another country's military network when that other country
uses only 3 levels. The GW must map between the two sets according to a
mutually agreed mapping. In each network, only one R-P header is used. 

I believe it was agreed long ago that the R-P header draft should not try to
define the interoperation/mapping between different namespaces. I think this
approach should remain, since any attempt to define relationships would be
overly complex and may conflict with what network operators want to do. 

The first example above, with two namespaces being carried, is the reason
why the required mode of operation in the dsn is that unknown namespaces are
simply ignored and passed on. Known namespaces with unknown priorities need
to be treated as errors, but not necessarily by failing a call. 

Mike Pierce 
Artel 





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<DIV><FONT face=Arial color=#0000ff size=2><SPAN 
class=906091116-16072001>Mike,</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN 
class=906091116-16072001></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=906091116-16072001>I 
agree with you that it would be overly complex to address all possible mappings 
between namespaces in this draft.&nbsp;Mappings between&nbsp;namespaces issues 
should be&nbsp;specifically addressed in&nbsp;different documents. Can we just 
add a sentence to the draft saying mapping between namespaces is out of scope 
and it should be addressed separately? Otherwise, we would never get this draft 
published.</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff><SPAN 
class=906091116-16072001></SPAN></FONT><FONT face=Tahoma><FONT face=Arial 
color=#0000ff size=2></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=Tahoma><SPAN class=906091116-16072001><FONT face=Arial 
color=#0000ff size=2>An</FONT></SPAN></DIV>
<DIV><BR><FONT size=2><SPAN class=906091116-16072001><FONT face=Arial 
color=#0000ff>[Nguyen, An]&nbsp;&nbsp;</FONT></SPAN><STRONG>From:</STRONG> 
Mpierce1@aol.com [mailto:Mpierce1@aol.com]<BR><B>Sent:</B> Friday, November 12, 
2004 9:54 AM<BR><B>To:</B> coreya@nortelnetworks.com; 
sip@ietf.org<BR><B>Subject:</B> Re: [Sip] -resource-priority-05.txt: Comments 
and Recommendations (1)<BR><BR></DIV></FONT></FONT>
<BLOCKQUOTE><FONT face=arial,helvetica><FONT size=2 PTSIZE="10">Corey, 
  <BR><BR>Thanks for the great analysis of the situation and the suggestions. I 
  agree almost completely with what you suggested. I'll follow this up with a 
  few minor points on your suggestions. But first, some comments on your 
  questions, which might help understand why this approach was taken. <BR><BR>In 
  a message dated 11/11/2004 6:12:39 PM Eastern Standard Time, 
  coreya@nortelnetworks.com writes: <BR><BR><BR>
  <BLOCKQUOTE 
  style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px" 
  TYPE="CITE">1. Why support multiple namespace/priority (n-p) values in a 
    single <BR>message, in one or more R-P headers? &nbsp;And if supported, 
    should not the <BR>n-p values in the message equate to the same priority 
    level (for <BR>example, dsn.flash and q735.1 would be equal)? &nbsp;Would it 
    not be better <BR>to support only one n-p value per message, and recommend 
    that a <BR>proxy/gateway between networks supporting two different 
    namespaces <BR>handle mapping n-p values? <BR>---- There may be/are networks 
    which would support multiple levels. &nbsp;The <BR>draft is not dictating 
    that there must be multiple levels. <BR>---- As for having the n-p levels be 
    "equal", James cited a GETS call <BR>with a specific precedence which 
    becomes a Routine call on the DSN. <BR>---- Having a proxy/gateway mapped 
    between namespaces is an viable <BR>alternative. <BR></FONT><FONT lang=0 
    style="BACKGROUND-COLOR: #ffffff" face=Arial size=3 PTSIZE="12" 
    FAMILY="SANSSERIF" BACK="#ffffff"></BLOCKQUOTE><BR></FONT><FONT lang=0 
  style="BACKGROUND-COLOR: #ffffff" face=Arial size=2 PTSIZE="10" 
  FAMILY="SANSSERIF" BACK="#ffffff"><BR><BR>While it is a little hard to 
  describe specific cases for the use of multiple namespaces, I believe it could 
  be used, such as a GETS call originated in a military network that normally 
  uses only the dsn namespace. The caller, knowing that the call will likely go 
  outside the dsn, might be able to specify both dsn=flash and ETS=3. It is not 
  possible to say that they "equal", since they are two separately defined lists 
  of priorities which have no direct relationship. In one application, only the 
  dsn namespace would have an effect while the call remains within the private 
  network (DOD) for which it was defined. When the call gets to a PSTN gateway 
  to jump into the "public" network, the dsn namespace/priority is discarded, 
  since the "public" network does not support it. At the GW, the ETS 
  namespace/priority is converted to the appropriate PSTN signaling which that 
  netwrok supports. Note that in this case, the PSTN signaling does not have any 
  further signaling of this value beyond the initial call setup message, which 
  is why the SIP signaling would not include this R-P header in anything beyond 
  the INVITE. <BR><BR>When an ETS call from the public network enters the dsn, 
  it could be mapped to Routine or to some higher priority level if the PSTN 
  signaling provided that information. That will be determine by the 
  administrator of the network and what the GW is told to do. It certainly can't 
  be defined in the R-P header draft. <BR><BR>Another application is a gateway 
  between two networks which each use the R-P header but with different 
  definitions. Lets say between the US DOD network with 5 values and another 
  country's military network when that other country uses only 3 levels. The GW 
  must map between the two sets according to a mutually agreed mapping. In each 
  network, only one R-P header is used. <BR><BR>I believe it was agreed long ago 
  that the R-P header draft should not try to define the interoperation/mapping 
  between different namespaces. I think this approach should remain, since any 
  attempt to define relationships would be overly complex and may conflict with 
  what network operators want to do. <BR><BR>The first example above, with two 
  namespaces being carried, is the reason why the required mode of operation in 
  the dsn is that unknown namespaces are simply ignored and passed on. Known 
  namespaces with unknown priorities need to be treated as errors, but not 
  necessarily by failing a call. <BR><BR>Mike Pierce <BR>Artel <BR></FONT><FONT 
  size=3><BR><BR></BLOCKQUOTE></FONT></FONT></BODY></HTML>

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Sip mailing list  https://www1.ietf.org/mailman/listinfo/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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Mike,
 
I agree with you that it would be overly complex to address all possible
mappings between namespaces in this draft. Mappings between namespaces
issues should be specifically addressed in different documents. Can we just
add a sentence to the draft saying mapping between namespaces is out of
scope and it should be addressed separately? Otherwise, we would never get
this draft published.
 
An

[Nguyen, An]  From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: Friday, November 12, 2004 9:54 AM
To: coreya@nortelnetworks.com; sip@ietf.org
Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations
(1)



Corey, 

Thanks for the great analysis of the situation and the suggestions. I agree
almost completely with what you suggested. I'll follow this up with a few
minor points on your suggestions. But first, some comments on your
questions, which might help understand why this approach was taken. 

In a message dated 11/11/2004 6:12:39 PM Eastern Standard Time,
coreya@nortelnetworks.com writes: 




1. Why support multiple namespace/priority (n-p) values in a single 
message, in one or more R-P headers?  And if supported, should not the 
n-p values in the message equate to the same priority level (for 
example, dsn.flash and q735.1 would be equal)?  Would it not be better 
to support only one n-p value per message, and recommend that a 
proxy/gateway between networks supporting two different namespaces 
handle mapping n-p values? 
---- There may be/are networks which would support multiple levels.  The 
draft is not dictating that there must be multiple levels. 
---- As for having the n-p levels be "equal", James cited a GETS call 
with a specific precedence which becomes a Routine call on the DSN. 
---- Having a proxy/gateway mapped between namespaces is an viable 
alternative. 





While it is a little hard to describe specific cases for the use of multiple
namespaces, I believe it could be used, such as a GETS call originated in a
military network that normally uses only the dsn namespace. The caller,
knowing that the call will likely go outside the dsn, might be able to
specify both dsn=flash and ETS=3. It is not possible to say that they
"equal", since they are two separately defined lists of priorities which
have no direct relationship. In one application, only the dsn namespace
would have an effect while the call remains within the private network (DOD)
for which it was defined. When the call gets to a PSTN gateway to jump into
the "public" network, the dsn namespace/priority is discarded, since the
"public" network does not support it. At the GW, the ETS namespace/priority
is converted to the appropriate PSTN signaling which that netwrok supports.
Note that in this case, the PSTN signaling does not have any further
signaling of this value beyond the initial call setup message, which is why
the SIP signaling would not include this R-P header in anything beyond the
INVITE. 

When an ETS call from the public network enters the dsn, it could be mapped
to Routine or to some higher priority level if the PSTN signaling provided
that information. That will be determine by the administrator of the network
and what the GW is told to do. It certainly can't be defined in the R-P
header draft. 

Another application is a gateway between two networks which each use the R-P
header but with different definitions. Lets say between the US DOD network
with 5 values and another country's military network when that other country
uses only 3 levels. The GW must map between the two sets according to a
mutually agreed mapping. In each network, only one R-P header is used. 

I believe it was agreed long ago that the R-P header draft should not try to
define the interoperation/mapping between different namespaces. I think this
approach should remain, since any attempt to define relationships would be
overly complex and may conflict with what network operators want to do. 

The first example above, with two namespaces being carried, is the reason
why the required mode of operation in the dsn is that unknown namespaces are
simply ignored and passed on. Known namespaces with unknown priorities need
to be treated as errors, but not necessarily by failing a call. 

Mike Pierce 
Artel 





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<DIV><FONT face=Arial color=#0000ff size=2><SPAN 
class=906091116-16072001>Mike,</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN 
class=906091116-16072001></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=906091116-16072001>I 
agree with you that it would be overly complex to address all possible mappings 
between namespaces in this draft.&nbsp;Mappings between&nbsp;namespaces issues 
should be&nbsp;specifically addressed in&nbsp;different documents. Can we just 
add a sentence to the draft saying mapping between namespaces is out of scope 
and it should be addressed separately? Otherwise, we would never get this draft 
published.</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff><SPAN 
class=906091116-16072001></SPAN></FONT><FONT face=Tahoma><FONT face=Arial 
color=#0000ff size=2></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=Tahoma><SPAN class=906091116-16072001><FONT face=Arial 
color=#0000ff size=2>An</FONT></SPAN></DIV>
<DIV><BR><FONT size=2><SPAN class=906091116-16072001><FONT face=Arial 
color=#0000ff>[Nguyen, An]&nbsp;&nbsp;</FONT></SPAN><STRONG>From:</STRONG> 
Mpierce1@aol.com [mailto:Mpierce1@aol.com]<BR><B>Sent:</B> Friday, November 12, 
2004 9:54 AM<BR><B>To:</B> coreya@nortelnetworks.com; 
sip@ietf.org<BR><B>Subject:</B> Re: [Sip] -resource-priority-05.txt: Comments 
and Recommendations (1)<BR><BR></DIV></FONT></FONT>
<BLOCKQUOTE><FONT face=arial,helvetica><FONT size=2 PTSIZE="10">Corey, 
  <BR><BR>Thanks for the great analysis of the situation and the suggestions. I 
  agree almost completely with what you suggested. I'll follow this up with a 
  few minor points on your suggestions. But first, some comments on your 
  questions, which might help understand why this approach was taken. <BR><BR>In 
  a message dated 11/11/2004 6:12:39 PM Eastern Standard Time, 
  coreya@nortelnetworks.com writes: <BR><BR><BR>
  <BLOCKQUOTE 
  style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px" 
  TYPE="CITE">1. Why support multiple namespace/priority (n-p) values in a 
    single <BR>message, in one or more R-P headers? &nbsp;And if supported, 
    should not the <BR>n-p values in the message equate to the same priority 
    level (for <BR>example, dsn.flash and q735.1 would be equal)? &nbsp;Would it 
    not be better <BR>to support only one n-p value per message, and recommend 
    that a <BR>proxy/gateway between networks supporting two different 
    namespaces <BR>handle mapping n-p values? <BR>---- There may be/are networks 
    which would support multiple levels. &nbsp;The <BR>draft is not dictating 
    that there must be multiple levels. <BR>---- As for having the n-p levels be 
    "equal", James cited a GETS call <BR>with a specific precedence which 
    becomes a Routine call on the DSN. <BR>---- Having a proxy/gateway mapped 
    between namespaces is an viable <BR>alternative. <BR></FONT><FONT lang=0 
    style="BACKGROUND-COLOR: #ffffff" face=Arial size=3 PTSIZE="12" 
    FAMILY="SANSSERIF" BACK="#ffffff"></BLOCKQUOTE><BR></FONT><FONT lang=0 
  style="BACKGROUND-COLOR: #ffffff" face=Arial size=2 PTSIZE="10" 
  FAMILY="SANSSERIF" BACK="#ffffff"><BR><BR>While it is a little hard to 
  describe specific cases for the use of multiple namespaces, I believe it could 
  be used, such as a GETS call originated in a military network that normally 
  uses only the dsn namespace. The caller, knowing that the call will likely go 
  outside the dsn, might be able to specify both dsn=flash and ETS=3. It is not 
  possible to say that they "equal", since they are two separately defined lists 
  of priorities which have no direct relationship. In one application, only the 
  dsn namespace would have an effect while the call remains within the private 
  network (DOD) for which it was defined. When the call gets to a PSTN gateway 
  to jump into the "public" network, the dsn namespace/priority is discarded, 
  since the "public" network does not support it. At the GW, the ETS 
  namespace/priority is converted to the appropriate PSTN signaling which that 
  netwrok supports. Note that in this case, the PSTN signaling does not have any 
  further signaling of this value beyond the initial call setup message, which 
  is why the SIP signaling would not include this R-P header in anything beyond 
  the INVITE. <BR><BR>When an ETS call from the public network enters the dsn, 
  it could be mapped to Routine or to some higher priority level if the PSTN 
  signaling provided that information. That will be determine by the 
  administrator of the network and what the GW is told to do. It certainly can't 
  be defined in the R-P header draft. <BR><BR>Another application is a gateway 
  between two networks which each use the R-P header but with different 
  definitions. Lets say between the US DOD network with 5 values and another 
  country's military network when that other country uses only 3 levels. The GW 
  must map between the two sets according to a mutually agreed mapping. In each 
  network, only one R-P header is used. <BR><BR>I believe it was agreed long ago 
  that the R-P header draft should not try to define the interoperation/mapping 
  between different namespaces. I think this approach should remain, since any 
  attempt to define relationships would be overly complex and may conflict with 
  what network operators want to do. <BR><BR>The first example above, with two 
  namespaces being carried, is the reason why the required mode of operation in 
  the dsn is that unknown namespaces are simply ignored and passed on. Known 
  namespaces with unknown priorities need to be treated as errors, but not 
  necessarily by failing a call. <BR><BR>Mike Pierce <BR>Artel <BR></FONT><FONT 
  size=3><BR><BR></BLOCKQUOTE></FONT></FONT></BODY></HTML>

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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
--===============1602173489==--



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Hello ,
               I am an active member of this work group. It so happened that
i deleted some old mails.
Earlier i Use to see Old mails on the Archives link. But now if i open ,it
give today's and yesterday's mail.
I am force to go to link
http://www1.ietf.org/mail-archive/web/sip/current/index.html and there is no
other link there :(
Please help !!


Vikram


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Date: Wed, 17 Nov 2004 08:03:57 EST
Subject: Re: [Sip] -resource-priority-05.txt: Comments and Recommendations (1)
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In a message dated 11/16/2004 7:42:29 PM W. Europe Standard Time, 
jmpolk@cisco.com writes:


> 
> 
> I removed any comment about dependency because there may be a dependency in 
> a domain - based on local policy, and this document should not make any 
> comment that could contradict local policy.
> 
> >[MAP] What I think is needed is a very clear statement something like 
> "neither 
> >this document nor namespaces specify the interrelationship or interworking 
> >between namespaces, such as mapping from one namespace to another at 
> >domain boundaries".
> 
> What you have above is unnecessary. If the document does not state a 
> dependency, there is not one defined.
> 
> 
> >[MAP] As a result, the first bullet in Section 7.1, which states "There 
> MUST be 
> >one order of priorities a SIP element has to process to", while talking 
> >about multiple namespaces in a message, seems to be wrong (or 
> misunderstood),
> 
> It is not wrong, and I'm sorry if it is misunderstood.
> 
> The section is about *ordering* of namespaces, and gives clear examples of 
> ordering that is in line with this specification (to be) and clear examples 
> of ordering that is not allowed in this specification (to be).
> 
> >  [MAP] since it seems to require something about the relationship.
> 
> no it does not. It either does, or it does not - there is no "seems".
> 
> >[MAP] In fact the example following this bullet list is trying to say 
> something 
> >about the "acceptable" ways that the SIP element may process multiple 
> >namespaces.
> 
> That it does, as it is a violation to reorder priority values just because 
> another namespace is introduced. It primarily gives examples of how 
> multiple namespaces can be interleaved in acceptable and unacceptable orders.
> 
> >[MAP] The part about what is "not acceptable" is not needed.
> 
> I do not agree
> 
> >[MAP] The examples of incorrect order within each individual namespace are 
> not 
> >allowed even without multiple namespaces.
> 
> Well, how many coders have you spoken or emailed with that understood this 
> from the previous version(s)? I'll bet none. I have fielded many such 
> communications - and in each given them an explanation just like what is in 
> 7.1 and it becomes clear fairly quickly what was meant in previous versions 
> (that was not explained well enough).
> 
> 
> >[MAP] Since the document should say that it does not address interworking 
> >between namespaces, it should not contain this material.
> 
> giving examples to clarify a known area of confusion is a good thing - and 
> will stay in the document.
> 
> 



[MAP] The above results in a serious contradiction. You agree that the 
document should not describe dependencies, then include material that certainly 
describes dependencies (what combinations are and are not allowed). This will 
confuse more readers into thinking that there are some dependencies.

> 
> 
> it will not be deleted, though it might be modified to add or clarify meaning
> 
> 


[MAP] It's too bad that you're taking the position that you have the final 
say on what is and is not included in the document regardless of the comments 
and results of any e-mail discussion. As far as I'm concerned, you must get the 
WG consensus for anything that you added in -05 in order to retain it in -06. 
Contrary to what Jon Peterson stated at the meeting, RFC2418 defines the 
responsibilities of the editor as follows:

   Most IETF working groups focus their efforts on a document, or set of
   documents, that capture the results of the group's work.  A working
   group generally designates a person or persons to serve as the Editor
   for a particular document.  The Document Editor is responsible for
   ensuring that the contents of the document accurately reflect the
   decisions that have been made by the working group.


Mike


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/16/2004 7:42:29 PM W. Europe Standard Time, jmpolk@cisco.com writes=
:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
<BR>
<BR>I removed any comment about dependency because there may be a dependency=
 in=20
<BR>a domain - based on local policy, and this document should not make any=20
<BR>comment that could contradict local policy.
<BR>
<BR>&gt;[MAP] What I think is needed is a very clear statement something lik=
e "neither=20
<BR>&gt;this document nor namespaces specify the interrelationship or interw=
orking=20
<BR>&gt;between namespaces, such as mapping from one namespace to another at=
=20
<BR>&gt;domain boundaries".
<BR>
<BR>What you have above is unnecessary. If the document does not state a=20
<BR>dependency, there is not one defined.
<BR>
<BR>
<BR>&gt;[MAP] As a result, the first bullet in Section 7.1, which states "Th=
ere MUST be=20
<BR>&gt;one order of priorities a SIP element has to process to", while talk=
ing=20
<BR>&gt;about multiple namespaces in a message, seems to be wrong (or misund=
erstood),
<BR>
<BR>It is not wrong, and I'm sorry if it is misunderstood.
<BR>
<BR>The section is about *ordering* of namespaces, and gives clear examples=20=
of=20
<BR>ordering that is in line with this specification (to be) and clear examp=
les=20
<BR>of ordering that is not allowed in this specification (to be).
<BR>
<BR>&gt; &nbsp;[MAP] since it seems to require something about the relations=
hip.
<BR>
<BR>no it does not. It either does, or it does not - there is no "seems".
<BR>
<BR>&gt;[MAP] In fact the example following this bullet list is trying to sa=
y something=20
<BR>&gt;about the "acceptable" ways that the SIP element may process multipl=
e=20
<BR>&gt;namespaces.
<BR>
<BR>That it does, as it is a violation to reorder priority values just becau=
se=20
<BR>another namespace is introduced. It primarily gives examples of how=20
<BR>multiple namespaces can be interleaved in acceptable and unacceptable or=
ders.
<BR>
<BR>&gt;[MAP] The part about what is "not acceptable" is not needed.
<BR>
<BR>I do not agree
<BR>
<BR>&gt;[MAP] The examples of incorrect order within each individual namespa=
ce are not=20
<BR>&gt;allowed even without multiple namespaces.
<BR>
<BR>Well, how many coders have you spoken or emailed with that understood th=
is=20
<BR>from the previous version(s)? I'll bet none. I have fielded many such=20
<BR>communications - and in each given them an explanation just like what is=
 in=20
<BR>7.1 and it becomes clear fairly quickly what was meant in previous versi=
ons=20
<BR>(that was not explained well enough).
<BR>
<BR>
<BR>&gt;[MAP] Since the document should say that it does not address interwo=
rking=20
<BR>&gt;between namespaces, it should not contain this material.
<BR>
<BR>giving examples to clarify a known area of confusion is a good thing - a=
nd=20
<BR>will stay in the document.
<BR>
<BR></BLOCKQUOTE>
<BR>
<BR>
<BR>
<BR>[MAP] The above results in a serious contradiction. You agree that the d=
ocument should not describe dependencies, then include material that certain=
ly describes dependencies (what combinations are and are not allowed). This=20=
will confuse more readers into thinking that there are some dependencies.
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
<BR>
<BR>it will not be deleted, though it might be modified to add or clarify me=
aning
<BR>
<BR></BLOCKQUOTE>
<BR>
<BR>
<BR>[MAP] It's too bad that you're taking the position that you have the fin=
al say on what is and is not included in the document regardless of the comm=
ents and results of any e-mail discussion. As far as I'm concerned, you must=
 get the WG consensus for anything that you added in -05 in order to retain=20=
it in -06. Contrary to what Jon Peterson stated at the meeting, RFC2418 defi=
nes the responsibilities of the editor as follows:
<BR>
<BR> &nbsp;&nbsp;Most IETF working groups focus their efforts on a document,=
 or set of
<BR> &nbsp;&nbsp;documents, that capture the results of the group's work. &n=
bsp;A working
<BR> &nbsp;&nbsp;group generally designates a person or persons to serve as=20=
the Editor
<BR> &nbsp;&nbsp;for a particular document. &nbsp;The Document Editor is res=
ponsible for
<BR> &nbsp;&nbsp;ensuring that the contents of the document accurately refle=
ct the
<BR> &nbsp;&nbsp;decisions that have been made by the working group.
<BR>
<BR>
<BR>Mike
<BR></FONT></HTML>

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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
--===============0394264048==--



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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'PROUVOST Sebastien RD-CORE-ISS'" <sebastien.prouvost@francetelecom.com>
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Subject: [Sip] RE: Comments regarding the reason in
	draft-ietf-sip-history-info- 04
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Hi Sebastien,
=20
I think the need for these specific reasons depends upon how you =
implement
your voicemail service.  I think what you perhaps might need in these
situations, such as the "call forwarding immediate", is extensions to =
the
Reason header as discussed in =
draft-elwell-sipping-redirection-reason-01.
=20
I don't think this is something that should at all be addressed in the
history-info draft, which is just one the proposed building blocks for
providing voicemail services.=20
=20
Mary=20

-----Original Message-----
From: PROUVOST Sebastien RD-CORE-ISS
[mailto:sebastien.prouvost@francetelecom.com]=20
Sent: Wednesday, November 17, 2004 2:36 AM
To: Barnes, Mary [NGC:B601:EXCH]
Cc: sip@ietf.org
Subject: Comments regarding the reason in =
draft-ietf-sip-history-info-04




Hi Mary,=20

One comment on the history draft regarding the reason in history-info
header.

Section 4.3.3.1.2 mentions that for retarget as a result of timeouts of
internal events, a Reason MAY be associated with the =
hi-targeted-to-uri.

I think that 2 specific values of the reason header shall be specified =
by
the draft as they are necessary to voicemail services (for example in =
order
to render to the caller an annonce associated with the reason such as =
"the
user you are calling is not responding" or "the user you are trying to
contact is not available (registered) at the moment") :=20

        - a value for timeout (reason =3D 'no response' for example)=20
        - a value for direct retargeting to a default contact URI or to
another AoR (reason =3D 'call forwarding immediate' for example).

I am therefore in favour of adding a text looking like this :=20
"=20
As a result of timeouts of internal events, a Reason MAY be associated =
with
the hi-targeted-to-uri. Although other values may be possible this =
draft
defines two values for the Reason in such cases :=20

- Reason =3D'no response' when the retargeting occurs after timeout=20
- Reason =3D 'call forwarding immediate' when the proxy directly =
retargets to
a default contact URI or to another AoR=20
"=20

S=E9bastien=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1476" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D176275114-17112004>Hi=20
Sebastien,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D176275114-17112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D176275114-17112004>I=20
think the need for these specific reasons depends upon how you =
implement your=20
voicemail service.&nbsp; I think what you perhaps might need in these=20
situations, such as the "call forwarding immediate", is extensions to =
the Reason=20
header as discussed in=20
draft-elwell-sipping-redirection-reason-01.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D176275114-17112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D176275114-17112004>I=20
don't think this is something that should at all be addressed in the=20
history-info draft, which is just one the proposed building blocks for =
providing=20
voicemail services. </SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff><SPAN lang=3Den-us><FONT face=3DArial=20
size=3D2>Mary</FONT></SPAN> </FONT></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
PROUVOST=20
  Sebastien RD-CORE-ISS [mailto:sebastien.prouvost@francetelecom.com]=20
  <BR><B>Sent:</B> Wednesday, November 17, 2004 2:36 AM<BR><B>To:</B> =
Barnes,=20
  Mary [NGC:B601:EXCH]<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> =
Comments=20
  regarding the reason in =
draft-ietf-sip-history-info-04<BR><BR></FONT></DIV><!-- Converted from =
text/rtf format --><BR>
  <P><FONT face=3DArial size=3D2>Hi Mary,</FONT> </P>
  <P><FONT face=3DArial size=3D2>One comment on the history draft =
regarding the=20
  reason in history-info header.<BR></FONT><BR><FONT face=3DArial =
size=3D2>Section=20
  4.3.3.1.2 mentions that for retarget as a result of timeouts of =
internal=20
  events, a Reason MAY be associated with the =
hi-targeted-to-uri.</FONT></P>
  <P><FONT face=3DArial size=3D2>I think that 2 specific values of the =
reason header=20
  shall be specified by the draft as they are necessary to voicemail =
services=20
  (for example in order to render to the caller an annonce associated =
with the=20
  reason such as "the user you are calling is not responding" or "the =
user you=20
  are trying to contact is not available (registered) at the moment") : =

  </FONT></P>
  <P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT face=3DArial =
size=3D2>- a=20
  value for timeout (reason =3D 'no response' for example)</FONT>=20
  <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT face=3DArial =
size=3D2>- a=20
  value for direct retargeting to a default contact URI or to another =
AoR=20
  (reason =3D 'call forwarding immediate' for example).</FONT></P>
  <P><FONT face=3DArial size=3D2>I am therefore in favour of adding a =
text looking=20
  like this :</FONT> <BR><FONT face=3DArial size=3D2>"</FONT> <BR><FONT =
face=3DArial=20
  size=3D2>As a result of timeouts of internal events, a Reason MAY be =
associated=20
  with the hi-targeted-to-uri. Although other values may be possible =
this draft=20
  defines two values for the Reason in such cases : </FONT></P>
  <P><FONT face=3DArial size=3D2>- Reason =3D'no response' when the =
retargeting occurs=20
  after timeout</FONT> <BR><FONT face=3DArial size=3D2>- Reason =3D =
'call forwarding=20
  immediate' when the proxy directly retargets to a default contact URI =
or to=20
  another AoR</FONT> <BR><FONT face=3DArial size=3D2>"</FONT> </P>
  <P><FONT face=3DArial size=3D2>S=E9bastien</FONT> =
</P></BLOCKQUOTE></BODY></HTML>

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From sip-bounces@ietf.org  Wed Nov 17 10:42:47 2004
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Subject: RE: [Sip] PING/PONG
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From: "Christian Stredicke" <Christian.Stredicke@snom.de>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>
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So my understanding is:

1. We should use *only* STUN for refreshing bindings (either UDP or TCP,
what about TLS)

2. The user agent indicates if it can do it (no negotiation on
capabilities)

3. Refreshing must be done from the client

Can we agree on that?

Christian

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: Tuesday, November 16, 2004 2:46 PM
> To: Christian Stredicke
> Cc: Spencer Dawkins; sip@ietf.org
> Subject: Re: [Sip] PING/PONG
>=20
> inline.
>=20
> Christian Stredicke wrote:
>=20
> > Hi Spencer!
> >=20
> > You just mentioned another way of keeping TCP connections=20
> alive. Thats
> > good!=20
> >=20
> > So far we have:
> >=20
> > - CRLF (frequenly used today, but not indicated by user agent)
> > - STUN (same as CRLF)
>=20
> It is not the same. STUN solicits a response, which allows=20
> the client to=20
> determine if the NAT binding has changed, and if the server=20
> is still alive.
>=20
> > - PING/PONG or whatever (tbd, maybe superflous)
>=20
> I am very much against SIP-based keepalives. This turns into a huge=20
> scalability nightmare. It ends up being the dominant=20
> processing cost for=20
> proxies along with the dominant percentage of the signaling=20
> bandwidth,=20
> and it buys you nothing.
>=20
>=20
> > - OPTIONS (big overhead, but compatible)
> > - REGISTER (same as OPTIONS, but even bigger overhead)
> > - TCP low-level (how can this be addressed by the application?)
> > - No refreshing ("legacy device")
> >=20
> > Lets have a way to indicate these options and let the SBC=20
> decide what
> > method it likes/supports.
>=20
> Generally speaking, having multiple choices reduces=20
> interoperability. I=20
> think we should pick a solution, and only describe multiple ones if=20
> there is a compelling reason why multiple ones need to exist.
>=20
> Its worth pointing out that there are several functions these various=20
> keepalvies provide:
>=20
> 1. they keep nat bindings fresh
> 2. they allow the client to detect if its nat binding as seen by the=20
> server has changed
> 3. they allow the client to determine if the server is still=20
> alive and=20
> responding to messages
> 4. they allow the client to determine if the nat has dropped=20
> its binding=20
> altogether, so that packets are just going into the ether
>=20
> I *think* Bob is proposing another usage:
>=20
> 5. they allow the server to determine that the nat binding for the=20
> client has changed, so it can update its registration to=20
> reflect the new=20
> address
>=20
>=20
> Using the keepalive to update the binding on the server introduces=20
> security risks. If the keepalive is not authenticated, an=20
> attacker can=20
> inject the keepalives with a falsified source IP address, and as a=20
> result force incoming calls to be directed towards it. This is not=20
> acceptable. Thus, any request which causes the server to=20
> update any kind=20
>   of bindings MUST be authenticated. I would rather not require the=20
> keepalives to be authenticated, as this substantially=20
> increases the cost=20
> of processing them. If you use unauthenticated STUN packets,=20
> the client=20
> detects that its nat binding has changed, and then it can re-register=20
> with a proper REGISTER, which can be authenticated and that=20
> can be used=20
> to update the binding, thus avoiding the attack.
>=20
> It is possible to use different mechanisms to address each of the 4=20
> different issues above. An empty UDP packet or a CRLF on the TCP=20
> connection can address the binding keepalive issue but not any of the=20
> liveness checks. I strongly prefer a single solution that=20
> addresses all=20
> of these.
>=20
> -Jonathan R.
>=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20
>=20
>=20

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Paul Kyzivat wrote:


> By "this" I assume you mean the to-from-change option?
> 
>  > only if you assume that the problem lies in the UA; I would
> 
>> be worred about other nasties along the way.
> 
> 
> For changing on subsequent requests within the dialog, what nasties are 
> you thinking of? I think only a call stateful proxy would notice, and it 
> seems there aren't many of those. Of course there are B2BUAs, but they 
> at least *ought* to be UAs and behave as such.

Ought, being the operative word here. I fear that a b2bua would pass the 
Supported headers unmodified and then things would be bad.

>> In any case, I don't see a need for a Supported header field. The 
>> feature is supposed to work with any rfc3261 client out there.
> 
> 
> Other than just hoping for the best, why would we expect existing 3261 
> clients to work with this change?


It is explicitly called out in 12.2.1.1:

> The URI in the To field of the request MUST be set to the remote URI
>    from the dialog state.  The tag in the To header field of the request
>    MUST be set to the remote tag of the dialog ID.  The From URI of the
>    request MUST be set to the local URI from the dialog state.  The tag
>    in the From header field of the request MUST be set to the local tag
>    of the dialog ID.  If the value of the remote or local tags is null,
>    the tag parameter MUST be omitted from the To or From header fields,
>    respectively.
> 
>       Usage of the URI from the To and From fields in the original
>       request within subsequent requests is done for backwards
>       compatibility with RFC 2543, which used the URI for dialog
>       identification.  In this specification, only the tags are used for
>       dialog identification.  It is expected that mandatory reflection
>       of the original To and From URI in mid-dialog requests will be
>       deprecated in a subsequent revision of this specification.



Only the tags are ever used in any normative processing of dialogs in 
RFC 3261. The requirement to set the From field *URI* in the reverse 
direction to the value of the To URI in the original INVITE is *only* 
for backwards compatibility, and rfc3261 stated that this is likely to 
change one day. The day appears to be at hand.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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inline.

Paul Kyzivat wrote:

>> So it isn't that the auth service will reject the request; it MAY just
>> forward the request normally. What will happen in that instance? Well, if
>> the auth service role is instantiated by a proxy server, it will 
>> forward the
>> request in accordance with the URI without adding an Identity header; it
>> won't challenge the request, or what have you.
> 
> 
> Good. You confirm what I was assuming.

The MAY isn't good enough here. You want to be certain it WONT reject 
the request just because the From is wrong.

> 
> I agree the basic request identity problem is separable from the others. 
> But it does seem that connected-party, response-identity, and reverse 
> direction requests in dialog are all entangled, and that changing 
> to/from may be a single solution good for all of them.

I see a clear line between the various problems, but perhaps its just 
because I've not thought about it enough ;)

One problem, which I am calling the "request identity problem", is how 
to identify the sender of a request. That problem exists for initial 
requests and mid-dialog requests alike. indeed, because the identity 
exists on a request by request basis, the fact that a request is part of 
a dialog has no bearing. If you buy this argument, it would strongly 
argue in favor of the approach I suggested - modifying the From for 
reverse requests in a dialog so that the From properly identifies the 
sender.

A separate problem is how a UAC can securely determine the sequence of 
events that led to a request being retargeted. This is the much harder 
problem that I think is out of scope right now.

Another problem is how to identify the sender of a response, and it is 
the exact inverse of the request identity problem.

> 
>> I have a number of arguments about this, but here are just three:
>>
>> - Identity will succeed in the marketplace if it can be used
>> opportunistically. Requiring UAs to change the From header field in this
>> fashion forces them to become identity-aware, which is an impediment to
>> adoption. Basically, we force UAs to be either backwards-compliant with
>> RFC2543 or compliant with Identity. Tough choice, especially given 
>> that the
>> UA might not be aware of its need to be either of those things.
> 
> 
> Agree.

This is a fair point. There is a middle ground though. If a UA doesn't 
set the From properly, the identity service simply doesn't sign the 
request. Thus, it is always OK for a client to do nothing to support 
identity. The consequence is that, in the case of retargeting, their 
identity won't be delivered to the caller for a mid-dialog request. I 
think that this is reasonable. However, if a UA wants, it can set the 
 From properly, and then these requests can be signed.

> 
>> - I also question whether or not the callee UA will even know which 
>> identity
>> it needs to stick in the From header of requests in the backwards 
>> direction:
>> after all, if the UA is registered under multiple AoRs, what exactly 
>> about
>> the dialog-forming request will tell it definitively which identity it is
>> supposed to use? While in some cases, this could be inferred from the
>> Request-URI of the dialog-forming requests, it won't be sufficient in all
>> cases, I think.  The previous hop (last proxy server to handle the 
>> request)
>> might also be a hint. Now, it could be that there's a good answer to this
>> question, but if so, we need to know that answer.
> 
> 
> There are many reasons why a UA with multiple registrations needs to 
> know which one was the source of an incoming call. And the UA can easily 
> achieve this by using distinct contacts in the registrations. I don't 
> think we need to lose sleep over those that choose not to do so.

I don't see the problem. It should be able to use any identity it might 
otherwise assert in a request outside of the dialog. If the route set 
doesn't include the identity service for that domain, then there won't 
be any asserted identity. What this means is that servers that provide 
the identity service probably should record-route.

> 
>> - Retargeting is inherently insecure, and no amount of wiggling with the
>> sip-identity mechanism will repair this. If Alice sent a request to 
>> Bob, and
>> a new request in the backwards direction comes from Edgar, I think 
>> there is
>> -no- circumstance in which Alice should not view this as a potential
>> security risk, regardless of the presence or absence of an Identity 
>> header,
>> SIPS, or what have you. Trying to explain this aware flies in the face of
>> common sense, and will be incomprehensible to actual users who will 
>> need to
>> make security decisions based on the evidence collected from the 
>> protocol by
>> their user agent.
> 
> 
> Of course it is a potential security risk, which is why it is good to 
> report that it happened. (A good UA will hopefully note that the 
> connected party is not the called party and make a big fuss at its UI.) 
> But it is still good to know who in fact you ended up talking to.

I agree. Indeed, with SIP today you not only don't know its been 
retargeted, nothing evne identifies who its been retargeted to. This 
seems even more of a risk.


>> Don't forget, for example, that SIPS does not apply after the domain
>> indicated by the Request-URI of the request, and so on. SIPS is not a
>> powerful mechanism. Until we all acknowledge that we were mistaken in our
>> formulation in RFC3261, and that SIPS, when present, signifies 
>> unambiguously
>> that TLS should be used end-to-end, I do not think SIPS can be said to 
>> solve
>> any of our problems. Once we have the connect-reuse mechanism nailed down
>> firmly, I think it will be possible to shift to that understanding of 
>> SIPS,
>> if we want to.

SIPS is clear that there MUST be a secure channel at each hop - it just 
doesn't mandate TLS. In hindsight I agree it should have ideally done 
that, it does imply a security property which does, IMHO, prevent 
outsiders from launching the attack you describe.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From: Jonathan Rosenberg <jdrosen@cisco.com>
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To: "Peterson, Jon" <jon.peterson@neustar.biz>
Subject: Re: third alternative (was RE: [Sip] RE: Identity after reinvite)
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inline.

Peterson, Jon wrote:

> I would like to suggest a third alternative:
> 
> 3) Change nothing about in To/From. Requests in the backwards direction in a
> dialog may go through the authentication service of the domain indicated in
> the host portion of the From header of the request. If that is not possible,
> do not supply request identity for such messages.

This is pretty close to what I had just suggested in a mail I wrote a 
few minutes ago. I should have read the whole thread before responding I 
suppose ;)

The difference between this and what I had proposed is that I think a UA 
should be *allowed* to change the From in its outbound request.

> 
> While this seem like a non-starter in retargeting cases, requests get
> retargeted for a reason. If the dialog-forming request was appropriately
> delivered to the domain indicated in the Request-URI (biloxi.example.com),
> and that domain chose to forward the request to another domain
> (chicago.example.com), it did so because some entity was authorized to add
> the chicago URI as a contact for the biloxi URI. In other words, this
> approach views retargeting as if the callee's UA were actually registered at
> biloxi "through" chicago, and that when the UA receives a dialog-forming
> request, it is the To header that determines who should assert identity for
> new requests in the backwards direction. In cases where the user contacted
> at chicago does not possess credentials to authenticate themselves to
> biloxi, I think it would be fair to say that requests in the backwards
> direction should not get an Identity header. 

I think that it will almost never have such credentials. If it did, this 
wouldn't be retargeting.


> To go any further, I think,
> requires us to solve the response identity problem. This approach is, I
> think, forward-compatible with any reasonable solution to the response
> identity problem, though.
> 
> The primary challenge of this approach is insuring that requests in the
> backwards direction will actually go through the authentication service of
> biloxi.com, since in many cases they ordinarily might not.

I don't think you need to worry about that, since I don't think the 
client that is contacted at chicago.com will ever really have 
credentials for biloxi. It is far more likely that it will have a tls 
connection to chicago.com and its authentication service.


  If biloxi.com
> Record-Routes its auth service in the dialog-forming request, this would
> provide the right assurance (and seems plausible from a deployment
> perspective); however, if chicago.com also Record-Routes itself, for
> example, then the callee will not be able to form a direct TLS connection to
> biloxi.com. It is also possible that the callee's UA could force a Route
> header to connect to biloxi.com first, but this requires the callee's UA to
> be identity-aware, which is undesirable when sip-identity is applied
> opportunistically.
> 
> But it isn't immediately clear how this is any different from normal cases
> of request identity outside the scope of a dialog. There are any number of
> possible reasons why the calling UA might not be able to connect directly
> via TLS to its auth service (including interposing intermediaries and/or the
> inability of the UA to force a Route through their auth service), and the
> draft currently just says that we shouldn't provide Identity when we cannot
> get the needed relationship between the UA and auth service. I've gotten a
> lot of feedback (from Paul, Kumiko and others) that we should try to weaken
> the requirement for direct TLS, and I intend to add some text to the next
> version about some specific ways in which it is still acceptable to provide
> identity even if a direct TLS connection cannot be formed. Certain forms of
> weakness there could be beneficial to this approach.

I think that having non-direct connections is a reasonable thing; 
generally my view has been that it should be possible to disaggregate a 
monolithic proxy into components in the same network, each of which is a 
SIP proxy performing parts of the overall processing in a domain. As 
long as appropriate security and trust relationships are maintained 
between these component proxies in a domain, I think equivalent security 
is afforded.

As a follow up to my previous note, I tihnk part of the difficulty here 
is a difference in view about the problem being solved by asserted 
identities in the reverse direction. In my view, the sips mechanism 
provides an assurance ot the caller that the call got to the eventual 
target as a result of authorized forwarding at each hop along the way. 
Thus, the identity of requests in the reverse direction is not used to 
*authorize* whether or not the request should be processed. Rather, it 
serves merely as a reliable identification mechanism, as authorization 
is provided by sips.

-JOnathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org  Wed Nov 17 11:39:26 2004
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Subject: RE: [Sip] Boss-Secretary functionalities over SIP
Date: Wed, 17 Nov 2004 16:21:26 -0000
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Hi,

There are various ways of offering a 'Boss-Secretary' feature. The
simplest, which we can call shared-line-monitor, would involve using the
dialog event package (see
http://www.ietf.org/proceedings/04aug/I-D/draft-ietf-sipping-dialog-pack
age-04.txt) to show the status of the monitored (the boss's) line. The
SIP service examples ID
(http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples
-07.txt) takes this a step further in its 'Single Line Extension'
example, by letting another user join a monitored call.

A more popular requirement, in my experience, permits shared lines/AORs
to be 'seized' by any one of the users that have registered against that
AOR. Seizing a line effectively involves obtaining a mutex on the shared
line resource so that no other user can make a call out on that instance
of the AOR. This has a number of benefits when it comes to managing
calls on the shared line. The ID 'Implementing Bridged Line Appearances
Using Session Initiation Protocol' makes use of the dialog event package
to provide the line seize feature
(http://ietfreport.isoc.org/all-ids/draft-anil-sipping-bla-01.txt) -
this draft has now expired, but an updated version should be out soon.

Paul.

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Pianigiani, Jacopo (Jacopo)
Sent: Tuesday, November 16, 2004 5:10 PM
To: 'sip@ietf.org'
Subject: [Sip] Boss-Secretary functionalities over SIP


Hi all,
I am looking for an overview on SIP of the classic PABX Boss-Secretary
functionality from the high level call flow perspective.
Does anyone know where I could find this ?=20
thanks in advance and kind regards

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From sip-bounces@ietf.org  Wed Nov 17 11:44:25 2004
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From: "Thomas Gal" <ThomasGal@LumenVox.com>
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Subject: RE: [Sip] How to get  old messages on archives!!
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-Tom

thomasgal@lumenvox.com  

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of vikram bhuskute
> Sent: Wednesday, November 17, 2004 1:51 AM
> To: sip@ietf.org
> Subject: [Sip] How to get old messages on archives!!
> 
> Hello ,
>                I am an active member of this work group. It 
> so happened that i deleted some old mails.
> Earlier i Use to see Old mails on the Archives link. But now 
> if i open ,it give today's and yesterday's mail.
> I am force to go to link
> http://www1.ietf.org/mail-archive/web/sip/current/index.html 
> and there is no other link there :( Please help !!
> 
> 
> Vikram
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip
> 


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Christian Stredicke wrote:

> So my understanding is:
> 
> 1. We should use *only* STUN for refreshing bindings (either UDP or TCP,
> what about TLS)

Yes, TLS too. TLS runs ontop of TCP after all.

> 
> 2. The user agent indicates if it can do it (no negotiation on
> capabilities)
> 
> 3. Refreshing must be done from the client
> 
> Can we agree on that?

Yes.

Server originating refreshing is in the wrong direction. It is the 
client utilizing the service; if it wishes to make use of that service, 
it has to be responsible for refreshing the connection.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.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


From sip-bounces@ietf.org  Wed Nov 17 12:31:21 2004
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Subject: RE: [Sip] PING/PONG
Date: Wed, 17 Nov 2004 16:58:11 -0000
Message-ID: <CD9775120D600F43B9C50329395E9DB64F3213@ivor.citel.com>
Thread-Topic: [Sip] PING/PONG
Thread-Index: AcTMxdvvfCy/R4b+R7CteRr8GeoUkQAALiag
From: "Steve Langstaff" <steve.langstaff@citel.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>,
        "Christian Stredicke" <Christian.Stredicke@snom.de>
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Should point 2 read "The user agent server indicates if it can respond =
to client refresh requests"?

--
Steve Langstaff.


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Jonathan Rosenberg
Sent: 17 November 2004 16:40
To: Christian Stredicke
Cc: sip@ietf.org
Subject: Re: [Sip] PING/PONG




Christian Stredicke wrote:

> So my understanding is:
>=20
> 1. We should use *only* STUN for refreshing bindings (either UDP or =
TCP,
> what about TLS)

Yes, TLS too. TLS runs ontop of TCP after all.

>=20
> 2. The user agent indicates if it can do it (no negotiation on
> capabilities)
>=20
> 3. Refreshing must be done from the client
>=20
> Can we agree on that?

Yes.

Server originating refreshing is in the wrong direction. It is the=20
client utilizing the service; if it wishes to make use of that service,=20
it has to be responsible for refreshing the connection.

-Jonathan R.

--=20
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.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

_______________________________________________
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: "'PROUVOST Sebastien RD-CORE-ISS'" <sebastien.prouvost@francetelecom.com>,
        GARCIN Sebastien RD-CORE-ISS <sebastien.garcin@francetelecom.com>,
        "Jesske, R" <R.Jesske@t-com.net>, sip@ietf.org
Subject: RE: [Sip] Privacy statements and History(draft-ietf-sip-history-i
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Hi all,
=20
I've again snipped the thread, but did want to propose the change that =
I
think will satisfy the concerns raised.
=20
The change proposed is to change the last statement in the first =
paragraph
in section 4.3.3.1.1 from:
"...the proxy MUST remove any hi-entry(s) prior to forwarding."
to:
"...the proxy SHOULD remove any hi-entry(s) prior to forwarding, =
depending
upon local policy and whether the proxy might know apriori that it can =
rely
on a downstream privacy service to apply the requested privacy."=20
=20
So, unless further concerns are raised on this proposed change, I'll =
plan on
incorporating that with any other last call comments in the -05 =
version.
=20
Mary=20


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
PROUVOST Sebastien RD-CORE-ISS
Sent: Thursday, November 11, 2004 8:44 AM
To: Barnes, Mary [NGC:B601:EXCH]; GARCIN Sebastien RD-CORE-ISS; Jesske, =
R;
sip@ietf.org
Subject: RE: [Sip] Privacy statements and
History(draft-ietf-sip-history-info-04.txt)


Mary, Sebastien, Roland,=20
=20
I agree that we should let the possibility for a proxy not to remove =
the
hi-entry even if privacy is requested and even if the request is =
forwarded
to a Request-URI associated with a domain for which the proxy is not
responsible (if there is an agreement between the domains that ensures =
the
proxy that privacy will be applied to the request).=20
I suggest a text that would look like (in case privacy is requested): =
"the
hi-entry SHOULD be removed by the proxy unless it knows that it can =
rely on
a downstream privacy service to apply the requested privacy ".
=20
Sebastien.


  _____ =20

De : sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] De la part de =
Mary
Barnes
Envoy=E9 : mercredi 10 novembre 2004 19:54
=C0 : GARCIN Sebastien RD-CORE-ISS; Jesske, R; sip@ietf.org
Objet : RE: [Sip] Privacy statements and History
(draft-ietf-sip-history-info-04.txt)



I still contend that the SHOULD is sufficient.  SHOULD means that in =
general
processing, if the hi-entry has a privacy header, then the usual =
processing
would be that it would be removed (i.e it was added for a specific =
reason
and in general should be used to remove the entries based on well =
defined
criteria).  If there are reasons, such as local policy, that would =
allow the
forwarding in specific cases, then it's okay that it is forwarded.  I =
think
the use of MAY results in less precision and I think the value and =
critera
for associating and removing the privacy header with the hi-entry =
becomes
much less clear. =20

I'd like to hear more opinions on this topic, prior to agreeing to =
making
the change (from the MUST to MAY rather than MUST to SHOULD).=20

Regards,=20
Mary=20


-----Original Message-----=20
From: GARCIN Sebastien RD-CORE-ISS
[mailto:sebastien.garcin@francetelecom.com
<mailto:sebastien.garcin@francetelecom.com> ]=20
Sent: Wednesday, November 10, 2004 12:28 PM=20
To: Barnes, Mary [NGC:B601:EXCH]; Jesske, R; sip@ietf.org=20
Cc: VL-T-Com-T-TE332@vli.telekom.de=20
Subject: RE: [Sip] Privacy statements and History
(draft-ietf-sip-history-info-04.txt)=20


Mary and roland,=20

Actually, the statement regarding the forwarding of hi-entries beyond =
the
domain for which the proxy is responsible should rather be a matter of =
local
policy and thus a MAY should be used instead of SHOULD. We are =
potentially
dealing with network boundaries where agreements for forwarding such =
kind
information can be reached.=20

Section 4.3.3.1.1=20
This section should be re-reworded in accordance with the statement =
above (I
can provide some text is we can agree).=20

Section 4.3.3.1 is ok (apologies for not being explicit)=20

Best regards,=20
s=E9bastien=20

                                =20

                                ------Remainder of thread has been =
deleted
by Mary------------------



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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; charset=3Diso-885=
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<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D514513817-=
17112004>Hi=20
all,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D514513817-17112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D514513817-=
17112004>I've=20
again snipped the thread, but did want to propose the change that I think=
 will=20
satisfy the concerns raised.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D514513817-17112004></SPAN></FONT><FONT face=3DArial color=3D#0000=
ff=20
size=3D2><SPAN class=3D514513817-17112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D514513817-=
17112004>The=20
change proposed is to change the last statement in the first paragraph in=
=20
section 4.3.3.1.1 from:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D514513817-17112004>"...<FONT size=3D3><FONT color=3D#000000><FONT=
=20
face=3D"Courier New">the proxy MUST remove any hi-entry(s) prior to=20
forwarding."</FONT></FONT></FONT></SPAN></FONT></DIV>
<DIV><FONT><SPAN class=3D514513817-17112004><FONT><FONT face=3DArial colo=
r=3D#0000ff=20
size=3D2>to:</FONT></FONT></SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D514513817-=
17112004><FONT=20
size=3D3><FONT color=3D#000000><FONT face=3D"Courier New">"...the proxy&n=
bsp;SHOULD=20
remove any hi-entry(s) prior to forwarding, depending upon local policy&n=
bsp;and=20
whether the proxy might know apriori that it can rely on a downstream pri=
vacy=20
service to apply the requested privacy."<SPAN=20
style=3D"mso-spacerun: yes">&nbsp;</SPAN></FONT></FONT></FONT></SPAN></FO=
NT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D514513817-=
17112004><FONT=20
size=3D3><FONT color=3D#000000><FONT face=3D"Courier New"><SPAN=20
style=3D"mso-spacerun: yes"></SPAN></FONT></FONT></FONT></SPAN></FONT>&nb=
sp;</DIV>
<DIV><FONT><SPAN class=3D514513817-17112004><FONT><FONT face=3DArial colo=
r=3D#0000ff=20
size=3D2><SPAN style=3D"mso-spacerun: yes">So, unless further concerns ar=
e raised on=20
this proposed change, I'll plan on incorporating that with any other last=
 call=20
comments in the -05 version.</SPAN></FONT></FONT></SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D514513817-=
17112004><FONT=20
size=3D3><FONT color=3D#000000><FONT face=3D"Courier New"><SPAN=20
style=3D"mso-spacerun: yes"></SPAN></FONT></FONT></FONT></SPAN></FONT>&nb=
sp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D514513817-17112004><FONT=20
size=3D3><FONT><FONT face=3D"Courier New"><SPAN=20
style=3D"mso-spacerun: yes"></SPAN></FONT></FONT></FONT></SPAN></FONT><FO=
NT=20
color=3D#0000ff><SPAN lang=3Den-us><FONT face=3DArial size=3D2>Mary</FONT=
></SPAN>=20
</FONT></DIV>
<BLOCKQUOTE style=3D"MARGIN-RIGHT: 0px">
  <DIV><FONT color=3D#0000ff></FONT></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><=
FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
  sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B>On Behalf Of=20
  </B>PROUVOST Sebastien RD-CORE-ISS<BR><B>Sent:</B> Thursday, November 1=
1, 2004=20
  8:44 AM<BR><B>To:</B> Barnes, Mary [NGC:B601:EXCH]; GARCIN Sebastien=20
  RD-CORE-ISS; Jesske, R; sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Priva=
cy=20
  statements and=20
History(draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2>Mary, Sebastien, Roland, </FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2>I agree that we should let the possibility for=
 a proxy=20
  not to remove the hi-entry even if privacy is requested and even if the=
=20
  request is forwarded to a Request-URI associated with a domain for whic=
h the=20
  proxy is not responsible&nbsp;(if there&nbsp;is an agreement&nbsp;betwe=
en=20
  the&nbsp;domains that ensures the proxy that privacy will be applied to=
 the=20
  request). </FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2>I suggest a&nbsp;text that would look like&nbs=
p;(in case=20
  privacy is requested): "the hi-entry SHOULD be removed by the proxy unl=
ess it=20
  knows that it can rely on a downstream privacy service to apply the req=
uested=20
  privacy ".</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2>Sebastien.</FONT></SPAN></DIV><FONT face=3DAri=
al=20
  color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D#0000ff size=
=3D2></FONT><FONT=20
  face=3DArial color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D=
#0000ff=20
  size=3D2></FONT><FONT face=3DArial color=3D#0000ff size=3D2></FONT><BR>
  <DIV class=3DOutlookMessageHeader lang=3Dfr dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>De&nbsp;:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>De la part de</B> Mary=20
  Barnes<BR><B>Envoy=E9&nbsp;:</B> mercredi 10 novembre 2004=20
  19:54<BR><B>=C0&nbsp;:</B> GARCIN Sebastien RD-CORE-ISS; Jesske, R;=20
  sip@ietf.org<BR><B>Objet&nbsp;:</B> RE: [Sip] Privacy statements and Hi=
story=20
  (draft-ietf-sip-history-info-04.txt)<BR></FONT><BR></DIV>
  <DIV></DIV>
  <P><FONT size=3D2>I still contend that the SHOULD is sufficient.&nbsp; =
SHOULD=20
  means that in general processing, if the hi-entry has a privacy header,=
 then=20
  the usual processing would be that it would be removed (i.e it was adde=
d for a=20
  specific reason and in general should be used to remove the entries bas=
ed on=20
  well defined criteria).&nbsp; If there are reasons, such as local polic=
y, that=20
  would allow the forwarding in specific cases, then it's okay that it is=
=20
  forwarded.&nbsp; I think the use of MAY results in less precision and I=
 think=20
  the value and critera for associating and removing the privacy header w=
ith the=20
  hi-entry becomes much less clear.&nbsp; </FONT></P>
  <P><FONT size=3D2>I'd like to hear more opinions on this topic, prior t=
o=20
  agreeing to making the change (from the MUST to MAY rather than MUST to=
=20
  SHOULD). </FONT></P>
  <P><FONT size=3D2>Regards, </FONT><BR><FONT size=3D2>Mary</FONT> </P><B=
R>
  <P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT size=3D2>=
From:=20
  GARCIN Sebastien RD-CORE-ISS [<A=20
  href=3D"mailto:sebastien.garcin@francetelecom.com">mailto:sebastien.gar=
cin@francetelecom.com</A>]=20
  </FONT><BR><FONT size=3D2>Sent: Wednesday, November 10, 2004 12:28 PM</=
FONT>=20
  <BR><FONT size=3D2>To: Barnes, Mary [NGC:B601:EXCH]; Jesske, R;=20
  sip@ietf.org</FONT> <BR><FONT size=3D2>Cc:=20
  VL-T-Com-T-TE332@vli.telekom.de</FONT> <BR><FONT size=3D2>Subject: RE: =
[Sip]=20
  Privacy statements and History (draft-ietf-sip-history-info-04.txt)</FO=
NT>=20
  </P><BR>
  <P><FONT size=3D2>Mary and roland,</FONT> </P>
  <P><FONT size=3D2>Actually, the statement regarding the forwarding of h=
i-entries=20
  beyond the domain for which the proxy is responsible should rather be a=
 matter=20
  of local policy and thus a MAY should be used instead of SHOULD. We are=
=20
  potentially dealing with network boundaries where agreements for forwar=
ding=20
  such kind information can be reached. </FONT></P>
  <P><FONT size=3D2>Section 4.3.3.1.1</FONT> <BR><FONT size=3D2>This sect=
ion should=20
  be re-reworded in accordance with the statement above (I can provide so=
me text=20
  is we can agree).</FONT> </P>
  <P><FONT size=3D2>Section 4.3.3.1 is ok (apologies for not being=20
  explicit)</FONT> </P>
  <P><FONT size=3D2>Best regards,</FONT> <BR><FONT size=3D2>s=E9bastien</=
FONT> </P>
  <P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT=20
  size=3D2> </FONT></P>
  <P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT size=3D2>------Remaind=
er of=20
  thread has been deleted by=20
Mary------------------</FONT></P><BR></BLOCKQUOTE></BODY></HTML>

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--===============0857168419==--



From sip-bounces@ietf.org  Wed Nov 17 13:07:16 2004
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Subject: RE: [Sip] PING/PONG
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Thread-Topic: [Sip] PING/PONG
thread-index: AcTMxdvvfCy/R4b+R7CteRr8GeoUkQAALiagAAD4T3A=
From: "Christian Stredicke" <Christian.Stredicke@snom.de>
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        "Jonathan Rosenberg" <jdrosen@cisco.com>
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Well, thats a good question. I think everybody should prepare their
server for responding to STUN requests on the SIP port. We do since two
years.-)

CS=20

> -----Original Message-----
> From: Steve Langstaff [mailto:steve.langstaff@citel.com]=20
> Sent: Wednesday, November 17, 2004 12:16 PM
> To: Jonathan Rosenberg; Christian Stredicke
> Cc: sip@ietf.org
> Subject: RE: [Sip] PING/PONG
>=20
> Should point 2 read "The user agent server indicates if it=20
> can respond to client refresh requests"?
>=20
> --
> Steve Langstaff.
>=20
>=20
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On=20
> Behalf Of Jonathan Rosenberg
> Sent: 17 November 2004 16:40
> To: Christian Stredicke
> Cc: sip@ietf.org
> Subject: Re: [Sip] PING/PONG
>=20
>=20
>=20
>=20
> Christian Stredicke wrote:
>=20
> > So my understanding is:
> >=20
> > 1. We should use *only* STUN for refreshing bindings (either UDP or=20
> > TCP, what about TLS)
>=20
> Yes, TLS too. TLS runs ontop of TCP after all.
>=20
> >=20
> > 2. The user agent indicates if it can do it (no negotiation on
> > capabilities)
> >=20
> > 3. Refreshing must be done from the client
> >=20
> > Can we agree on that?
>=20
> Yes.
>=20
> Server originating refreshing is in the wrong direction. It=20
> is the client utilizing the service; if it wishes to make use=20
> of that service, it has to be responsible for refreshing the=20
> connection.
>=20
> -Jonathan R.
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20
>=20
>=20

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Subject: Re: [Sip] -RP-05.txt: section 7.1 and (rough) consensus debate
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At 08:03 AM 11/17/2004 -0500, Mpierce1@aol.com wrote:
>In a message dated 11/16/2004 7:42:29 PM W. Europe Standard Time, 
>jmpolk@cisco.com writes:
>> >[MAP] Since the document should say that it does not address interworking
>> >between namespaces, it should not contain this material.
>>
>>giving examples to clarify a known area of confusion is a good thing - and
>>will stay in the document.
>
>[MAP] The above results in a serious contradiction. You agree that the 
>document should not describe dependencies, then include material that 
>certainly describes dependencies (what combinations are and are not allowed).

Mike, pay attention to the ordering of two namespaces in the section - 
there are ZERO dependencies between namespaces, just that the order within 
each namespace MUST be maintained regardless of where one value from one 
namespace is places in relative priority order to one or more from another 
namespace. Implementors MUST have one order - no matter how many namespaces 
are supported at a server/endpoint - to relieve confusion of which is 
prioritized over or under another. This is the same rule that applies 
within a namespace: there MUST be one order of priority, period (No 
wildcards) to ensure interoperability.

>This will confuse more readers into thinking that there are some 
>dependencies.

Because it is confusing you does not mean it is confusing others. Every 
single coder I've talked to on this subject understands once they have seen 
these examples. That makes them worthy for inclusion.

>>it will not be deleted, though it might be modified to add or clarify 
>>meaning
>[MAP] It's too bad that you're taking the position that you have the final 
>say on what is and is not included in the document regardless of the 
>comments and results of any e-mail discussion.

This email discussion is from you, not from a rough consensus of the list.

>As far as I'm concerned, you must get the WG consensus

The IETF mantra is "rough consensus and running code"

You don't represent rough consensus - despite taking over the microphone 
last week

>  for anything that you added in -05 in order to retain it in -06.

Are you saying I cannot submit a document without getting consensus from 
the list?

>Contrary to what Jon Peterson stated at the meeting

Contrary to what an Area Director said to you in a meeting.... was that 
before or after you threatened the IETF with US Government force if the 
IETF didn't comply with your demands?  remember - about 15 people heard you 
do this

>, RFC2418 defines the responsibilities of the editor as follows:
>
>   Most IETF working groups focus their efforts on a document, or set of
>   documents, that capture the results of the group's work.  A working
>   group generally designates a person or persons to serve as the Editor
>   for a particular document.  The Document Editor is responsible for
>   ensuring that the contents of the document accurately reflect the
>   decisions that have been made by the working group.

If you feel this way, you have a right to complain to the WG chairs



>Mike


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Subject: Re: [Sip] -RP-05.txt: section 7.1 and (rough) consensus debate
To: "James M. Polk" <jmpolk@cisco.com>
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Technical question about ordering of the namespaces and values

One approach we have used (in WPS) is

a) When the resource pool is relatively small (e.g., radio traffic
channels) we serve the  queue based on priority order first, and time stamp
second.

b) When the resource pool is large (e.g. landline trunk groups) we  serve
the queue (amongst all and only the "labeled" calls) by time stamp, without
regard for the actual priority level.

Does the text in section 7.1 PREVENT me from serving SIP messages WITH
regard to BOTH the namespace and value  (and timestamp) in some contexts,
and with regard to ONLY the namespace (and timestamp), but NOT the value in
other contexts?

I can foresee work arounds if it does prevent it, but I would be happier if
that behavior were NOT prevented.

Janet


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                      "James M. Polk"                                                                                        
                      <jmpolk                  To:      Mpierce1@aol.com                                                     
                      @cisco.com>              cc:      An.Nguyen@ncs.gov, Janet P Gunn/FED/CSC@CSC, oran@cisco.com,         
                                               sip@ietf.org, Dean Willis <dean.willis@softarmor.com>, rohan@ekabal.com,      
                      11/17/2004 03:36         "Pete Babendreier" <pbabendr@cisco.com>, "Scott Morrison"                     
                      PM                       <scmorris@cisco.com>                                                          
                                               Subject: Re: [Sip] -RP-05.txt: section 7.1 and (rough) consensus debate       
                                                                                                                             




At 08:03 AM 11/17/2004 -0500, Mpierce1@aol.com wrote:
>In a message dated 11/16/2004 7:42:29 PM W. Europe Standard Time,
>jmpolk@cisco.com writes:

snip

Mike, pay attention to the ordering of two namespaces in the section -
there are ZERO dependencies between namespaces, just that the order within
each namespace MUST be maintained regardless of where one value from one
namespace is places in relative priority order to one or more from another
namespace. Implementors MUST have one order - no matter how many namespaces

are supported at a server/endpoint - to relieve confusion of which is
prioritized over or under another. This is the same rule that applies
within a namespace: there MUST be one order of priority, period (No
wildcards) to ensure interoperability.

snip

cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented






_______________________________________________
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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Thread-Topic: RFC 3261: BUG or clarification is needed
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From: "Orit Levin" <oritl@microsoft.com>
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Subject: [Sip] RFC 3261: BUG or clarification is needed
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Guys,
I would like to get a clarification on the properties of the Contact
header in requests and responses different from REGISTER.
Is there a common understanding, that a Contact header can contain any
valid SIP URI that, especially when used outside the original dialog,
needs to be resolved according to RFC 3263 (i.e. including the SRV
lookup) ?
=20
I couldn't find any statement along these lines in any of the relevant
RFCs.
=20
For example, in Redirect requests, we are reusing the Contact header in
order to point to an alternative destination which can have nothing in
common with the original domain.
=20
Yet another example, the current GRUU draft is silent about the ability
of resolving GRUU using RFC 3263 procedures. From the draft:
"the GRUU MUST exhibit the following properties:

   o  The domain part of the URI is an IP address present on the public

      Internet, or, if it is a host name, exists in the global DNS and

      corresponds to an IP address present on the public Internet."
=20
=20
For the interoperability sake, I believe that it is very important to
file this clarification against RFC 3261 and also point to RFC 3263 for
GRUU properties/location procedures.
=20
Thanks,
Orit.

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2800.1476" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D154024020-16112004>Guys,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D154024020-16112004>I =
would like to=20
get&nbsp;a clarification on the properties of the Contact header in =
requests and=20
responses different from REGISTER.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D154024020-16112004>Is =
there a common=20
understanding, that&nbsp;a&nbsp;Contact header can contain any valid SIP =
URI=20
that, especially&nbsp;when used outside the original dialog, needs to be =

resolved according to RFC <FONT size=3D3>3263</FONT> =
(i.e.&nbsp;including the SRV=20
lookup) ?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D154024020-16112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D154024020-16112004>I =
couldn't find any=20
statement along these lines in any of the relevant =
RFCs.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D154024020-16112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D154024020-16112004>For =
example, in=20
Redirect requests, we are reusing the Contact header in order to point =
to an=20
alternative destination which can have nothing in common with the =
original=20
domain.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D154024020-16112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D154024020-16112004>Yet=20
another&nbsp;example, the current GRUU draft&nbsp;is silent =
about&nbsp;the=20
ability of resolving GRUU using RFC&nbsp;<FONT size=3D3>3263</FONT> =
procedures.=20
>From the draft:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D154024020-16112004>"<FONT =

face=3D"Courier New">the GRUU MUST exhibit the following =
properties:</FONT></DIV>
<P class=3DMsoPlainText style=3D"MARGIN: 0in 0in 0pt"><FONT =
face=3D"Courier New"><SPAN=20
style=3D"mso-spacerun: yes">&nbsp;&nbsp; </SPAN>o<SPAN=20
style=3D"mso-spacerun: yes">&nbsp; </SPAN>The domain part of the URI is =
an IP=20
address present on the public</FONT></P>
<P class=3DMsoPlainText style=3D"MARGIN: 0in 0in 0pt"><FONT =
face=3D"Courier New"><SPAN=20
style=3D"mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</SPAN>Internet, or, if=20
it is a host name, exists in the global DNS and</FONT></P>
<DIV><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Courier New'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-US; =
mso-fareast-language: EN-US; mso-bidi-language: AR-SA"><SPAN=20
style=3D"mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</SPAN>corresponds to=20
an IP address present on the public Internet."</SPAN></DIV>
<DIV><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Courier New'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-US; =
mso-fareast-language: EN-US; mso-bidi-language: =
AR-SA"></SPAN>&nbsp;</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D154024020-16112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D154024020-16112004>For =
the=20
interoperability sake, I believe that it is very important to file this=20
clarification against RFC&nbsp;3261 and also&nbsp;point to RFC =
3263&nbsp;for=20
GRUU&nbsp;properties/location procedures.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D154024020-16112004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D154024020-16112004>Thanks,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D154024020-16112004>Orit.</SPAN></FONT></DIV></BODY></HTML>

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--===============1589178438==--



From sip-bounces@ietf.org  Wed Nov 17 17:22:09 2004
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Date: Wed, 17 Nov 2004 15:47:37 -0600
To: Janet P Gunn <jgunn6@csc.com>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] -RP-05.txt: section 7.1 on priority-value
  interleaving
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Section 7.1 is merely to give examples to readers that show the order 
within a namespace must be maintained regardless of how many other 
namespaces exist (that are supported) by that SIP element.

The section also provides examples of how the orders can be in multiple 
combinations of groupings, as long as the overall order within each 
namespace is not changed.

          Foo.3        Foo.3       Bar.C
          Foo.2        Bar.C       Foo.3
          Foo.1   or   Foo.2   or  Foo.2
          Bar.C        Bar.B       Foo.1
          Bar.B        Foo.1       Bar.B
          Bar.A        Bar.A       Bar.A

In each of these examples from section 7.1, the order of Foo.* remains the 
same (3 then 2 then 1), as does the order for Bar.* (C then B then A). 
Where each individual priority-value is relative to the other namespace 
priority-values is NOT specified (i.e where 3 fits into C,B,A, or where B 
fits into 3,2,1). The interleaved order of multiple namespaces is up to 
implementations and local policy to decide.

The only aspect you are PREVENTED from doing in this regard (to comply with 
this spec - whenever that is) is reordering priority-values within the same 
namespace no matter how many namespaces are supported.

 From a slightly different angle, you can consider two priority-values from 
two namespaces equivalent. The example given is:

      Bar.C    (highest priority)
  Foo.3  Bar.B <= these 2 are considered equivalent)
  Foo.2  Bar.A <= these 2 are considered equivalent)
      Foo.1    (lowest priority)

In this case, you can use some other indication to determine within your 
network if Foo.3 is to be treated above or below Bar.B - or you can leave 
them as identical in treatment.

other answers below

At 04:13 PM 11/17/2004 -0500, Janet P Gunn wrote:
>Technical question about ordering of the namespaces and values
>
>One approach we have used (in WPS) is
>
>a) When the resource pool is relatively small (e.g., radio traffic
>channels) we serve the  queue based on priority order first, and time stamp
>second.

ok


>b) When the resource pool is large (e.g. landline trunk groups) we  serve
>the queue (amongst all and only the "labeled" calls) by time stamp, without
>regard for the actual priority level.

the order of priority-value cannot change - if you wish to comply with this 
specification (once it becomes one). I know you plan on primarily using 1 
of your 5 priority levels. Within the same level, you can use whatever you 
want to apply whatever policy/treatment you want and still comply with this 
spec (once it becomes one).


>Does the text in section 7.1 PREVENT me from serving SIP messages WITH
>regard to BOTH the namespace and value  (and timestamp) in some contexts,
>and with regard to ONLY the namespace (and timestamp), but NOT the value in
>other contexts?

no prevention here


>I can foresee work arounds if it does prevent it,

it doesn't

>but I would be happier if
>that behavior were NOT prevented.

it's not

Are you happier?


>Janet
>
>
>----------------------------------------------------------------------------------------
>
>This is a PRIVATE message. If you are not the intended recipient, please
>delete without copying and kindly advise us by e-mail of the mistake in
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>for such purpose.
>----------------------------------------------------------------------------------------
>
>
>
>
> 
>
>                       "James M. 
> Polk" 
>
>                       <jmpolk                  To:      Mpierce1@aol.com 
 >
>                       @cisco.com>              cc: 
> An.Nguyen@ncs.gov, Janet P Gunn/FED/CSC@CSC, oran@cisco.com,
>                                                sip@ietf.org, Dean Willis 
> <dean.willis@softarmor.com>, rohan@ekabal.com,
>                       11/17/2004 03:36         "Pete Babendreier" 
> <pbabendr@cisco.com>, "Scott Morrison"
>                       PM                       <scmorris@cisco.com> 
 >
>                                                Subject: Re: [Sip] 
> -RP-05.txt: section 7.1 and (rough) consensus debate
> 
>
>
>
>
>
>At 08:03 AM 11/17/2004 -0500, Mpierce1@aol.com wrote:
> >In a message dated 11/16/2004 7:42:29 PM W. Europe Standard Time,
> >jmpolk@cisco.com writes:
>
>snip
>
>Mike, pay attention to the ordering of two namespaces in the section -
>there are ZERO dependencies between namespaces, just that the order within
>each namespace MUST be maintained regardless of where one value from one
>namespace is places in relative priority order to one or more from another
>namespace. Implementors MUST have one order - no matter how many namespaces
>are supported at a server/endpoint - to relieve confusion of which is
>prioritized over or under another. This is the same rule that applies
>within a namespace: there MUST be one order of priority, period (No
>wildcards) to ensure interoperability.
>
>snip
>
>cheers,
>James
>
>                                 *******************
>                  Truth is not to be argued... it is to be presented


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Subject: Re: [Sip] -RP-05.txt: section 7.1 on priority-value  interleaving
To: "James M. Polk" <jmpolk@cisco.com>
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Yes, I am happier.

Janet


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

This is a PRIVATE message. If you are not the intended recipient, please
delete without copying and kindly advise us by e-mail of the mistake in
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----------------------------------------------------------------------------------------




                                                                                                                             
                      "James M. Polk"                                                                                        
                      <jmpolk                  To:      Janet P Gunn/FED/CSC@CSC                                             
                      @cisco.com>              cc:      An.Nguyen@ncs.gov, Dean Willis <dean.willis@softarmor.com>,          
                                               Mpierce1@aol.com, oran@cisco.com, "Pete Babendreier" <pbabendr@cisco.com>,    
                      11/17/2004 04:47         rohan@ekabal.com, "Scott Morrison" <scmorris@cisco.com>, sip@ietf.org         
                      PM                       Subject: Re: [Sip] -RP-05.txt: section 7.1 on priority-value  interleaving    
                                                                                                                             
                                                                                                                             








Are you happier?


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented






_______________________________________________
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To: Orit Levin <oritl@microsoft.com>
Subject: Re: [Sip] RFC 3261: BUG or clarification is needed
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Orit Levin wrote:
> Guys,
> I would like to get a clarification on the properties of the Contact 
> header in requests and responses different from REGISTER.
> Is there a common understanding, that a Contact header can contain any 
> valid SIP URI that, especially when used outside the original dialog, 
> needs to be resolved according to RFC 3263 (i.e. including the SRV lookup) ?

Yes.

> I couldn't find any statement along these lines in any of the relevant RFCs.

Why would any special statement be required? The address in the contact 
gets included in some message, and then the rules for how to resolve it 
are spelled out. It doesn't matter where it came from.

> For example, in Redirect requests, we are reusing the Contact header in 
> order to point to an alternative destination which can have nothing in 
> common with the original domain.

BTW, in some contexts (such as 3xx) the contact addresses need not even 
be ones used by sip. You are allowed to redirect to an http uri if you like.

> Yet another example, the current GRUU draft is silent about the ability 
> of resolving GRUU using RFC 3263 procedures. >From the draft:
> "the GRUU MUST exhibit the following properties:
> 
>    o  The domain part of the URI is an IP address present on the public
> 
>       Internet, or, if it is a host name, exists in the global DNS and
> 
>       corresponds to an IP address present on the public Internet."
>  
>  
> For the interoperability sake, I believe that it is very important to 
> file this clarification against RFC 3261 and also point to RFC 3263 for 
> GRUU properties/location procedures.

I think this requirement is implicit.

	Paul

> Thanks,
> Orit.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From sip-bounces@ietf.org  Wed Nov 17 18:54:51 2004
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Subject: RE: [Sip] RFC 3261: BUG or clarification is needed
Date: Wed, 17 Nov 2004 15:29:08 -0800
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Thread-Topic: [Sip] RFC 3261: BUG or clarification is needed
Thread-Index: AcTM95vqoqQ8+1aeTnuBFhdMUB9FrgABKKXg
From: "Orit Levin" <oritl@microsoft.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Paul,
Thanks!
As long as it is accepted by everyone - it definitely works for me.
More inline.

Orit.

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Wednesday, November 17, 2004 2:49 PM
> To: Orit Levin
> Cc: sip@ietf.org
> Subject: Re: [Sip] RFC 3261: BUG or clarification is needed
>=20
>=20
>=20
> Orit Levin wrote:
> > Guys,
> > I would like to get a clarification on the properties of=20
> the Contact=20
> > header in requests and responses different from REGISTER.
> > Is there a common understanding, that a Contact header can=20
> contain any=20
> > valid SIP URI that, especially when used outside the=20
> original dialog,=20
> > needs to be resolved according to RFC 3263 (i.e. including=20
> the SRV lookup) ?
>=20
> Yes.
Great.
>=20
> > I couldn't find any statement along these lines in any of=20
> the relevant RFCs.
>=20
> Why would any special statement be required? The address in=20
> the contact gets included in some message, and then the rules=20
> for how to resolve it are spelled out. It doesn't matter=20
> where it came from.
Because I am not sure at all that many implementations actually do it
today and we need to encourage them to do so.
>=20
> > For example, in Redirect requests, we are reusing the=20
> Contact header=20
> > in order to point to an alternative destination which can=20
> have nothing=20
> > in common with the original domain.
>=20
> BTW, in some contexts (such as 3xx) the contact addresses=20
> need not even be ones used by sip. You are allowed to=20
> redirect to an http uri if you like.
Yes, I know :-)
>=20
> > Yet another example, the current GRUU draft is silent about the=20
> > ability of resolving GRUU using RFC 3263 procedures. >From=20
> the draft:
> > "the GRUU MUST exhibit the following properties:
> >=20
> >    o  The domain part of the URI is an IP address present on the=20
> > public
> >=20
> >       Internet, or, if it is a host name, exists in the=20
> global DNS and
> >=20
> >       corresponds to an IP address present on the public Internet."
> > =20
> > =20
> > For the interoperability sake, I believe that it is very=20
> important to=20
> > file this clarification against RFC 3261 and also point to RFC 3263=20
> > for GRUU properties/location procedures.
>=20
> I think this requirement is implicit.
Yes, most probably, but since this case is left out from the section
above - the reader can be mislead by the current text.
>=20
> 	Paul
>=20
> > Thanks,
> > Orit.
> >=20
> >=20
> >=20
> ----------------------------------------------------------------------
> > --
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
>=20
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Wed Nov 17 19:40:56 2004
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Jon said:

>Don't forget, for example, that SIPS does not apply after the domain
>indicated by the Request-URI of the request, and so on. SIPS is not a
>powerful mechanism. Until we all acknowledge that we were mistaken in our
>formulation in RFC3261, and that SIPS, when present, signifies unambiguously
>that TLS should be used end-to-end, I do not think SIPS can be said to solve
>any of our problems. Once we have the connect-reuse mechanism nailed down
>firmly, I think it will be possible to shift to that understanding of SIPS,
>if we want to.
>
>  
>

Even if SIPS was specified to be end-to-end, it would not resolve the 
issue of  a non-compliant (for whatever reason) proxy. This is one of 
the reasons I argued for the current text in 3261.  That which is on 
"the other side" of the responsible proxy might be SIP, it might be 
SIPS, or it might be H.323, there's no way to tell using just SIPS from 
"outside", and no amount of "requiring" otherwise in a specification 
will change this reality. In SIPS, each proxy is a transitive trust 
boundary, and that's just the way it is.

--
Dean



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From sip-bounces@ietf.org  Wed Nov 17 20:28:56 2004
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From: Dean Willis <dean.willis@softarmor.com>
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To: Mpierce1@aol.com, sip@ietf.org
Subject: Re: [Sip] Comments on Resource-priority-05
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Mpierce1@aol.com wrote:

> Based on the contents of previous versions of this draft, and comments 
> I and others have made, I believe that the following are essential for 
> the next version of this draft.
>
> The only normative information included per namespace should be:
>
> - name of the namespace
> - names of the priority levels
> - order of the priority values
> - default priority level (may be used if the receiving end does not 
> recognize a valid priority value)
>
> While the draft could detail various procedures which might be 
> applied, such as "modes", it should not associate any specific 
> procedure with any specific namespace.
>
> There should not be any other normative, mandatory behavior associated 
> with the use of the Resoure-priority header, such as specific 
> authorization procedures.
>
> In general, the draft can be made much shorter by deletion of 
> irrelevant material. For example, Section 7 describes procedures for 
> "move to the head of the line" scenarios for processing of messages in 
> SIP servers or UAs. It has been proposed in 
> draft-pierce-tsvwg-pref-treat-examples since April 2002 (and never 
> commented on) that there is no need to do such a thing. In fact, it is 
> not a good idea to intentionally change the order of messages..
>
This is a useful start. I believe we agree that there may be some 
content in -05 that is not required for publication.

Let, me, however, assert requirements for publishing a standards-track 
SIP extension. These are the kind of things that I expect any such 
document to meet, above the obvious "must have boilerplate, IANA 
considerations" and so on.

1) The document must contain, or contain references to, all information 
that a reasonable developer would need to build an interoperable 
implementation of the specification. Any referenced specifications must 
be publicly accessible, under the IETF's general guidelines for such. 
Examples might include documents from ITU, IEEE, W3C, ANSI, ETSI, and 
similar public specification authorities.

2) The SIP behavior of the extension and how it influences SIP dialog 
states, transaction results, etc. must be fully documented within the 
specification.

3) A system built using the specification can be analyzed and debugged 
using the non-SIP parts as a "black box". That is, the impact of the 
non-SIP functions must be apparent at the SIP level. If a request is 
rejected because of an extension, I want to know which extension, and by 
whom.

4) The threat model for attacks on this extension is clearly elucidated, 
and that the usage of available SIP mechanisms such as authentication, 
identity representation,SIPS, S/MIME, etc. is demonstrated in the text 
as adequate to mitigate those  threats or that additional mechanisms are 
detailed within the text as needed to mitigate those threats.

5) The requirements for any "extension of the extension" must be clearly 
spelled out.

Now, let's apply that to what has to be in this draft.

If the presence of header FOO in a request can cause that request or any 
other extant request, transaction, or dialog to be treated differently 
by ANY SIP element, including the terminating and originating UA, then 
the related SIP messaging must be explicitly detailed.

For example, if  a resource-priority header field in one request causes 
another dialog to be preempted, I want to know  what SIP messaging is 
sent to the participant(s) in the terminated dialog. (Example: BYE, with 
a Reason header, and which specific reason?) If the lack of a 
resource-priority header field (or a low-value priority)  in a request 
causes that request to be rejected, I want to know what SIP messaging is 
sent in the rejection. If a request is carrying a resource-priority 
header field or value that it is not authorized to use, and that causes 
the request to be rejected, diverted, or otherwise treated differently, 
I want to know what SIP messaging is sent back indicating the special 
handling.

I do not care WHAT the priority order is in a namespace that is 
externally documented, as long as that external document defines the 
priority ordering in a manner sufficient for a SIP implementor to build 
an interoperable implementation, or for an administrator to be able to 
determine whether observed system behavior is correct.

I do not care what the relative priority between namespaces in a 
particular node is, as long as there is some way to indicate which 
namespace held priority in any SIP messaging resulting from a priority 
comparison.

I do care if  attackers can, by simply inserting this new header, cause 
preemption or rejection of other calls. If this is a possible threat, I 
want it analyzed, and I want to have an explanation of how this relates 
to the security model and how applications of our security tools can 
keep this from happening.

I do care that fallbacks to "normal" behavior be documented, and that 
such issues as "What happens when a priority-aware element doesn't 
recognize a namespace" are properly explained.

I do care that the requirements for registering additional namespaces 
are clearly spelled out, and that it indicates that such namespaces must 
be defined by a referenceable specification, and that any registration 
of additional namespaces must include thorough documentation of any new 
SIP behavior induced by the new namespace. To me, that means a new 
namespace registration will need at least an informational track RFC, 
with expert review to make sure it meets these requirements. Obviously, 
it also requires that any namespaces registered IN THIS DOCUMENT be 
documented in adequate fashion so as to serve as guides for the 
documentation of any namespaces to be registered in the future.

My position is that we WILL NOT publish a standards-track SIP 
specification that simply defines a new header and defines a series of 
namespaces and possible values without explaining the effect on SIP of 
the presence of that a header, namespace, and values. We wouldn't even 
publish a non-standard P-header with this level of guidance, and we 
expect much more from a standards-track document.

--
Dean Willis
SIP co-chair







_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org  Wed Nov 17 21:14:39 2004
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Subject: RE: [Sip] Comments on Resource-priority-05
Date: Wed, 17 Nov 2004 20:10:13 -0600
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Thread-Topic: [Sip] Comments on Resource-priority-05
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From: "Dolly, Martin C, ALABS" <mdolly@att.com>
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Dean,

The SIP behavior needs to be defined in this draft. A SIP stack =
developer should be able to read this draft and understand the desired =
SIP behavior, without knowledge of the application processing.

Cheers,

Martin

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Dean Willis
Sent: Wednesday, November 17, 2004 8:17 PM
To: Mpierce1@aol.com; sip@ietf.org
Subject: Re: [Sip] Comments on Resource-priority-05


Mpierce1@aol.com wrote:

> Based on the contents of previous versions of this draft, and comments =

> I and others have made, I believe that the following are essential for =

> the next version of this draft.
>
> The only normative information included per namespace should be:
>
> - name of the namespace
> - names of the priority levels
> - order of the priority values
> - default priority level (may be used if the receiving end does not=20
> recognize a valid priority value)
>
> While the draft could detail various procedures which might be=20
> applied, such as "modes", it should not associate any specific=20
> procedure with any specific namespace.
>
> There should not be any other normative, mandatory behavior associated =

> with the use of the Resoure-priority header, such as specific=20
> authorization procedures.
>
> In general, the draft can be made much shorter by deletion of=20
> irrelevant material. For example, Section 7 describes procedures for=20
> "move to the head of the line" scenarios for processing of messages in =

> SIP servers or UAs. It has been proposed in=20
> draft-pierce-tsvwg-pref-treat-examples since April 2002 (and never=20
> commented on) that there is no need to do such a thing. In fact, it is =

> not a good idea to intentionally change the order of messages..
>
This is a useful start. I believe we agree that there may be some=20
content in -05 that is not required for publication.

Let, me, however, assert requirements for publishing a standards-track=20
SIP extension. These are the kind of things that I expect any such=20
document to meet, above the obvious "must have boilerplate, IANA=20
considerations" and so on.

1) The document must contain, or contain references to, all information=20
that a reasonable developer would need to build an interoperable=20
implementation of the specification. Any referenced specifications must=20
be publicly accessible, under the IETF's general guidelines for such.=20
Examples might include documents from ITU, IEEE, W3C, ANSI, ETSI, and=20
similar public specification authorities.

2) The SIP behavior of the extension and how it influences SIP dialog=20
states, transaction results, etc. must be fully documented within the=20
specification.

3) A system built using the specification can be analyzed and debugged=20
using the non-SIP parts as a "black box". That is, the impact of the=20
non-SIP functions must be apparent at the SIP level. If a request is=20
rejected because of an extension, I want to know which extension, and by =

whom.

4) The threat model for attacks on this extension is clearly elucidated, =

and that the usage of available SIP mechanisms such as authentication,=20
identity representation,SIPS, S/MIME, etc. is demonstrated in the text=20
as adequate to mitigate those  threats or that additional mechanisms are =

detailed within the text as needed to mitigate those threats.

5) The requirements for any "extension of the extension" must be clearly =

spelled out.

Now, let's apply that to what has to be in this draft.

If the presence of header FOO in a request can cause that request or any =

other extant request, transaction, or dialog to be treated differently=20
by ANY SIP element, including the terminating and originating UA, then=20
the related SIP messaging must be explicitly detailed.

For example, if  a resource-priority header field in one request causes=20
another dialog to be preempted, I want to know  what SIP messaging is=20
sent to the participant(s) in the terminated dialog. (Example: BYE, with =

a Reason header, and which specific reason?) If the lack of a=20
resource-priority header field (or a low-value priority)  in a request=20
causes that request to be rejected, I want to know what SIP messaging is =

sent in the rejection. If a request is carrying a resource-priority=20
header field or value that it is not authorized to use, and that causes=20
the request to be rejected, diverted, or otherwise treated differently,=20
I want to know what SIP messaging is sent back indicating the special=20
handling.

I do not care WHAT the priority order is in a namespace that is=20
externally documented, as long as that external document defines the=20
priority ordering in a manner sufficient for a SIP implementor to build=20
an interoperable implementation, or for an administrator to be able to=20
determine whether observed system behavior is correct.

I do not care what the relative priority between namespaces in a=20
particular node is, as long as there is some way to indicate which=20
namespace held priority in any SIP messaging resulting from a priority=20
comparison.

I do care if  attackers can, by simply inserting this new header, cause=20
preemption or rejection of other calls. If this is a possible threat, I=20
want it analyzed, and I want to have an explanation of how this relates=20
to the security model and how applications of our security tools can=20
keep this from happening.

I do care that fallbacks to "normal" behavior be documented, and that=20
such issues as "What happens when a priority-aware element doesn't=20
recognize a namespace" are properly explained.

I do care that the requirements for registering additional namespaces=20
are clearly spelled out, and that it indicates that such namespaces must =

be defined by a referenceable specification, and that any registration=20
of additional namespaces must include thorough documentation of any new=20
SIP behavior induced by the new namespace. To me, that means a new=20
namespace registration will need at least an informational track RFC,=20
with expert review to make sure it meets these requirements. Obviously,=20
it also requires that any namespaces registered IN THIS DOCUMENT be=20
documented in adequate fashion so as to serve as guides for the=20
documentation of any namespaces to be registered in the future.

My position is that we WILL NOT publish a standards-track SIP=20
specification that simply defines a new header and defines a series of=20
namespaces and possible values without explaining the effect on SIP of=20
the presence of that a header, namespace, and values. We wouldn't even=20
publish a non-standard P-header with this level of guidance, and we=20
expect much more from a standards-track document.

--
Dean Willis
SIP co-chair







_______________________________________________
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 sip-bounces@ietf.org  Wed Nov 17 21:34:36 2004
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To: sip@ietf.org, Jonathan Rosenberg <jdrosen@cisco.com>
Subject: Re: [Sip] WGLC for App Interact
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The last paragraph in section 4.3 discusses what I believe is an
approach where a client-remote interface is used to provide context and
a client-local interface (KPML) is used to receive input for
applications that where some prolonged dialog is needed.  The last
sentence leads me to this conclusion.  But unlike the beginning of the
section says, it implies that KPML is inappropriate for applications
that need to play a voice prompt and collect a response.  Clearly, the
approach described in this section is appropriate for a pre-paid
application, but this last paragraph seems to contradict this.  Some
text needs to be added that this paragraph is discussing using the model
for applications that require prolonged (say more than one
prompt-collect sequence) dialog with the user.

Section 9.1 has the following statement in the last paragraph.

   In the case of KPML, key events are passed to remote user
   interfaces by encoding them in RFC 2833 [17].

I can't figure out if this is a typo or statement that RFC 2833 is to be
used when a user presses a key on their keypad when application focus is
a question.  For one thing, it may be that the DTMF tones themselves
should be present in the media stream, and not just encoded using RFC
2833 for some scenarios.  (This would be negotiated using SDP.)  Or, if
this is a typo, KPML is a markup language that is not based on RFC 2833.
I recommend adding "for example" to the end of the sentence or
re-wording it to accurately describe how KPML works.

Best regards,
Bert


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From sip-bounces@ietf.org  Wed Nov 17 23:53:17 2004
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To: Steve Langstaff <steve.langstaff@citel.com>
Subject: Re: [Sip] PING/PONG
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There isn't a UAS involved here; these requests go from the client (UAC) 
to its proxy.

How the proxy indicates to the UAC that it is capable of processing 
these keepalives is a good question. My first thought is that this would 
be something in DNS; a new service in the NAPTR record for indicating 
the sip+stun combination.

-Jonathan R.

Steve Langstaff wrote:

> Should point 2 read "The user agent server indicates if it can respond to client refresh requests"?
> 
> --
> Steve Langstaff.
> 
> 
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
> Jonathan Rosenberg
> Sent: 17 November 2004 16:40
> To: Christian Stredicke
> Cc: sip@ietf.org
> Subject: Re: [Sip] PING/PONG
> 
> 
> 
> 
> Christian Stredicke wrote:
> 
> 
>>So my understanding is:
>>
>>1. We should use *only* STUN for refreshing bindings (either UDP or TCP,
>>what about TLS)
> 
> 
> Yes, TLS too. TLS runs ontop of TCP after all.
> 
> 
>>2. The user agent indicates if it can do it (no negotiation on
>>capabilities)
>>
>>3. Refreshing must be done from the client
>>
>>Can we agree on that?
> 
> 
> Yes.
> 
> Server originating refreshing is in the wrong direction. It is the 
> client utilizing the service; if it wishes to make use of that service, 
> it has to be responsible for refreshing the connection.
> 
> -Jonathan R.
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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To: Orit Levin <oritl@microsoft.com>
Subject: Re: [Sip] RFC 3261: BUG or clarification is needed
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I agree with Paul here.

-Jonathan R.

Orit Levin wrote:

> Paul,
> Thanks!
> As long as it is accepted by everyone - it definitely works for me.
> More inline.
> 
> Orit.
> 
> 
>>-----Original Message-----
>>From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>>Sent: Wednesday, November 17, 2004 2:49 PM
>>To: Orit Levin
>>Cc: sip@ietf.org
>>Subject: Re: [Sip] RFC 3261: BUG or clarification is needed
>>
>>
>>
>>Orit Levin wrote:
>>
>>>Guys,
>>>I would like to get a clarification on the properties of 
>>
>>the Contact 
>>
>>>header in requests and responses different from REGISTER.
>>>Is there a common understanding, that a Contact header can 
>>
>>contain any 
>>
>>>valid SIP URI that, especially when used outside the 
>>
>>original dialog, 
>>
>>>needs to be resolved according to RFC 3263 (i.e. including 
>>
>>the SRV lookup) ?
>>
>>Yes.
> 
> Great.
> 
>>>I couldn't find any statement along these lines in any of 
>>
>>the relevant RFCs.
>>
>>Why would any special statement be required? The address in 
>>the contact gets included in some message, and then the rules 
>>for how to resolve it are spelled out. It doesn't matter 
>>where it came from.
> 
> Because I am not sure at all that many implementations actually do it
> today and we need to encourage them to do so.
> 
>>>For example, in Redirect requests, we are reusing the 
>>
>>Contact header 
>>
>>>in order to point to an alternative destination which can 
>>
>>have nothing 
>>
>>>in common with the original domain.
>>
>>BTW, in some contexts (such as 3xx) the contact addresses 
>>need not even be ones used by sip. You are allowed to 
>>redirect to an http uri if you like.
> 
> Yes, I know :-)
> 
>>>Yet another example, the current GRUU draft is silent about the 
>>>ability of resolving GRUU using RFC 3263 procedures. >From 
>>
>>the draft:
>>
>>>"the GRUU MUST exhibit the following properties:
>>>
>>>   o  The domain part of the URI is an IP address present on the 
>>>public
>>>
>>>      Internet, or, if it is a host name, exists in the 
>>
>>global DNS and
>>
>>>      corresponds to an IP address present on the public Internet."
>>> 
>>> 
>>>For the interoperability sake, I believe that it is very 
>>
>>important to 
>>
>>>file this clarification against RFC 3261 and also point to RFC 3263 
>>>for GRUU properties/location procedures.
>>
>>I think this requirement is implicit.
> 
> Yes, most probably, but since this case is left out from the section
> above - the reader can be mislead by the current text.
> 
>>	Paul
>>
>>
>>>Thanks,
>>>Orit.
>>>
>>>
>>>
>>
>>----------------------------------------------------------------------
>>
>>>--
>>>
>>>_______________________________________________
>>>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
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Jonathan Rosenberg'" <jdrosen@cisco.com>
Subject: RE: third alternative (was RE: [Sip] RE: Identity after reinvite)
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> > 3) Change nothing in To/From. Requests in the backwards direction in a
> > dialog may go through the authentication service of the domain indicated
in
> > the host portion of the From header of the request. If that is not
possible,
> > do not supply request identity for such messages.
> 
> This is pretty close to what I had just suggested in a mail I wrote a 
> few minutes ago. I should have read the whole thread before responding I 
> suppose ;)

I'm glad we're on the same page (or at least the same chapter), but from the
remainder below, it sounds like we're on the same page for very different
reasons.

> The difference between this and what I had proposed is that I think a UA 
> should be *allowed* to change the From in its outbound request.

Well, we can't stop the callee UA from changing the From header field value
in its backwards-direction requests, yes, but I certainly think that the
caller UA should discard such requests. 

For me, this comes down to a very, very simple case: Alice sends a call to
Bob, and then in the middle of the call, she gets a BYE request from Edgar.
Does Alice accept the BYE or not? This is perhaps the simplest authorization
decision we could ask a client to make, and what I keep hearing people say
is that it's okay if the BYE comes from Edgar instead of Bob. I'm really not
sure how this could possibly be okay - this is unrelated to whether SIPS is
used or any other transitive security is used (see the end of this message);
this is the fundamental question of who should be authorized to terminate a
call.  If that set of authorized parties isn't identical to the first and
second parties in the call, then I think we're creating a non-deterministic
security environment. 

Are we really going to tell end users of SIP that it's okay if such a BYE
comes from an Edgar? What do I not understand here?

> > In cases where the user contacted
> > at chicago does not possess credentials to authenticate themselves to
> > biloxi, I think it would be fair to say that requests in the backwards
> > direction should not get an Identity header. 
> 
> I think that it will almost never have such credentials. If it did, this 
> wouldn't be retargeting.

Again, there's something fundamental I don't understand here, then. Alice
sends a request to Bob. For some reason, Bob's domain thinks it is
appropriate to forward the request to, say, Carol. Why does Bob's domain
think that? Because someone, with some relevant permissions, provisioned
Bob's domain with a directive that requests for Bob should be forwarded to
Carol. Of course, Carol personally does not possess Bob's credentials. If
Bob is at Carol's station, though, then Bob will possess Bob's credentials.
If the request arrives at Carol's station, and Bob fields the request, then
Bob will be capable of providing his credentials to any auth service as
necessary. Carol will not. 

If the user at Carol's station possesses Bob's credentials, then it is
reasonable for requests in the backwards direction to receive bear an
Identity header stating that this is from Bob; if not, then there shouldn't
be an Identity header in the request. This seems very intuitive to me; I'm
not sure why this is so radically implausible. I guess this begs the
question of why retargeting happens at all; if retargeting happens for some
other reason than what I describe above, then it most likely violates
RFC3261 as I understand it.

> > I've gotten a
> > lot of feedback (from Paul, Kumiko and others) that we should try to
weaken
> > the requirement for direct TLS, and I intend to add some text to the
next
> > version about some specific ways in which it is still acceptable to
provide
> > identity even if a direct TLS connection cannot be formed. Certain forms
of
> > weakness there could be beneficial to this approach.
> 
> I think that having non-direct connections is a reasonable thing; 
> generally my view has been that it should be possible to disaggregate a 
> monolithic proxy into components in the same network, each of which is a 
> SIP proxy performing parts of the overall processing in a domain. As 
> long as appropriate security and trust relationships are maintained 
> between these component proxies in a domain, I think equivalent security 
> is afforded.

I'm largely sympathetic to this argument, yes, and thus I agree that the
text should be amended. What I think should replace the strong requirement
for TLS is something to the effect that the UA must have an association with
an auth service that has a certain set of qualities - qualities that would
be satisfied by a direct TLS connection, but which hypothetically might be
derived through some other means.

> As a follow up to my previous note, I tihnk part of the difficulty here 
> is a difference in view about the problem being solved by asserted 
> identities in the reverse direction. In my view, the sips mechanism 
> provides an assurance ot the caller that the call got to the eventual 
> target as a result of authorized forwarding at each hop along the way. 

The sufficiency of SIPS for this purpose must depend on the practical
efficacy of the mechanism, not its theoretically efficacy. My own
reservations about the hops beyond the target administrative domain of a
SIPS request, coupled with Dean's valid concern that SIPS is not verifiable
by the endpoints, cast considerable doubt on the assurance that SIPS
provides. SIPS provides no assurance to the caller whatsoever - SIPS is
merely heaved over the wall at the network, and the caller can only hope
that the directive is obeyed. Even if the SIPS directive is strictly obeyed
by all intermediaries, given the laxity surrouding last-hop delivery,
potentials for eavesdropping arise. Eavesdropping alone is sufficient for an
attacker to construct a forged BYE request with a valid Identity header. 

That much said, I do think there would be some value in having an identity
mechanism that prevents trivial initial-request impersonation without
addressing eavesdropping. But if we so reduce the scope, that would argue
for a fourth alternative: something like "in the context of a dialog,
request identity is only used for the dialog-forming request". Identity
isn't adding any appreciable value to any subsequent requests.

> Thus, the identity of requests in the reverse direction is not used to 
> *authorize* whether or not the request should be processed. Rather, it 
> serves merely as a reliable identification mechanism, as authorization 
> is provided by sips.

If you trust SIPS far enough to authorize backwards-direction requests and
responses reliably, then surely Identity would be superfluous. What
additional properties would Identity convey beyond those that can be assumed
from SIPS?

This may be the crux of our disagreement. I don't believe that SIPS is
strong enough to substantiate an authorization decision on the part of the
caller. Moreover, I'd like to believe that sip-identity can be used in the
absence of SIPS; that is, I don't think that every potential callee must
publish a SIPS URI in order for a caller to make reasonable authorization
decisions the source of requests in the backwards direction in a dialog. In
the long term, when we want to provide a mandatory-to-implement mechanism
for ascertaining identity, I don't think this can require SIPS as it is
written now, anyway.

The authorization decision made by the caller should be based on a
cryptographic end-to-end assertion stating that the intended domain vouches
for the callee; it should not merely be inferred from a speculation that a
directive like SIPS was universally obeyed by an unfathomable set of unknown
intermediaries.

In summary, this is the ultimate distinction between the "trusted networks"
of RFC3325 and the aptly-named "long term cryptographic identity solution"
we envisioned years ago. If something like SIPS is genuinely sufficient,
then I'd argue that P-Asserted-Identity provided by the first-hop proxy in
the request path would be all we'd need, regardless of whether we're talking
about dialog-forming requests or requests in the backwards direction in a
dialog.

Jon Peterson
NeuStar, Inc.

> -JOnathan R.
> 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ 
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From: "vikram bhuskute" <vikramb@aftek.com>
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Subject: [Sip] What if 200 Ok  response on Hold-Invite is late
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This is a multi-part message in MIME format.

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Hello,
              If dialogue is established between user A and B. and if C =
calls
 then the call between A & B should be put on hold( If multiple active =
call is not suppported)

 Now should A wait for a 200 OK response(for hold-reinvite) from B(whcih =
could be late because=20
 of n/w problem)  or It can assume that Call is on hold and he can =
staighaway accept the other call ?
 =20
 I think as soon as I send Hold on invite , I can assume that call is =
put on Hold and i can do
 furhter processing based on this assumption ? Am i right ?

regards

Vikram Bhuskute
------=_NextPart_000_01CC_01C4CD72.0C3E45B0
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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1476" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hello,</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
If dialogue is established between user A and B.&nbsp;and if&nbsp;C=20
calls</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;then the call between A &amp; B =
should be put=20
on hold( If multiple active call is&nbsp;not suppported)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;Now should A wait for a 200 OK =
response(for=20
hold-reinvite)&nbsp;from B(whcih could be late because </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;of n/w problem)&nbsp; =
</FONT><FONT face=3DArial=20
size=3D2>or It can assume that Call is on hold</FONT><FONT face=3DArial=20
size=3D2>&nbsp;and he can staighaway accept the other call =
?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;I think as soon as I send Hold on =
invite , I=20
can assume that call is put on Hold and i can do</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;furhter processing based on this =
assumption ?=20
Am i right ?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>regards</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Vikram =
Bhuskute</FONT></DIV></BODY></HTML>

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From: "vikram bhuskute" <vikramb@aftek.com>
To: <sip@ietf.org>
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Subject: [Sip] What if 200 Ok  response on Hold-Invite is late
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Hello,
              If dialogue is established between user A and B. and if C =
calls
 then the call between A & B should be put on hold( If multiple active =
call is not suppported)
=20
 Now should A wait for a 200 OK response(for hold-reinvite) from B(whcih =
could be late because=20
 of n/w problem)  or It can assume that Call is on hold and he can =
staighaway accept the other call ?
 =20
 I think as soon as I send Hold on invite , I can assume that call is =
put on Hold and i can do
 furhter processing based on this assumption ? Am i right ?

regards

Vikram Bhuskute
------=_NextPart_000_0197_01C4CD71.AAF04C90
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	charset="iso-8859-1"
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<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1476" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hello,</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
If dialogue is established between user A and B.&nbsp;and if&nbsp;C=20
calls</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;then the call between A &amp; B =
should be put=20
on hold( If multiple active call is&nbsp;not suppported)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;Now should A wait for a 200 OK =
response(for=20
hold-reinvite)&nbsp;from B(whcih could be late because </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;of n/w problem)&nbsp; =
</FONT><FONT face=3DArial=20
size=3D2>or It can assume that Call is on hold</FONT><FONT face=3DArial=20
size=3D2>&nbsp;and he can staighaway accept the other call =
?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;I think as soon as I send Hold on =
invite , I=20
can assume that call is put on Hold and i can do</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;furhter processing based on this =
assumption ?=20
Am i right ?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>regards</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Vikram =
Bhuskute</FONT></DIV></BODY></HTML>

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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
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From: "vikram bhuskute" <vikramb@aftek.com>
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Subject: [Sip] What if 200 Ok  response on Hold-Invite is late
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Hello,
              If dialogue is established between user A and B. and if C =
calls
 then the call between A & B should be put on hold( If multiple active =
call is not suppported)
=20
 Now should A wait for a 200 OK response(for hold-reinvite) from B(whcih =
could be late because=20
 of n/w problem)  or It can assume that Call is on hold and he can =
staighaway accept the other call ?
 =20
 I think as soon as I send Hold on invite , I can assume that call is =
put on Hold and i can do
 furhter processing based on this assumption ? Am i right ?

regards

Vikram Bhuskute
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<HTML><HEAD>
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hello,</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
If dialogue is established between user A and B.&nbsp;and if&nbsp;C=20
calls</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;then the call between A &amp; B =
should be put=20
on hold( If multiple active call is&nbsp;not suppported)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;Now should A wait for a 200 OK =
response(for=20
hold-reinvite)&nbsp;from B(whcih could be late because </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;of n/w problem)&nbsp; =
</FONT><FONT face=3DArial=20
size=3D2>or It can assume that Call is on hold</FONT><FONT face=3DArial=20
size=3D2>&nbsp;and he can staighaway accept the other call =
?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;I think as soon as I send Hold on =
invite , I=20
can assume that call is put on Hold and i can do</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;furhter processing based on this =
assumption ?=20
Am i right ?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>regards</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Vikram =
Bhuskute</FONT></DIV></BODY></HTML>

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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
--===============1640857693==--




From sip-bounces@ietf.org  Thu Nov 18 06:39:16 2004
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Date: Thu, 18 Nov 2004 06:31:15 EST
Subject: Re: [Sip] -RP-05.txt: section 7.1 on priority-value interleaving
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In a message dated 11/17/2004 11:26:59 PM W. Europe Standard Time, 
jmpolk@cisco.com writes:


> Section 7.1 is merely to give examples to readers that show the order 
> within a namespace must be maintained regardless of how many other 
> namespaces exist (that are supported) by that SIP element.
> 
> The section also provides examples of how the orders can be in multiple 
> combinations of groupings, as long as the overall order within each 
> namespace is not changed.
> 
> 


The point in the first paragraph above is important, and needs to be made so 
by not including it in a section titled "multiple namespaces", since, as you 
said, it is true "regardless of how many other namespaces exist". Somewhere 
else in the document there should be a clear statement that the order of priority 
values within each individual namespace must be maintained. In fact, I 
believe that the actual requirement should be stated as follows:

-----------------------
The order of priority values MUST be maintained. That is, if priority value 
"A" is specified as being higher than priority value "B" in the IANA 
registration, then an entity MUST NOT give traffic marked with value "A" a lower level 
of preferential treatment than traffic marked with priority value "B". An 
entity MUST NOT reorder the priority values.
-----------------------

It needs to be stated this way, since it is permissible for an entity to give 
two levels the same treatment. For example, in the DSN, Flash and Flash 
Override may get the same treatment for packet transport at core routers while 
other entities may treat them different (endpoints or edge routers).

The above requirement is too important to "hide" in a section about multiple 
namespaces.

The second paragraph above, and the text that shows relationships between 
different namespaces remains confusing since there is a clear agreement that 
there is no relationship between namespaces. This text clearly implies that there 
is a relationship. If someything needs to be said to explain this to people, 
then let's work on text to address the problem. I believe that the current text 
will just further confuse others (not me). I'm not confused by it because I 
know the truth. Others won't.

Mike Pierce




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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/17/2004 11:26:59 PM W. Europe Standard Time, jmpolk@cisco.com write=
s:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Section 7.1 is merely to gi=
ve examples to readers that show the order=20
<BR>within a namespace must be maintained regardless of how many other=20
<BR>namespaces exist (that are supported) by that SIP element.
<BR>
<BR>The section also provides examples of how the orders can be in multiple=20
<BR>combinations of groupings, as long as the overall order within each=20
<BR>namespace is not changed.
<BR>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>The point in the first paragraph above is important, and needs to be mad=
e so by not including it in a section titled "multiple namespaces", since, a=
s you said, it is true "regardless of how many other namespaces exist". Some=
where else in the document there should be a clear statement that the order=20=
of priority values within each individual namespace must be maintained. In f=
act, I believe that the actual requirement should be stated as follows:
<BR>
<BR>-----------------------
<BR>The order of priority values MUST be maintained. That is, if priority va=
lue "A" is specified as being higher than priority value "B" in the IANA reg=
istration, then an entity MUST NOT give traffic marked with value "A" a lowe=
r level of preferential treatment than traffic marked with priority value "B=
". An entity MUST NOT reorder the priority values.
<BR>-----------------------
<BR>
<BR>It needs to be stated this way, since it is permissible for an entity to=
 give two levels the same treatment. For example, in the DSN, Flash and Flas=
h Override may get the same treatment for packet transport at core routers w=
hile other entities may treat them different (endpoints or edge routers).
<BR>
<BR>The above requirement is too important to "hide" in a section about mult=
iple namespaces.
<BR>
<BR>The second paragraph above, and the text that shows relationships betwee=
n different namespaces remains confusing since there is a clear agreement th=
at there is no relationship between namespaces. This text clearly implies th=
at there is a relationship. If someything needs to be said to explain this t=
o people, then let's work on text to address the problem. I believe that the=
 current text will just further confuse others (not me). I'm not confused by=
 it because I know the truth. Others won't.
<BR>
<BR>Mike Pierce
<BR>
<BR>
<BR></FONT></HTML>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/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
--===============0571908988==--



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Subject: RE: [Sip] PING/PONG
Date: Thu, 18 Nov 2004 11:14:42 -0000
Message-ID: <CD9775120D600F43B9C50329395E9DB64F321F@ivor.citel.com>
Thread-Topic: [Sip] PING/PONG
Thread-Index: AcTNKYWvD4vdewpKQSyep4WKbocUPgAM9r8w
From: "Steve Langstaff" <steve.langstaff@citel.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>
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Sorry, I didn't realise the scope of this discussion was client-to-proxy =
- that's where my confusion lay.

A further question from my (simplistic) viewpoint... If this issue =
is/could be generic to all STUNned traffic (not just sip+stun), would it =
not be better to push the problem down into the STUN 'layer' and let the =
STUN client and server sort out keeping the NAT bindings alive? I'm =
guessing that the SIP client already 'knows' it has to use STUN to reach =
the proxy, and so could ask it's STUN client to keep the binding(s) =
alive as appropriate.


--
Steve Langstaff.


-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
Sent: 18 November 2004 04:46
To: Steve Langstaff
Cc: Christian Stredicke; sip@ietf.org
Subject: Re: [Sip] PING/PONG


There isn't a UAS involved here; these requests go from the client (UAC) =

to its proxy.

How the proxy indicates to the UAC that it is capable of processing=20
these keepalives is a good question. My first thought is that this would =

be something in DNS; a new service in the NAPTR record for indicating=20
the sip+stun combination.

-Jonathan R.

Steve Langstaff wrote:

> Should point 2 read "The user agent server indicates if it can respond =
to client refresh requests"?
>=20
> --
> Steve Langstaff.
>=20
>=20
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
> Jonathan Rosenberg
> Sent: 17 November 2004 16:40
> To: Christian Stredicke
> Cc: sip@ietf.org
> Subject: Re: [Sip] PING/PONG
>=20
>=20
>=20
>=20
> Christian Stredicke wrote:
>=20
>=20
>>So my understanding is:
>>
>>1. We should use *only* STUN for refreshing bindings (either UDP or =
TCP,
>>what about TLS)
>=20
>=20
> Yes, TLS too. TLS runs ontop of TCP after all.
>=20
>=20
>>2. The user agent indicates if it can do it (no negotiation on
>>capabilities)
>>
>>3. Refreshing must be done from the client
>>
>>Can we agree on that?
>=20
>=20
> Yes.
>=20
> Server originating refreshing is in the wrong direction. It is the=20
> client utilizing the service; if it wishes to make use of that =
service,=20
> it has to be responsible for refreshing the connection.
>=20
> -Jonathan R.
>=20

--=20
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Subject: Re: [Sip] Comments on Resource-priority-05
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Those sound like good guidelines to me.

Janet


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                      Dean Willis                                                                                            
                      <dean.willis             To:      Mpierce1@aol.com, sip@ietf.org                                       
                      @softarmor.com>          cc:                                                                           
                      Sent by:                 Subject: Re: [Sip] Comments on Resource-priority-05                           
                      sip-bounces                                                                                            
                                                                                                                             
                                                                                                                             
                      11/17/2004 08:17                                                                                       
                      PM                                                                                                     
                                                                                                                             
                                                                                                                             




Mpierce1@aol.com wrote:

> Based on the contents of previous versions of this draft, and comments
> I and others have made, I believe that the following are essential for
> the next version of this draft.
>
> The only normative information included per namespace should be:
>
> - name of the namespace
> - names of the priority levels
> - order of the priority values
> - default priority level (may be used if the receiving end does not
> recognize a valid priority value)
>
> While the draft could detail various procedures which might be
> applied, such as "modes", it should not associate any specific
> procedure with any specific namespace.
>
> There should not be any other normative, mandatory behavior associated
> with the use of the Resoure-priority header, such as specific
> authorization procedures.
>
> In general, the draft can be made much shorter by deletion of
> irrelevant material. For example, Section 7 describes procedures for
> "move to the head of the line" scenarios for processing of messages in
> SIP servers or UAs. It has been proposed in
> draft-pierce-tsvwg-pref-treat-examples since April 2002 (and never
> commented on) that there is no need to do such a thing. In fact, it is
> not a good idea to intentionally change the order of messages..
>
This is a useful start. I believe we agree that there may be some
content in -05 that is not required for publication.

Let, me, however, assert requirements for publishing a standards-track
SIP extension. These are the kind of things that I expect any such
document to meet, above the obvious "must have boilerplate, IANA
considerations" and so on.

1) The document must contain, or contain references to, all information
that a reasonable developer would need to build an interoperable
implementation of the specification. Any referenced specifications must
be publicly accessible, under the IETF's general guidelines for such.
Examples might include documents from ITU, IEEE, W3C, ANSI, ETSI, and
similar public specification authorities.

2) The SIP behavior of the extension and how it influences SIP dialog
states, transaction results, etc. must be fully documented within the
specification.

3) A system built using the specification can be analyzed and debugged
using the non-SIP parts as a "black box". That is, the impact of the
non-SIP functions must be apparent at the SIP level. If a request is
rejected because of an extension, I want to know which extension, and by
whom.

4) The threat model for attacks on this extension is clearly elucidated,
and that the usage of available SIP mechanisms such as authentication,
identity representation,SIPS, S/MIME, etc. is demonstrated in the text
as adequate to mitigate those  threats or that additional mechanisms are
detailed within the text as needed to mitigate those threats.

5) The requirements for any "extension of the extension" must be clearly
spelled out.

Now, let's apply that to what has to be in this draft.

If the presence of header FOO in a request can cause that request or any
other extant request, transaction, or dialog to be treated differently
by ANY SIP element, including the terminating and originating UA, then
the related SIP messaging must be explicitly detailed.

For example, if  a resource-priority header field in one request causes
another dialog to be preempted, I want to know  what SIP messaging is
sent to the participant(s) in the terminated dialog. (Example: BYE, with
a Reason header, and which specific reason?) If the lack of a
resource-priority header field (or a low-value priority)  in a request
causes that request to be rejected, I want to know what SIP messaging is
sent in the rejection. If a request is carrying a resource-priority
header field or value that it is not authorized to use, and that causes
the request to be rejected, diverted, or otherwise treated differently,
I want to know what SIP messaging is sent back indicating the special
handling.

I do not care WHAT the priority order is in a namespace that is
externally documented, as long as that external document defines the
priority ordering in a manner sufficient for a SIP implementor to build
an interoperable implementation, or for an administrator to be able to
determine whether observed system behavior is correct.

I do not care what the relative priority between namespaces in a
particular node is, as long as there is some way to indicate which
namespace held priority in any SIP messaging resulting from a priority
comparison.

I do care if  attackers can, by simply inserting this new header, cause
preemption or rejection of other calls. If this is a possible threat, I
want it analyzed, and I want to have an explanation of how this relates
to the security model and how applications of our security tools can
keep this from happening.

I do care that fallbacks to "normal" behavior be documented, and that
such issues as "What happens when a priority-aware element doesn't
recognize a namespace" are properly explained.

I do care that the requirements for registering additional namespaces
are clearly spelled out, and that it indicates that such namespaces must
be defined by a referenceable specification, and that any registration
of additional namespaces must include thorough documentation of any new
SIP behavior induced by the new namespace. To me, that means a new
namespace registration will need at least an informational track RFC,
with expert review to make sure it meets these requirements. Obviously,
it also requires that any namespaces registered IN THIS DOCUMENT be
documented in adequate fashion so as to serve as guides for the
documentation of any namespaces to be registered in the future.

My position is that we WILL NOT publish a standards-track SIP
specification that simply defines a new header and defines a series of
namespaces and possible values without explaining the effect on SIP of
the presence of that a header, namespace, and values. We wouldn't even
publish a non-standard P-header with this level of guidance, and we
expect much more from a standards-track document.

--
Dean Willis
SIP co-chair







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From: "Thomas Gal" <ThomasGal@LumenVox.com>
To: "'Patrick Mourot'" <patrick.mourot@online.fr>
Subject: RE: [Sip] meeting minutes from SIP session 2 at IETF 61
Date: Thu, 18 Nov 2004 09:18:43 -0800
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Now......sorry I named the file .doc.html for some reason.

Tom

> -----Original Message-----
> From: Patrick Mourot [mailto:patrick.mourot@online.fr] 
> Sent: Thursday, November 18, 2004 8:21 AM
> To: Thomas Gal
> Subject: Re: [Sip] meeting minutes from SIP session 2 at IETF 61
> 
> Thomas,
> 
> Do you know when ?
> 
> Best Regards,
> 
> Patrick
> 
> Thomas Gal said the following on 11/11/04 11:40 pm:
> 
> >The minutes will be at 
> http://natural.salk.edu/~tgal/ietf61/index.html shortly.
> >
> >Tom
> >
> >
> >
> >_______________________________________________
> >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: Mpierce1@aol.com, jgunn6@csc.com
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] -RP-05.txt: section 7.1 on priority-value
  interleaving
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At 06:31 AM 11/18/2004 -0500, Mpierce1@aol.com wrote:

>since it is permissible for an entity to give two levels the same 
>treatment. For example, in the DSN, Flash and Flash Override may get the 
>same treatment for packet transport at core routers while other entities 
>may treat them different (endpoints or edge routers).

this is a good point, I will adjust the text to reflect this


>The above requirement is too important to "hide" in a section about 
>multiple namespaces.
>
>The second paragraph above, and the text that shows relationships between 
>different namespaces remains confusing since there is a clear agreement 
>that there is no relationship between namespaces.

yet some remained confused until they saw examples of various combinations 
that didn't violate the rule.

>This text clearly implies that there is a relationship. If someything 
>needs to be said to explain this to people, then let's work on text to 
>address the problem. I believe that the current text will just further 
>confuse others (not me). I'm not confused by it because I know the truth. 
>Others won't.
>
>Mike Pierce
>


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Subject: Re: third alternative (was RE: [Sip] RE: Identity after reinvite)
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Trimming the discussion...

Peterson, Jon wrote:

> I'm glad we're on the same page (or at least the same chapter), but from the
> remainder below, it sounds like we're on the same page for very different
> reasons.
> 
>>The difference between this and what I had proposed is that I think a UA 
>>should be *allowed* to change the From in its outbound request.
> 
> Well, we can't stop the callee UA from changing the From header field value
> in its backwards-direction requests, yes, but I certainly think that the
> caller UA should discard such requests. 
> 
> For me, this comes down to a very, very simple case: Alice sends a call to
> Bob, and then in the middle of the call, she gets a BYE request from Edgar.
> Does Alice accept the BYE or not? This is perhaps the simplest authorization
> decision we could ask a client to make, and what I keep hearing people say
> is that it's okay if the BYE comes from Edgar instead of Bob. I'm really not
> sure how this could possibly be okay - this is unrelated to whether SIPS is
> used or any other transitive security is used (see the end of this message);
> this is the fundamental question of who should be authorized to terminate a
> call.  If that set of authorized parties isn't identical to the first and
> second parties in the call, then I think we're creating a non-deterministic
> security environment. 
> 
> Are we really going to tell end users of SIP that it's okay if such a BYE
> comes from an Edgar? What do I not understand here?

I will admit that this scenario seems pretty broken.

However, if the call to bob reached edgar, and he has done as you say, 
it is futile for alice to reject the BYE. It is an indication that the 
other party has hung up, and refusing it will not change that.

If edgar is just a MITM trying to disrupt a valid call between alice and 
bob, then maybe rejecting the BYE would accomplish something. But things 
will still be messed up - for instance CSeq values will be wrong. And if 
edgar was in a position to do this with his own name, he could just as 
easily have done it with bob's - aside from the issue of signing. Maybe 
it makes a little sense in that edgar may be able to get the BYE signed 
with his own identity, though that will be tricky.

I do agree it would be better if we could come up with some explicit way 
to inform alice that the request has been delegated from bob to edgar. 
Maybe it is just a matter of choosing among existing mechanisms, rather 
than inventing something new.

>>>In cases where the user contacted
>>>at chicago does not possess credentials to authenticate themselves to
>>>biloxi, I think it would be fair to say that requests in the backwards
>>>direction should not get an Identity header. 
>>
>>I think that it will almost never have such credentials. If it did, this 
>>wouldn't be retargeting.
> 
> Again, there's something fundamental I don't understand here, then. Alice
> sends a request to Bob. For some reason, Bob's domain thinks it is
> appropriate to forward the request to, say, Carol. Why does Bob's domain
> think that? Because someone, with some relevant permissions, provisioned
> Bob's domain with a directive that requests for Bob should be forwarded to
> Carol. Of course, Carol personally does not possess Bob's credentials. If
> Bob is at Carol's station, though, then Bob will possess Bob's credentials.
> If the request arrives at Carol's station, and Bob fields the request, then
> Bob will be capable of providing his credentials to any auth service as
> necessary. Carol will not. 

That of course assumes that Carol's phone is capable of prompting for 
credentials, such as asking for Bob's password. Maybe someday most 
phones will have that capability, but not today.

And what if bob just forwarded to carol because he is away?

> If the user at Carol's station possesses Bob's credentials, then it is
> reasonable for requests in the backwards direction to receive bear an
> Identity header stating that this is from Bob; if not, then there shouldn't
> be an Identity header in the request. This seems very intuitive to me; I'm
> not sure why this is so radically implausible. I guess this begs the
> question of why retargeting happens at all; if retargeting happens for some
> other reason than what I describe above, then it most likely violates
> RFC3261 as I understand it.

In most scenarios I am aware of today, phone is either configured with 
the credentials, or in a hotelling situation the phone requests 
credentials to REGISTER somewhere. The hotelling situation is certainly 
better from a security viewpoint, and it doesn't look like retargetting 
at all to the caller.

But hotelling isn't always going to be possible. Perhaps bob is visiting 
carol, and can configure his own AOR to retarget to carol, but can't 
hotel on carol's phone and can't provide credentials for an incoming 
call that was retargetted from bob to carol.

What does this violate in 3261?

One solution to this would be for for the biloxi.com server to act as a 
b2bua, relaying the call to carol. Maybe that is a *model* to consider, 
with hopes of finding a better implementation. That is also roughly the 
model when a gateway is being used. We already know that in the gateway 
case the connected party on the far side of the gateway can change, and 
there are desires to convey that info.

	Paul


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Subject: Re: [Sip] -RP-05.txt: section 7.1 on priority-value  interleaving
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That would help with one of my concerns too- that for some resources we
need to prioritize based on the value and for other resources we only care
about the namespace.


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                      "James M. Polk"                                                                                        
                      <jmpolk                  To:      Mpierce1@aol.com, Janet P Gunn/FED/CSC@CSC                           
                      @cisco.com>              cc:      An.Nguyen@ncs.gov, rohan@ekabal.com, scmorris@cisco.com,             
                      Sent by:                 dean.willis@softarmor.com, sip@ietf.org, oran@cisco.com, pbabendr@cisco.com   
                      sip-bounces              Subject: Re: [Sip] -RP-05.txt: section 7.1 on priority-value  interleaving    
                                                                                                                             
                                                                                                                             
                      11/18/2004 01:34                                                                                       
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At 06:31 AM 11/18/2004 -0500, Mpierce1@aol.com wrote:

>since it is permissible for an entity to give two levels the same
>treatment. For example, in the DSN, Flash and Flash Override may get the
>same treatment for packet transport at core routers while other entities
>may treat them different (endpoints or edge routers).

this is a good point, I will adjust the text to reflect this


>The above requirement is too important to "hide" in a section about
>multiple namespaces.
>
>The second paragraph above, and the text that shows relationships between
>different namespaces remains confusing since there is a clear agreement
>that there is no relationship between namespaces.

yet some remained confused until they saw examples of various combinations
that didn't violate the rule.

>This text clearly implies that there is a relationship. If someything
>needs to be said to explain this to people, then let's work on text to
>address the problem. I believe that the current text will just further
>confuse others (not me). I'm not confused by it because I know the truth.
>Others won't.
>
>Mike Pierce
>


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Use sip-implementors@cs.columbia.edu for questions on current sip
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
Subject: RE: third alternative (was RE: [Sip] RE: Identity after reinvite)
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> > Are we really going to tell end users of SIP that it's okay if such a
BYE
> > comes from an Edgar? What do I not understand here?
> 
> I will admit that this scenario seems pretty broken.
> 
> However, if the call to bob reached edgar, and he has done as you say, 
> it is futile for alice to reject the BYE. It is an indication that the 
> other party has hung up, and refusing it will not change that.

What I really meant here is that it makes no sense to provide Identity for a
request in the backwards direction from Edgar if he is not the intended
target of the call. Identity is useful only if it is going to factor into an
authorization decision regarding impersonation. If literally any identity in
the request must be accepted by Alice, rather than the unique identity of
Bob, then Identity is a null op and we should stop trying to engineer for
this case. This is the "fourth alternative" I alluded to in my last mail -
just say that request identity is not meaningful for any requests in a
dialog other than the dialog-forming request. I do think we can do better
than that, though, without creating serious problems for ourselves. If we
choose the fourth alternative, we will be allowing a certain security hole
to remain open, one that I believe we can close without too much trouble.

> If edgar is just a MITM trying to disrupt a valid call between alice and 
> bob, then maybe rejecting the BYE would accomplish something. But things 
> will still be messed up - for instance CSeq values will be wrong. 

By "reject" here I mean something more along the lines of "silently
discard". A spoofed request should not alter the dialog state in the
recipient.

> And if 
> edgar was in a position to do this with his own name, he could just as 
> easily have done it with bob's - aside from the issue of signing. Maybe
> it makes a little sense in that edgar may be able to get the BYE signed 
> with his own identity, though that will be tricky.

This is the rub. If Alice will accept only BYE requests from Bob, then
Identity adds value - Edgar cannot impersonate Bob because of Identity.
Identity only addresses threats of impersonation.

> I do agree it would be better if we could come up with some explicit way 
> to inform alice that the request has been delegated from bob to edgar. 
> Maybe it is just a matter of choosing among existing mechanisms, rather 
> than inventing something new. 

Totally agreed. sip-identity-02 suggested, albeit without sufficient
motivation and argument, that the requirements for such a mechanism are
satisfied by simple SIP redirection. I'm trying to keep that whole issue out
of the scope of sip-identity-03/04, and one way to do that, I think, is to
go along with something roughly like my "third alternative" proposal, which
basically works when it can, and is forward-compatible with a future
redirection-based responde identity model in which it would work all the
time.

> > Alice
> > sends a request to Bob. For some reason, Bob's domain thinks it is
> > appropriate to forward the request to, say, Carol. Why does Bob's domain
> > think that? Because someone, with some relevant permissions, provisioned
> > Bob's domain with a directive that requests for Bob should be forwarded
to
> > Carol. Of course, Carol personally does not possess Bob's credentials.
If
> > Bob is at Carol's station, though, then Bob will possess Bob's
credentials.
> > If the request arrives at Carol's station, and Bob fields the request,
then
> > Bob will be capable of providing his credentials to any auth service as
> > necessary. Carol will not. 
> 
> That of course assumes that Carol's phone is capable of prompting for 
> credentials, such as asking for Bob's password. Maybe someday most 
> phones will have that capability, but not today.
> 
> And what if bob just forwarded to carol because he is away?

In both of these cases, clearly, Identity should not be provided in the
backwards direction. If the responder isn't actually Bob, then it shouldn't
claim to be. But claiming to be someone else (Carol, or Edgar, or what have
you) doesn't help Alice to make any meaningful authorization decisions, I
think, unless Bob's domain has some way to communicate to Alice that the
request has been forwarded to that particular someone.

This isn't to say that there might not be some counterexamples to that last
principle - I mean, it's possible that the identity of the connected party
will make it clear how/why the request was retargeted, and that this will
seem reasonable from an authorization perspective to a caller with some
pre-knowledge of the callee - but the existence of such counterexamples does
not, I think, justify leaving this security hole wide open for those cases
where we cannot rely on human pre-association and presumption to fill in the
gaps of our protocol's authorization story. That is a non-deterministic
security environment.

Of course, if Bob's domain redirected rather than retargeting, the
underlying problem would be solved.

> 
> > If the user at Carol's station possesses Bob's credentials, then it is
> > reasonable for requests in the backwards direction to receive bear an
> > Identity header stating that this is from Bob; if not, then there
shouldn't
> > be an Identity header in the request. This seems very intuitive to me;
I'm
> > not sure why this is so radically implausible. I guess this begs the
> > question of why retargeting happens at all; if retargeting happens for
some
> > other reason than what I describe above, then it most likely violates
> > RFC3261 as I understand it.
> 
> In most scenarios I am aware of today, phone is either configured with 
> the credentials, or in a hotelling situation the phone requests 
> credentials to REGISTER somewhere. The hotelling situation is certainly 
> better from a security viewpoint, and it doesn't look like retargetting 
> at all to the caller.
> 
> But hotelling isn't always going to be possible. Perhaps bob is visiting 
> carol, and can configure his own AOR to retarget to carol, but can't 
> hotel on carol's phone and can't provide credentials for an incoming 
> call that was retargetted from bob to carol.
> 
> What does this violate in 3261?

Your two paragraphs above violate nothing about RFC3261. I was referring to
the motivations for legitimate retargeting (that is, that Bob's domain is
the only one that can retarget a request for Bob, and that Bob has an
association with his domain that allows him to control his contact list).

While I don't disagree with you that the case described above is possible,
I'm just saying that we shouldn't provide an Identity header in that case.

> One solution to this would be for for the biloxi.com server to act as a 
> b2bua, relaying the call to carol. Maybe that is a *model* to consider, 
> with hopes of finding a better implementation. That is also roughly the 
> model when a gateway is being used. We already know that in the gateway 
> case the connected party on the far side of the gateway can change, and 
> there are desires to convey that info.
> 

(weep)

Maybe we could at least consider redirection before we start suggesting that
this is a B2BUA problem.

I really think that the problem of mid-call changes in connected-party is a
very different case, and much more specialized than what we're concerned
with here. We're talking here about assuring the identity of the first and
second parties to a dialog at the time of dialog establishment. I'd like to
think that the ultimate solutions to the mid-call 'identity update' sort of
problems are more along the transfer/replaces line of thinking.

Jon Peterson
NeuStar, Inc.

> 	Paul
> 

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Subject: Re: third alternative (was RE: [Sip] RE: Identity after reinvite)
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Peterson, Jon wrote:
>>>Are we really going to tell end users of SIP that it's okay if such a
>>
> BYE
> 
>>>comes from an Edgar? What do I not understand here?
>>
>>I will admit that this scenario seems pretty broken.
>>
>>However, if the call to bob reached edgar, and he has done as you say, 
>>it is futile for alice to reject the BYE. It is an indication that the 
>>other party has hung up, and refusing it will not change that.
> 
> 
> What I really meant here is that it makes no sense to provide Identity for a
> request in the backwards direction from Edgar if he is not the intended
> target of the call. Identity is useful only if it is going to factor into an
> authorization decision regarding impersonation. If literally any identity in
> the request must be accepted by Alice, rather than the unique identity of
> Bob, then Identity is a null op and we should stop trying to engineer for
> this case.

Providing identity for edgar help with any *automatic* authorization 
decision by alice's UA. But knowing that it is really edgar may 
nevertheless be useful to alice. Maybe alice knows that edgar is an 
associate of Bob's. That would be sigificant compared to getting an 
unauthenticated identity.

You mention this later, but discount its value more than I do.

  This is the "fourth alternative" I alluded to in my last mail -
> just say that request identity is not meaningful for any requests in a
> dialog other than the dialog-forming request.

Well, it may be another step down the path to someplace better.

  I do think we can do better
> than that, though, without creating serious problems for ourselves. If we
> choose the fourth alternative, we will be allowing a certain security hole
> to remain open, one that I believe we can close without too much trouble.
> 
> 
>>If edgar is just a MITM trying to disrupt a valid call between alice and 
>>bob, then maybe rejecting the BYE would accomplish something. But things 
>>will still be messed up - for instance CSeq values will be wrong. 
> 
> 
> By "reject" here I mean something more along the lines of "silently
> discard". A spoofed request should not alter the dialog state in the
> recipient.

But what if it was really a BYE from the other participant in my call? 
Just ignoring it isn't going to be a good thing. I think one would at 
least want to check it out - maybe send an update or reinvite to see if 
there is still a dialog up.

>>And if 
>>edgar was in a position to do this with his own name, he could just as 
>>easily have done it with bob's - aside from the issue of signing. Maybe
>>it makes a little sense in that edgar may be able to get the BYE signed 
>>with his own identity, though that will be tricky.
> 
> This is the rub. If Alice will accept only BYE requests from Bob, then
> Identity adds value - Edgar cannot impersonate Bob because of Identity.
> Identity only addresses threats of impersonation.

But if the far end party really is Edgar rather than Bob (because of 
retargetting) then the end result of this strategy is that no signalling 
from Edgar will work. No putting on hold, etc. Essentially the call is 
broken. Not a great strategy.

>>I do agree it would be better if we could come up with some explicit way 
>>to inform alice that the request has been delegated from bob to edgar. 
>>Maybe it is just a matter of choosing among existing mechanisms, rather 
>>than inventing something new. 
> 
> Totally agreed. sip-identity-02 suggested, albeit without sufficient
> motivation and argument, that the requirements for such a mechanism are
> satisfied by simple SIP redirection. I'm trying to keep that whole issue out
> of the scope of sip-identity-03/04, and one way to do that, I think, is to
> go along with something roughly like my "third alternative" proposal, which
> basically works when it can, and is forward-compatible with a future
> redirection-based responde identity model in which it would work all the
> time.

I think most of the things we are discussing are compatible with a 
future redirection based approach. What you propose is just a minimalist 
approach among those.

>>>Alice
>>>sends a request to Bob. For some reason, Bob's domain thinks it is
>>>appropriate to forward the request to, say, Carol. Why does Bob's domain
>>>think that? Because someone, with some relevant permissions, provisioned
>>>Bob's domain with a directive that requests for Bob should be forwarded
>>
> to
> 
>>>Carol. Of course, Carol personally does not possess Bob's credentials.
>>
> If
> 
>>>Bob is at Carol's station, though, then Bob will possess Bob's
>>
> credentials.
> 
>>>If the request arrives at Carol's station, and Bob fields the request,
>>
> then
> 
>>>Bob will be capable of providing his credentials to any auth service as
>>>necessary. Carol will not. 
>>
>>That of course assumes that Carol's phone is capable of prompting for 
>>credentials, such as asking for Bob's password. Maybe someday most 
>>phones will have that capability, but not today.
>>
>>And what if bob just forwarded to carol because he is away?
> 
> 
> In both of these cases, clearly, Identity should not be provided in the
> backwards direction. If the responder isn't actually Bob, then it shouldn't
> claim to be.

I don't think anybody (except maybe you once in this thread) is 
suggesting that should be able to claim to be Bob and receive 
identification as Bob.

However, the current rules for dialogs to suggest that within the 
dialog, Carol or Edgar will have to claim to be Bob. One question is 
whether to have them do that, or instead change the To/From within the 
dialog. Unless we simply ban this case altogether (mandating redirect to 
avoid it), then we must at least answer this question.

  But claiming to be someone else (Carol, or Edgar, or what have
> you) doesn't help Alice to make any meaningful authorization decisions, I
> think, unless Bob's domain has some way to communicate to Alice that the
> request has been forwarded to that particular someone.
> 
> This isn't to say that there might not be some counterexamples to that last
> principle - I mean, it's possible that the identity of the connected party
> will make it clear how/why the request was retargeted, and that this will
> seem reasonable from an authorization perspective to a caller with some
> pre-knowledge of the callee - but the existence of such counterexamples does
> not, I think, justify leaving this security hole wide open for those cases
> where we cannot rely on human pre-association and presumption to fill in the
> gaps of our protocol's authorization story. That is a non-deterministic
> security environment.

I think it is better to have Carol say she is Carol than say she is Bob. 
And it is better yet for that assertion to be verifiably signed. Yes, it 
still leaves a security hole that we can work on filling.

> Of course, if Bob's domain redirected rather than retargeting, the
> underlying problem would be solved.
> 
> 
>>>If the user at Carol's station possesses Bob's credentials, then it is
>>>reasonable for requests in the backwards direction to receive bear an
>>>Identity header stating that this is from Bob; if not, then there
>>
> shouldn't
> 
>>>be an Identity header in the request. This seems very intuitive to me;
>>
> I'm
> 
>>>not sure why this is so radically implausible. I guess this begs the
>>>question of why retargeting happens at all; if retargeting happens for
>>
> some
> 
>>>other reason than what I describe above, then it most likely violates
>>>RFC3261 as I understand it.
>>
>>In most scenarios I am aware of today, phone is either configured with 
>>the credentials, or in a hotelling situation the phone requests 
>>credentials to REGISTER somewhere. The hotelling situation is certainly 
>>better from a security viewpoint, and it doesn't look like retargetting 
>>at all to the caller.
>>
>>But hotelling isn't always going to be possible. Perhaps bob is visiting 
>>carol, and can configure his own AOR to retarget to carol, but can't 
>>hotel on carol's phone and can't provide credentials for an incoming 
>>call that was retargetted from bob to carol.
>>
>>What does this violate in 3261?
> 
> 
> Your two paragraphs above violate nothing about RFC3261. I was referring to
> the motivations for legitimate retargeting (that is, that Bob's domain is
> the only one that can retarget a request for Bob, and that Bob has an
> association with his domain that allows him to control his contact list).
> 
> While I don't disagree with you that the case described above is possible,
> I'm just saying that we shouldn't provide an Identity header in that case.
> 
> 
>>One solution to this would be for for the biloxi.com server to act as a 
>>b2bua, relaying the call to carol. Maybe that is a *model* to consider, 
>>with hopes of finding a better implementation. That is also roughly the 
>>model when a gateway is being used. We already know that in the gateway 
>>case the connected party on the far side of the gateway can change, and 
>>there are desires to convey that info.
> 
> (weep)
> 
> Maybe we could at least consider redirection before we start suggesting that
> this is a B2BUA problem.

Yes, of course. But there are those people who feel in fact that it is 
important that Alice *not* know the call has been retargetted, or at 
least *where* it has been retargetted. Those folks won't go for 
redirection. Of course they also then don't want to report back the 
connected party id either, so maybe that is out of scope for this 
discussion.

Like most things, we probably need a toolbag of alternatives that can be 
used according to the policies of the players.

> I really think that the problem of mid-call changes in connected-party is a
> very different case, and much more specialized than what we're concerned
> with here. We're talking here about assuring the identity of the first and
> second parties to a dialog at the time of dialog establishment. I'd like to
> think that the ultimate solutions to the mid-call 'identity update' sort of
> problems are more along the transfer/replaces line of thinking.

Depending on your preference for mechanisms, it may or may not be useful 
to conflate these problems. I guess I agree with you that we can 
consider them separately.

	Paul


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From: David R Oran <oran@cisco.com>
Subject: Re: third alternative (was RE: [Sip] RE: Identity after reinvite)
Date: Fri, 19 Nov 2004 09:30:08 -0500
To: "Peterson, Jon" <jon.peterson@neustar.biz>
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On Nov 18, 2004, at 3:46 PM, Peterson, Jon wrote:
>> One solution to this would be for for the biloxi.com server to act as 
>> a
>> b2bua, relaying the call to carol. Maybe that is a *model* to 
>> consider,
>> with hopes of finding a better implementation. That is also roughly 
>> the
>> model when a gateway is being used. We already know that in the 
>> gateway
>> case the connected party on the far side of the gateway can change, 
>> and
>> there are desires to convey that info.
>>
>
> (weep)
>
> Maybe we could at least consider redirection before we start 
> suggesting that
> this is a B2BUA problem.
>
Redirection is generally cleaner than retargeting, but does not satisfy 
the "Monica property". It may be that a B2BUA is inescapable in those 
cases anyway.

> I really think that the problem of mid-call changes in connected-party 
> is a
> very different case, and much more specialized than what we're 
> concerned
> with here. We're talking here about assuring the identity of the first 
> and
> second parties to a dialog at the time of dialog establishment. I'd 
> like to
> think that the ultimate solutions to the mid-call 'identity update' 
> sort of
> problems are more along the transfer/replaces line of thinking.
>
I tend to agree. It would be much cleaner if all 
identity/connected-party changes devolved to a flavor of transfer 
(except the Monica cases I mention above).

Dave.
> Jon Peterson
> NeuStar, Inc.
>
>> 	Paul
>>


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Date: Fri, 19 Nov 2004 08:59:15 -0800
From: Charles Eckel <eckelcu@cisco.com>
Organization: Cisco Systems
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To: Steve Langstaff <steve.langstaff@citel.com>
Subject: Re: [Sip] PING/PONG
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Cc: sip@ietf.org, Christian Stredicke <Christian.Stredicke@snom.de>
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The problem is that if you leave it entirely at the STUN level you are 
just ensuring that STUN server will be able to reach the STUN clients 
all times. However, what you really need is for the SIP proxy to be able 
to reach its SIP clients at all times. If the STUN server and proxy 
server are both listening and sending from the same port, you are fine 
using STUN by itself; otherwise, you need to do something else.

Cheers,
Charles

Steve Langstaff wrote:

> Sorry, I didn't realise the scope of this discussion was client-to-proxy 
> - that's where my confusion lay.
> 
> A further question from my (simplistic) viewpoint... If this issue 
> is/could be generic to all STUNned traffic (not just sip+stun), would it 
> not be better to push the problem down into the STUN 'layer' and let the 
> STUN client and server sort out keeping the NAT bindings alive? I'm 
> guessing that the SIP client already 'knows' it has to use STUN to reach 
> the proxy, and so could ask it's STUN client to keep the binding(s) 
> alive as appropriate.
> 
> 
> -- 
> Steve Langstaff.
> 
> 
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> Sent: 18 November 2004 04:46
> To: Steve Langstaff
> Cc: Christian Stredicke; sip@ietf.org
> Subject: Re: [Sip] PING/PONG
> 
> 
> There isn't a UAS involved here; these requests go from the client (UAC)
> to its proxy.
> 
> How the proxy indicates to the UAC that it is capable of processing
> these keepalives is a good question. My first thought is that this would
> be something in DNS; a new service in the NAPTR record for indicating
> the sip+stun combination.
> 
> -Jonathan R.
> 
> Steve Langstaff wrote:
> 
>  > Should point 2 read "The user agent server indicates if it can 
> respond to client refresh requests"?
>  >
>  > --
>  > Steve Langstaff.
>  >
>  >
>  > -----Original Message-----
>  > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
>  > Jonathan Rosenberg
>  > Sent: 17 November 2004 16:40
>  > To: Christian Stredicke
>  > Cc: sip@ietf.org
>  > Subject: Re: [Sip] PING/PONG
>  >
>  >
>  >
>  >
>  > Christian Stredicke wrote:
>  >
>  >
>  >>So my understanding is:
>  >>
>  >>1. We should use *only* STUN for refreshing bindings (either UDP or TCP,
>  >>what about TLS)
>  >
>  >
>  > Yes, TLS too. TLS runs ontop of TCP after all.
>  >
>  >
>  >>2. The user agent indicates if it can do it (no negotiation on
>  >>capabilities)
>  >>
>  >>3. Refreshing must be done from the client
>  >>
>  >>Can we agree on that?
>  >
>  >
>  > Yes.
>  >
>  > Server originating refreshing is in the wrong direction. It is the
>  > client utilizing the service; if it wishes to make use of that service,
>  > it has to be responsible for refreshing the connection.
>  >
>  > -Jonathan R.
>  >
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.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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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Mon Nov 22 02:18:44 2004
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Date: Mon, 22 Nov 2004 02:05:34 -0500
From: Jonathan Rosenberg <jdrosen@cisco.com>
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To: Charles Eckel <eckelcu@cisco.com>
Subject: Re: [Sip] PING/PONG
References: <CD9775120D600F43B9C50329395E9DB64F321F@ivor.citel.com>
	<419E2663.6000409@cisco.com>
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Cc: sip@ietf.org, Christian Stredicke <Christian.Stredicke@snom.de>,
        Steve Langstaff <steve.langstaff@citel.com>
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I think there may be some confusion here about how this works.

The STUN server is a logical function that *must* run on the same IP and 
port as the SIP server. In other words, if I have a proxy, and it is 
listening for SIP messages on 1.2.3.4:5060, then it would also be able 
to process STUN requests on 1.2.3.4:5060. A response to a stun request 
sent to that IP/port would get sent from that same port.

Thus, there is no separate stun server here. The stun functionality is 
co-resident with the sip proxy; this is necessary to make sure that the 
proxy knows that the client is still there, to allow the client to know 
that the proxy is still there, and lastly, to ensure that the keepalives 
refresh the nat bindings in a way that allow the client to continue to 
reach the SIP proxy server and vice-a-versa. Since the keepalives are 
sent to the same place where the signaling traffic goes/comes from, the 
keepalives work even with symmetric nat.

-Jonathan R.



Charles Eckel wrote:

> The problem is that if you leave it entirely at the STUN level you are 
> just ensuring that STUN server will be able to reach the STUN clients 
> all times. However, what you really need is for the SIP proxy to be able 
> to reach its SIP clients at all times. If the STUN server and proxy 
> server are both listening and sending from the same port, you are fine 
> using STUN by itself; otherwise, you need to do something else.
> 
> Cheers,
> Charles
> 
> Steve Langstaff wrote:
> 
>> Sorry, I didn't realise the scope of this discussion was 
>> client-to-proxy - that's where my confusion lay.
>>
>> A further question from my (simplistic) viewpoint... If this issue 
>> is/could be generic to all STUNned traffic (not just sip+stun), would 
>> it not be better to push the problem down into the STUN 'layer' and 
>> let the STUN client and server sort out keeping the NAT bindings 
>> alive? I'm guessing that the SIP client already 'knows' it has to use 
>> STUN to reach the proxy, and so could ask it's STUN client to keep the 
>> binding(s) alive as appropriate.
>>
>>
>> -- 
>> Steve Langstaff.
>>
>>
>> -----Original Message-----
>> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
>> Sent: 18 November 2004 04:46
>> To: Steve Langstaff
>> Cc: Christian Stredicke; sip@ietf.org
>> Subject: Re: [Sip] PING/PONG
>>
>>
>> There isn't a UAS involved here; these requests go from the client (UAC)
>> to its proxy.
>>
>> How the proxy indicates to the UAC that it is capable of processing
>> these keepalives is a good question. My first thought is that this would
>> be something in DNS; a new service in the NAPTR record for indicating
>> the sip+stun combination.
>>
>> -Jonathan R.
>>
>> Steve Langstaff wrote:
>>
>>  > Should point 2 read "The user agent server indicates if it can 
>> respond to client refresh requests"?
>>  >
>>  > --
>>  > Steve Langstaff.
>>  >
>>  >
>>  > -----Original Message-----
>>  > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
>>  > Jonathan Rosenberg
>>  > Sent: 17 November 2004 16:40
>>  > To: Christian Stredicke
>>  > Cc: sip@ietf.org
>>  > Subject: Re: [Sip] PING/PONG
>>  >
>>  >
>>  >
>>  >
>>  > Christian Stredicke wrote:
>>  >
>>  >
>>  >>So my understanding is:
>>  >>
>>  >>1. We should use *only* STUN for refreshing bindings (either UDP or 
>> TCP,
>>  >>what about TLS)
>>  >
>>  >
>>  > Yes, TLS too. TLS runs ontop of TCP after all.
>>  >
>>  >
>>  >>2. The user agent indicates if it can do it (no negotiation on
>>  >>capabilities)
>>  >>
>>  >>3. Refreshing must be done from the client
>>  >>
>>  >>Can we agree on that?
>>  >
>>  >
>>  > Yes.
>>  >
>>  > Server originating refreshing is in the wrong direction. It is the
>>  > client utilizing the service; if it wishes to make use of that 
>> service,
>>  > it has to be responsible for refreshing the connection.
>>  >
>>  > -Jonathan R.
>>  >
>>
>> -- 
>> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>> Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
>> Cisco Systems
>> jdrosen@cisco.com                              FAX:   (973) 952-5050
>> http://www.jdrosen.net                         PHONE: (973) 952-5000
>> http://www.cisco.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.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Mon Nov 22 03:11:52 2004
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From: Anurag Kabra <anuragk@aftek.com>
Organization: Aftek Infosys Ltd.
To: sip@ietf.org
Date: Mon, 22 Nov 2004 13:39:52 +0530
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Subject: [Sip] Call Pickup
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Hello Everyone

I have a query regarding call pickup, Is there any specific standard for=20
implementing directed call pickup.  I need to know the implementation det=
ails=20
of it.  I could only find an example in SIP service examples.

Thank You & Best Regards
         Anurag.

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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
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From sip-bounces@ietf.org  Mon Nov 22 04:59:33 2004
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From: "Jesske, R" <R.Jesske@t-com.net>
To: mary.barnes@nortelnetworks.com, sebastien.prouvost@francetelecom.com,
        sebastien.garcin@francetelecom.com, sip@ietf.org
Subject: AW: [Sip] Privacy statements and History(draft-ietf-sip-history-i
	nfo-04.txt)
Date: Mon, 22 Nov 2004 10:47:58 +0100
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Mary,
that looks good for me.
=20
Best Regards
=20
Roland

-----Urspr=FCngliche Nachricht-----
Von: Mary Barnes [mailto:mary.barnes@nortelnetworks.com]
Gesendet: Mittwoch, 17. November 2004 18:50
An: 'PROUVOST Sebastien RD-CORE-ISS'; GARCIN Sebastien RD-CORE-ISS; =
Jesske, Roland; sip@ietf.org
Betreff: RE: [Sip] Privacy statements and =
History(draft-ietf-sip-history-i nfo-04.txt)


Hi all,
=20
I've again snipped the thread, but did want to propose the change that =
I think will satisfy the concerns raised.
=20
The change proposed is to change the last statement in the first =
paragraph in section 4.3.3.1.1 from:
"...the proxy MUST remove any hi-entry(s) prior to forwarding."
to:
"...the proxy SHOULD remove any hi-entry(s) prior to forwarding, =
depending upon local policy and whether the proxy might know apriori =
that it can rely on a downstream privacy service to apply the requested =
privacy."=20
=20
So, unless further concerns are raised on this proposed change, I'll =
plan on incorporating that with any other last call comments in the -05 =
version.
=20
Mary=20


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
PROUVOST Sebastien RD-CORE-ISS
Sent: Thursday, November 11, 2004 8:44 AM
To: Barnes, Mary [NGC:B601:EXCH]; GARCIN Sebastien RD-CORE-ISS; Jesske, =
R; sip@ietf.org
Subject: RE: [Sip] Privacy statements and =
History(draft-ietf-sip-history-info-04.txt)


Mary, Sebastien, Roland,=20
=20
I agree that we should let the possibility for a proxy not to remove =
the hi-entry even if privacy is requested and even if the request is =
forwarded to a Request-URI associated with a domain for which the proxy =
is not responsible (if there is an agreement between the domains that =
ensures the proxy that privacy will be applied to the request).=20
I suggest a text that would look like (in case privacy is requested): =
"the hi-entry SHOULD be removed by the proxy unless it knows that it =
can rely on a downstream privacy service to apply the requested privacy =
".
=20
Sebastien.


  _____ =20

De : sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] De la part de =
Mary Barnes
Envoy=E9 : mercredi 10 novembre 2004 19:54
=C0 : GARCIN Sebastien RD-CORE-ISS; Jesske, R; sip@ietf.org
Objet : RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)



I still contend that the SHOULD is sufficient.  SHOULD means that in =
general processing, if the hi-entry has a privacy header, then the =
usual processing would be that it would be removed (i.e it was added =
for a specific reason and in general should be used to remove the =
entries based on well defined criteria).  If there are reasons, such as =
local policy, that would allow the forwarding in specific cases, then =
it's okay that it is forwarded.  I think the use of MAY results in less =
precision and I think the value and critera for associating and =
removing the privacy header with the hi-entry becomes much less clear.  =


I'd like to hear more opinions on this topic, prior to agreeing to =
making the change (from the MUST to MAY rather than MUST to SHOULD).=20

Regards,=20
Mary=20


-----Original Message-----=20
From: GARCIN Sebastien RD-CORE-ISS [ =
mailto:sebastien.garcin@francetelecom.com =
<mailto:sebastien.garcin@francetelecom.com> ]=20
Sent: Wednesday, November 10, 2004 12:28 PM=20
To: Barnes, Mary [NGC:B601:EXCH]; Jesske, R; sip@ietf.org=20
Cc: VL-T-Com-T-TE332@vli.telekom.de=20
Subject: RE: [Sip] Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)=20


Mary and roland,=20

Actually, the statement regarding the forwarding of hi-entries beyond =
the domain for which the proxy is responsible should rather be a matter =
of local policy and thus a MAY should be used instead of SHOULD. We are =
potentially dealing with network boundaries where agreements for =
forwarding such kind information can be reached. 

Section 4.3.3.1.1=20
This section should be re-reworded in accordance with the statement =
above (I can provide some text is we can agree).=20

Section 4.3.3.1 is ok (apologies for not being explicit)=20

Best regards,=20
s=E9bastien=20

                                =20

                                ------Remainder of thread has been =
deleted by Mary------------------



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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.50.4937.800" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D379244709-22112004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2>Mary,</FONT></SPAN></DIV>
<DIV><SPAN class=3D379244709-22112004><FONT face=3DArial =
color=3D#0000ff size=3D2>that=20
looks good for me.</FONT></SPAN></DIV>
<DIV><SPAN class=3D379244709-22112004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D379244709-22112004><FONT face=3DArial =
color=3D#0000ff size=3D2>Best=20
Regards</FONT></SPAN></DIV>
<DIV><SPAN class=3D379244709-22112004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D379244709-22112004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2>Roland</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Urspr=FCngliche Nachricht-----<BR><B>Von:</B> Mary =
Barnes=20
  [mailto:mary.barnes@nortelnetworks.com]<BR><B>Gesendet:</B> Mittwoch, =
17.=20
  November 2004 18:50<BR><B>An:</B> 'PROUVOST Sebastien RD-CORE-ISS'; =
GARCIN=20
  Sebastien RD-CORE-ISS; Jesske, Roland; =
sip@ietf.org<BR><B>Betreff:</B> RE:=20
  [Sip] Privacy statements and History(draft-ietf-sip-history-i=20
  nfo-04.txt)<BR><BR></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D514513817-17112004>Hi=20
  all,</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D514513817-17112004></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D514513817-17112004>I've=20
  again snipped the thread, but did want to propose the change that I =
think will=20
  satisfy the concerns raised.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D514513817-17112004></SPAN></FONT><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2><SPAN class=3D514513817-17112004></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D514513817-17112004>The=20
  change proposed is to change the last statement in the first =
paragraph in=20
  section 4.3.3.1.1 from:</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D514513817-17112004>"...<FONT size=3D3><FONT =
color=3D#000000><FONT=20
  face=3D"Courier New">the proxy MUST remove any hi-entry(s) prior to=20
  forwarding."</FONT></FONT></FONT></SPAN></FONT></DIV>
  <DIV><FONT size=3D+0><SPAN class=3D514513817-17112004><FONT =
size=3D+0><FONT=20
  face=3DArial color=3D#0000ff =
size=3D2>to:</FONT></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D514513817-17112004><FONT size=3D3><FONT color=3D#000000><FONT =

  face=3D"Courier New">"...the proxy&nbsp;SHOULD remove any hi-entry(s) =
prior to=20
  forwarding, depending upon local policy&nbsp;and whether the proxy =
might know=20
  apriori that it can rely on a downstream privacy service to apply the =

  requested privacy."<SPAN=20
  style=3D"mso-spacerun: =
yes">&nbsp;</SPAN></FONT></FONT></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D514513817-17112004><FONT size=3D3><FONT color=3D#000000><FONT =

  face=3D"Courier New"><SPAN=20
  style=3D"mso-spacerun: =
yes"></SPAN></FONT></FONT></FONT></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT size=3D+0><SPAN class=3D514513817-17112004><FONT =
size=3D+0><FONT=20
  face=3DArial color=3D#0000ff size=3D2><SPAN style=3D"mso-spacerun: =
yes">So, unless=20
  further concerns are raised on this proposed change, I'll plan on=20
  incorporating that with any other last call comments in the -05=20
  version.</SPAN></FONT></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D514513817-17112004><FONT size=3D3><FONT color=3D#000000><FONT =

  face=3D"Courier New"><SPAN=20
  style=3D"mso-spacerun: =
yes"></SPAN></FONT></FONT></FONT></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D514513817-17112004><FONT size=3D3><FONT=20
  size=3D+0><FONT face=3D"Courier New"><SPAN=20
  style=3D"mso-spacerun: =
yes"></SPAN></FONT></FONT></FONT></SPAN></FONT><FONT=20
  color=3D#0000ff><SPAN lang=3Den-us><FONT face=3DArial =
size=3D2>Mary</FONT></SPAN>=20
  </FONT></DIV>
  <BLOCKQUOTE style=3D"MARGIN-RIGHT: 0px">
    <DIV><FONT color=3D#0000ff></FONT></DIV>
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
    face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
    sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B>On Behalf Of=20
    </B>PROUVOST Sebastien RD-CORE-ISS<BR><B>Sent:</B> Thursday, =
November 11,=20
    2004 8:44 AM<BR><B>To:</B> Barnes, Mary [NGC:B601:EXCH]; GARCIN =
Sebastien=20
    RD-CORE-ISS; Jesske, R; sip@ietf.org<BR><B>Subject:</B> RE: [Sip] =
Privacy=20
    statements and=20
    History(draft-ietf-sip-history-info-04.txt)<BR><BR></FONT></DIV>
    <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
    color=3D#0000ff size=3D2>Mary, Sebastien, Roland, =
</FONT></SPAN></DIV>
    <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
    color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
    color=3D#0000ff size=3D2>I agree that we should let the possibility =
for a proxy=20
    not to remove the hi-entry even if privacy is requested and even if =
the=20
    request is forwarded to a Request-URI associated with a domain for =
which the=20
    proxy is not responsible&nbsp;(if there&nbsp;is an =
agreement&nbsp;between=20
    the&nbsp;domains that ensures the proxy that privacy will be =
applied to the=20
    request). </FONT></SPAN></DIV>
    <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
    color=3D#0000ff size=3D2>I suggest a&nbsp;text that would look =
like&nbsp;(in=20
    case privacy is requested): "the hi-entry SHOULD be removed by the =
proxy=20
    unless it knows that it can rely on a downstream privacy service to =
apply=20
    the requested privacy ".</FONT></SPAN></DIV>
    <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
    color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV dir=3Dltr align=3Dleft><SPAN class=3D000254113-11112004><FONT =
face=3DArial=20
    color=3D#0000ff size=3D2>Sebastien.</FONT></SPAN></DIV><FONT =
face=3DArial=20
    color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D#0000ff=20
    size=3D2></FONT><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT><FONT face=3DArial=20
    color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT><BR>
    <DIV class=3DOutlookMessageHeader lang=3Dfr dir=3Dltr align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>De&nbsp;:</B> sip-bounces@ietf.org=20
    [mailto:sip-bounces@ietf.org] <B>De la part de</B> Mary=20
    Barnes<BR><B>Envoy=E9&nbsp;:</B> mercredi 10 novembre 2004=20
    19:54<BR><B>=C0&nbsp;:</B> GARCIN Sebastien RD-CORE-ISS; Jesske, R; =

    sip@ietf.org<BR><B>Objet&nbsp;:</B> RE: [Sip] Privacy statements =
and History=20
    (draft-ietf-sip-history-info-04.txt)<BR></FONT><BR></DIV>
    <DIV></DIV>
    <P><FONT size=3D2>I still contend that the SHOULD is =
sufficient.&nbsp; SHOULD=20
    means that in general processing, if the hi-entry has a privacy =
header, then=20
    the usual processing would be that it would be removed (i.e it was =
added for=20
    a specific reason and in general should be used to remove the =
entries based=20
    on well defined criteria).&nbsp; If there are reasons, such as =
local policy,=20
    that would allow the forwarding in specific cases, then it's okay =
that it is=20
    forwarded.&nbsp; I think the use of MAY results in less precision =
and I=20
    think the value and critera for associating and removing the =
privacy header=20
    with the hi-entry becomes much less clear.&nbsp; </FONT></P>
    <P><FONT size=3D2>I'd like to hear more opinions on this topic, =
prior to=20
    agreeing to making the change (from the MUST to MAY rather than =
MUST to=20
    SHOULD). </FONT></P>
    <P><FONT size=3D2>Regards, </FONT><BR><FONT size=3D2>Mary</FONT> =
</P><BR>
    <P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From:=20
    GARCIN Sebastien RD-CORE-ISS [<A=20
    =
href=3D"mailto:sebastien.garcin@francetelecom.com">mailto:sebastien.garc=
in@francetelecom.com</A>]=20
    </FONT><BR><FONT size=3D2>Sent: Wednesday, November 10, 2004 12:28 =
PM</FONT>=20
    <BR><FONT size=3D2>To: Barnes, Mary [NGC:B601:EXCH]; Jesske, R;=20
    sip@ietf.org</FONT> <BR><FONT size=3D2>Cc:=20
    VL-T-Com-T-TE332@vli.telekom.de</FONT> <BR><FONT size=3D2>Subject: =
RE: [Sip]=20
    Privacy statements and History =
(draft-ietf-sip-history-info-04.txt)</FONT>=20
    </P><BR>
    <P><FONT size=3D2>Mary and roland,</FONT> </P>
    <P><FONT size=3D2>Actually, the statement regarding the forwarding =
of=20
    hi-entries beyond the domain for which the proxy is responsible =
should=20
    rather be a matter of local policy and thus a MAY should be used =
instead of=20
    SHOULD. We are potentially dealing with network boundaries where =
agreements=20
    for forwarding such kind information can be reached. </FONT></P>
    <P><FONT size=3D2>Section 4.3.3.1.1</FONT> <BR><FONT size=3D2>This =
section=20
    should be re-reworded in accordance with the statement above (I can =
provide=20
    some text is we can agree).</FONT> </P>
    <P><FONT size=3D2>Section 4.3.3.1 is ok (apologies for not being=20
    explicit)</FONT> </P>
    <P><FONT size=3D2>Best regards,</FONT> <BR><FONT =
size=3D2>s=E9bastien</FONT> </P>
    =
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<FONT=20
    size=3D2> </FONT></P>
    <P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
size=3D2>------Remainder of=20
    thread has been deleted by=20
Mary------------------</FONT></P><BR></BLOCKQUOTE></BLOCKQUOTE></BODY></=
HTML>

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'sip@ietf.org'" <sip@ietf.org>
Subject: concrete proposal (was RE: third alternative (was RE: [Sip] RE: I
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In the interests of trying to get this thread strictly onto the problems
with request identity, let me try to articulate the major point of
contention here, and make a concrete proposal that attempts to incorporate
the discussion to date.

The argument has been put forward that when a request travels in the
backwards direction in a dialog, it should be permissable for the remote UA
to change the addr-spec element (and potentially anything other than the
tag) in the From header field such that the new From header will not
correspond with the To header of the dialog-forming request. This seems like
an intuitive approach when a request has been retargeted, and consequently
the entity identified in the To header field of the dialog-forming request
is not the entity sending requests in the backwards direction (the From
header field of such requests provides, in this case, the 'connected
party'). RFC3261 allows the contents of the To and From headers to be
altered by the parties to a dialog, with the caveat that this practice is
known not to be compatible with RFC2543. 

An alternative argument has been made that the notion of an unanticipated
'connected party' is inherently antithetical to security, because the local
UA will be forced to accept a request from any party, as the potential set
of valid connected parties in unbounded.

Clearly, there are a number of possible ways to attack the connected party
problem, and not all of them have any impact on the sip-identity solution.
The main benefit of changing the From header is that in cases of
retargeting, the remote UA can thereby use the From header to provide
'connected party' information (rather than defining some new header or
something). The main downside is that the local UA will receive a request
within the dialog from a party other than the party to which the
dialog-forming request was sent, and that accordingly, attackers can easily
gain control of the dialog. In this environment, it is impossible to make
any automatic authorization decision about the validity of the request, and
the user of the local UA may or may not have some other means to think that
the request should be authorized, based on who the connected-party is.
Accordingly, the value of providing an Identity header for such requests is
questionable, since it isn't clear what impersonation threat might be met by
providing identity here. Really, the only possible motivation for using
sip-identity in this threat model is a case where a fourth party, Eve,
impersonates Edgar, anticipating that Alice knows that Edgar is an associate
of Bob, and that thus, when Alice calls Bob, it would be unsurprising if
Edgar answered. That begins to make my head hurt.

Given that RFC3261 permits UAS's to change the From header field value
(irregardless of whether or not the sip-identity mechanism is used), the
question here is how should an authentication service, as defined in
sip-identity, treat such requests? Should it add an Identity header, proxy
them, discard them, etc? Should sip-identity recommend that the From header
be changed in this manner when a request has been retargeted?

Currently, sip-identity-03 is silent about dialogs and their impact, if any,
on signing requests. It does say that an auth service MUST NOT provide an
Identity header for the request if it is not responsible for the domain
provided in the From header field of the request. Furthermore, the text
suggests that if the auth service is not responsible for the domain in the
>From header field, it MAY just proxy the request normally.

Given all of this, let me make a concrete proposal for sip-identity-04 which
attempts, for the most part, to dodge the issue of how we solve
connected-party:

We add some text about dialogs, and about the use of the Identity header
during a dialog. Obviously, the use of identity for dialog-forming requests
is a no-brainer (this is the standard 'caller-ID' assurance). Futhermore,
point out that if the intended second party is actually the connected party,
identity can be supplied normally in requests in the backwards direction.
Stipulate that an auth service instantiated in a proxy MUST Record-Route
itself in a dialog-forming request, so that the auth service can recognize a
mid-dialog request.

The Identity header simply would not be provided if the connected party is
different from the party to which the local UA sent the request. We keep the
MUST NOT for providing the Identity header in cases where the auth service
is not responsible for the domain in the From header field, but upgrade the
normative strength of the proxying MAY to STRONGLY RECOMMENDED (if the auth
service does not proxy in this instance, failures of basic SIP operations
can occur). This way, if an auth service in the request path sees a
backwards-direction request, and knows that it is mid-dialog, it can just
proxy the request normally if it is not responsible for the domain in
question.

Provide no recommendation that the From header field should be changed to
reflect the 'connected party' when the Identity mechanism is used; in fact,
RECOMMEND against doing so. Provide a reference to future work on this
'connected-party' problem.

Anyone unhappy with that?

Jon Peterson
NeuStar, Inc.

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'David R Oran'" <oran@cisco.com>
Subject: RE: third alternative (was RE: [Sip] RE: Identity after reinvite)
Date: Mon, 22 Nov 2004 20:29:53 -0500
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A few notes on the "Monica property": as RFC3323 suggests, I think that some
sort of anonymization service, which could be instantiated by a B2BUA, is an
appropriate way to conceal the identity of a called or calling party. What
scares me a little is the way people suggest that retargeting, as opposed to
redirection, can inherently be expected to provide privacy. When I register
a contact, I can't always guarantee that the location service in question
will be used exclusively for retargeting rather than redirection; moreover,
the revelation of the target Request-URI to the caller is not the only way
that the caller can learn who the connected-party is: contact headers of new
requests in the backwards direction of the dialog, SDP in a 200 OK, and so
on, are information leaks that are not addressed at all by retargeting, but
would be addressed by an anonymization service.

RFC3323 is about due for an update, I think, because the major functions
that it provides can be performed without a B2BUA, thanks to new SIP
mechanisms like GRUUs and session-policy. So while I agree that an
anonymization service is the right approach to this problem, these days I
think you can probably build an anonymization service without building a
B2BUA.

Finally, I think RFC3323 plays well with redirection. In fact, one can
redirect to a URI like "sip:anonymous0013@anonymizer.com", and the caller
can be satisfied, when they see new requests in the backwards direction,
that they are talking to the entity that the original target domain wanted
them to talk to. On the other hand, if my request is just retargeted to that
anonymous URI without my knowledge, I'll always be left to wonder, when I
see new requests in the backwards direction, if this is the person I'm
supposed to be talking to...

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: David R Oran [mailto:oran@cisco.com]
> Sent: Friday, November 19, 2004 6:30 AM
> To: Peterson, Jon
> Cc: 'Cullen Jennings'; 'Paul Kyzivat'; 'sip@ietf.org'
> Subject: Re: third alternative (was RE: [Sip] RE: Identity after
> reinvite)
> 
> 
[snip]
>
> Redirection is generally cleaner than retargeting, but does not satisfy 
> the "Monica property". It may be that a B2BUA is inescapable in those 
> cases anyway.
> 
> > I really think that the problem of mid-call changes in 
> connected-party 
> > is a
> > very different case, and much more specialized than what we're 
> > concerned
> > with here. We're talking here about assuring the identity 
> of the first 
> > and
> > second parties to a dialog at the time of dialog establishment. I'd 
> > like to
> > think that the ultimate solutions to the mid-call 'identity update' 
> > sort of
> > problems are more along the transfer/replaces line of thinking.
> >
> I tend to agree. It would be much cleaner if all 
> identity/connected-party changes devolved to a flavor of transfer 
> (except the Monica cases I mention above).
> 
> Dave.
> > Jon Peterson
> > NeuStar, Inc.
> >
> >> 	Paul
> >>
> 

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Hi All,

      Is there is any TLS version which will support IPv6  ??
     [ TLS (SSL) version 9.8 which supports Ipv6, is anybody knows wher=
e we
can download the same ]

      Is there is any ARES Library (C / C++) which supports IPv6 ??

With Regards,
K.Kalyana Chakravarthy,
Software Engineer,
Hughes Software Systems,
Ph : +91 - 80 - 51067085.


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<html><body>
<p>Hi All,<br>
<br>
      Is there is any TLS version which will support IPv6  ??<br>
     [ TLS (SSL) version 9.8 which supports Ipv6, is anybody knows wher=
e we can download the same ]<br>
<br>
      Is there is any ARES Library (C / C++) which supports IPv6 ??<br>=

<br>
With Regards,<br>
K.Kalyana Chakravarthy,<br>
Software Engineer,<br>
Hughes Software Systems,<br>
Ph : +91 - 80 - 51067085.<br>
<br>
<br>
***********************  HSS-Unclassified   ***********************<br>=

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Subject: Re: [Sip] openssl for IPv6  & ARES library availability
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kkotha@hssworld.com writes:
>       Is there is any TLS version which will support IPv6  ??
TLS doesn't care which version of IP it runs on, and most implementations
of TLS let you form your own socket and just hand it to the TLS engine.

-Ekr

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On Nov 22, 2004, at 8:28 PM, Peterson, Jon wrote:

>
> In the interests of trying to get this thread strictly onto the 
> problems
> with request identity, let me try to articulate the major point of
> contention here, and make a concrete proposal that attempts to 
> incorporate
> the discussion to date.
>
Thanks, this is a useful and cogent description/context for figuring 
out what the options are.

> The argument has been put forward that when a request travels in the
> backwards direction in a dialog, it should be permissable for the 
> remote UA
> to change the addr-spec element (and potentially anything other than 
> the
> tag) in the From header field such that the new From header will not
> correspond with the To header of the dialog-forming request. This 
> seems like
> an intuitive approach when a request has been retargeted, and 
> consequently
> the entity identified in the To header field of the dialog-forming 
> request
> is not the entity sending requests in the backwards direction (the From
> header field of such requests provides, in this case, the 'connected
> party'). RFC3261 allows the contents of the To and From headers to be
> altered by the parties to a dialog, with the caveat that this practice 
> is
> known not to be compatible with RFC2543.
>
> An alternative argument has been made that the notion of an 
> unanticipated
> 'connected party' is inherently antithetical to security, because the 
> local
> UA will be forced to accept a request from any party, as the potential 
> set
> of valid connected parties in unbounded.
>
This is true only in the absence of explicit delegation. At the risk of 
skipping ahead of your well-done chain of logic here, I think that we 
have in SIP a large number of scenarios (retargeting, REFER, 
Third-party control) where in the absence of explicit delegation we 
either (a) have no security, (b) force cooperating parties to use 
impersonation, or (c) in some cases, can rely on transitivity 
properties. Despite the complexity, we may be better off tackling the 
delegation problem now rather than allowing half-assed things to 
use/abuse the identity stuff because it solves a part of the problem 
(albeit a large part), but not enough.

> Clearly, there are a number of possible ways to attack the connected 
> party
> problem, and not all of them have any impact on the sip-identity 
> solution.
> The main benefit of changing the From header is that in cases of
> retargeting, the remote UA can thereby use the From header to provide
> 'connected party' information (rather than defining some new header or
> something). The main downside is that the local UA will receive a 
> request
> within the dialog from a party other than the party to which the
> dialog-forming request was sent, and that accordingly, attackers can 
> easily
> gain control of the dialog.
Unless the retarget is explicitly delegated...

> In this environment, it is impossible to make
> any automatic authorization decision about the validity of the 
> request, and
> the user of the local UA may or may not have some other means to think 
> that
> the request should be authorized, based on who the connected-party is.
Impossible is a bit too strong, but the general case of retargeting 
does have this problem.

> Accordingly, the value of providing an Identity header for such 
> requests is
> questionable, since it isn't clear what impersonation threat might be 
> met by
> providing identity here.
Questionable only if it is not extended to allow SAML or some other 
explicit means of delegation.

> Really, the only possible motivation for using
> sip-identity in this threat model is a case where a fourth party, Eve,
> impersonates Edgar, anticipating that Alice knows that Edgar is an 
> associate
> of Bob, and that thus, when Alice calls Bob, it would be unsurprising 
> if
> Edgar answered. That begins to make my head hurt.
>
For the 1-degree transitive closure of buddy-lists this does not make 
my head hurt and in fact might be quite natural in many IM and 
push-to-talk style-applications. For arbitrary n-degrees-of-separation 
scenarios I agree with you completely.

> Given that RFC3261 permits UAS's to change the From header field value
> (irregardless of whether or not the sip-identity mechanism is used), 
> the
> question here is how should an authentication service, as defined in
> sip-identity, treat such requests? Should it add an Identity header, 
> proxy
> them, discard them, etc? Should sip-identity recommend that the From 
> header
> be changed in this manner when a request has been retargeted?
>
It would be nice if the identity service did not have to be context 
aware and just signed identity headers for any request from a source it 
was authoritative for. Unless I misunderstand you I think you are 
contemplating the identity service being authz policy aware, and my 
intuition tells me that's a bad idea.

> Currently, sip-identity-03 is silent about dialogs and their impact, 
> if any,
> on signing requests. It does say that an auth service MUST NOT provide 
> an
> Identity header for the request if it is not responsible for the domain
> provided in the From header field of the request. Furthermore, the text
> suggests that if the auth service is not responsible for the domain in 
> the
>> From header field, it MAY just proxy the request normally.
>
This seems simple and keep the identity service out of authz policy.

> Given all of this, let me make a concrete proposal for sip-identity-04 
> which
> attempts, for the most part, to dodge the issue of how we solve
> connected-party:
>
> We add some text about dialogs, and about the use of the Identity 
> header
> during a dialog. Obviously, the use of identity for dialog-forming 
> requests
> is a no-brainer (this is the standard 'caller-ID' assurance). 
> Futhermore,
> point out that if the intended second party is actually the connected 
> party,
> identity can be supplied normally in requests in the backwards 
> direction.
> Stipulate that an auth service instantiated in a proxy MUST 
> Record-Route
> itself in a dialog-forming request, so that the auth service can 
> recognize a
> mid-dialog request.
>
I think I understand what you mean here, but the words could be better. 
I think you are saying that "if a sip server acts as an auth service 
and proxies a request that it auth'ed itself, it MUST record route".

> The Identity header simply would not be provided if the connected 
> party is
> different from the party to which the local UA sent the request.
Why? If the auth service can authenticate the From in the request, why 
should it refuse to do so?

> We keep the
> MUST NOT for providing the Identity header in cases where the auth 
> service
> is not responsible for the domain in the From header field, but 
> upgrade the
> normative strength of the proxying MAY to STRONGLY RECOMMENDED (if the 
> auth
> service does not proxy in this instance, failures of basic SIP 
> operations
> can occur). This way, if an auth service in the request path sees a
> backwards-direction request, and knows that it is mid-dialog, it can 
> just
> proxy the request normally if it is not responsible for the domain in
> question.
>
> Provide no recommendation that the From header field should be changed 
> to
> reflect the 'connected party' when the Identity mechanism is used; in 
> fact,
> RECOMMEND against doing so. Provide a reference to future work on this
> 'connected-party' problem.
>
> Anyone unhappy with that?
>
Modulo my one issue above, I'm ok with the identity part, but as I 
mentioned, doing just this doesn't actually solve anything other than 
getting identity done (which is laudable and necessary). I don't want 
to slow identity down UNLESS there's reason to believe we will have 
backward compatibility problems when we introduce delegation, OR we 
think that identity will get abused or ignored because too many 
real-world situations really need delegation.

Dave.

David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com

> Jon Peterson
> NeuStar, Inc.

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From sip-bounces@ietf.org  Tue Nov 23 13:17:49 2004
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Subject: [Sip] Does retargeting belong on the list of Evil SIP Ideas? 
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I try to keep a running list of things  that turned out to be a royal 
pain that needs (or needed) fixing. Some of these things sounded cool at 
first, and I championed them.  Others I never did like. Some of these 
things I'm not allowed to suggest deprecating anymore -- we took a hum 
on forking a few meetings back. One of these days I'm going to get this 
list better organized and  post it someplace more persistent in the hope 
that it may assist in avoiding repeptition of mistakes. And someday, 
perhaps we can do something about some of these quirks.

To-date, I believe the Evil List includes:

1) Replacing the Request-URI at each hop while routing was Evil. We 
fixed this with Loose Routing mode, which is still slightly Evil.

2) Abusing To and From to indicate branches is Evil. We should have left 
these as real endpoint identifiers.

3) Forking is Evil.

4)  Allowing (and requiring)  responses to carry a payload with more 
meaning than "I acknowledge having received that request" is Evil. Since 
we can't reject a response, we can't deal with big responses and can't 
negotiate what they carry. Acknowledgement and response are different 
beasties -- one is transport, the other is application, and we need to 
understand the difference.

5) The three-message INVITE transaction model and its dependence on 
payload in the response is Evil.

6) The two-message non-INVITE transaction model and its interaction with 
retransmission and hop-by-hop cascade (See Sparks'  non-invite analysis) 
is Evil.

7) Building transport state and reliability into the application state 
and requiring a transport protocol extension for every new application 
semantic is Evil.

8) Using z9hG4bK on via fields rather than a negotiated mechanism (like 
maybe a version) is Evil.

9) Jon argues that not requiring SIPS to be end-to-end is Evil. I think 
the jury is still out on this one, but we're still deliberating.

10) Provisional responses, as currently constructed, whether reliable or 
non-relaible, are Evil. This may be redundant with the NIT discussion.

11)  I believe I'm coming to believe that "retargeting",  i.e. changing 
a request in a proxy such that the entity to which it is delivered  has 
a different assertable identity than the identity to which the request 
was originally addressed,  is Evil. Perhaps it could be mitigated by a 
assertion of transitivity: "I, the proxy identified herein, do assert 
that I have altered the target of this request according to local 
policy, and you can either deal with it or go away." Oh wait, that's the 
302 response . . . Perhaps this is redundant with "Forking is evil" as 
forking is just a special case of retargeting. Or perhaps forking and 
retargeting are both special cases of "exploders" and could both be 
dealt with using the consent framework . . .

Sometimes I wonder if  the use of UDP was fundamentally Evil.

If I've missed something blatantly Evil, please comment back, and I may 
add it to the list I hope to post someplace.

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From sip-bounces@ietf.org  Tue Nov 23 16:39:08 2004
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I generally agree with Jon and Dave. I've snipped to something I want to 
comment on.

	Paul

David R Oran wrote:
> 
> On Nov 22, 2004, at 8:28 PM, Peterson, Jon wrote:
> 
>> We add some text about dialogs, and about the use of the Identity header
>> during a dialog. Obviously, the use of identity for dialog-forming 
>> requests
>> is a no-brainer (this is the standard 'caller-ID' assurance). Futhermore,
>> point out that if the intended second party is actually the connected 
>> party,
>> identity can be supplied normally in requests in the backwards direction.
>> Stipulate that an auth service instantiated in a proxy MUST Record-Route
>> itself in a dialog-forming request, so that the auth service can 
>> recognize a
>> mid-dialog request.
>>
> I think I understand what you mean here, but the words could be better. 
> I think you are saying that "if a sip server acts as an auth service and 
> proxies a request that it auth'ed itself, it MUST record route".

In the general case the two ends of the call are in different domains, 
and so there need to be two auth servers. The initial call setup (from 
Alice to Bob) presumably transits Alice's auth server so that it can be 
signed. Jon's requirement above would require *it* to record route. That 
might be a good thing, so that subsequent requests from Alice can also 
be signed. But that does nothing for requests from Bob to Alice.

There is no guarantee that Bob's auth server is in the path for incoming 
requests from Alict to Bob. Based on the needs of Identity-03 there is 
no need for it to be. If it happened to be, then the requirement that it 
record-route might be helpful in giving it the opportunity to sign 
future outbound requests in the dialog.

If it wasn't in the path initially, then Bob will have to add a Route 
header, in addition to what is received in Record-Route, to involve it 
when sending subsequent requests within the dialog.

It seems to me that depending on how the network components are 
constructed, either the auth server will be collocated with a component 
that is naturally on the path, or else each end will have to manipulate 
the Route headers to ensure it is. (It would naturally be on the path if 
  it was collocated with teh proxy managing the user's AOR and the user 
uses a GRUU managed by the same proxy for contacts.)

	Paul


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From sip-bounces@ietf.org  Tue Nov 23 16:45:38 2004
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This conversation just got too loose with its terminology for me.

3261 proxies have a UAS core.

Endpoints behind the things we are wanting to do this refresh things to 
can be signaling
directly with endpoints that aren't (a service provider could let 
people talk to voice-mail
or conference servers directly).

While the long-lived client-to-service-provider-proxy connection is a 
good example
to use for motivation, I would prefer that we ensure we build something 
that works
in general.

The DNS solution would work for those the services I call out above.  
I'm not yet sure
if its the right way to go.

I can see a use for being able to signal (or at least detect) support 
for this kind of keep-alive
between ephemeral endpoints too.

RjS

On Nov 18, 2004, at 5:14 AM, Steve Langstaff wrote:

> Sorry, I didn't realise the scope of this discussion was 
> client-to-proxy - that's where my confusion lay.
>
> A further question from my (simplistic) viewpoint... If this issue 
> is/could be generic to all STUNned traffic (not just sip+stun), would 
> it not be better to push the problem down into the STUN 'layer' and 
> let the STUN client and server sort out keeping the NAT bindings 
> alive? I'm guessing that the SIP client already 'knows' it has to use 
> STUN to reach the proxy, and so could ask it's STUN client to keep the 
> binding(s) alive as appropriate.
>
>
> --
> Steve Langstaff.
>
>
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> Sent: 18 November 2004 04:46
> To: Steve Langstaff
> Cc: Christian Stredicke; sip@ietf.org
> Subject: Re: [Sip] PING/PONG
>
>
> There isn't a UAS involved here; these requests go from the client 
> (UAC)
> to its proxy.
>
> How the proxy indicates to the UAC that it is capable of processing
> these keepalives is a good question. My first thought is that this 
> would
> be something in DNS; a new service in the NAPTR record for indicating
> the sip+stun combination.
>
> -Jonathan R.
>
> Steve Langstaff wrote:
>
>> Should point 2 read "The user agent server indicates if it can 
>> respond to client refresh requests"?
>>
>> --
>> Steve Langstaff.
>>
>>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
>> Jonathan Rosenberg
>> Sent: 17 November 2004 16:40
>> To: Christian Stredicke
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] PING/PONG
>>
>>
>>
>>
>> Christian Stredicke wrote:
>>
>>
>>> So my understanding is:
>>>
>>> 1. We should use *only* STUN for refreshing bindings (either UDP or 
>>> TCP,
>>> what about TLS)
>>
>>
>> Yes, TLS too. TLS runs ontop of TCP after all.
>>
>>
>>> 2. The user agent indicates if it can do it (no negotiation on
>>> capabilities)
>>>
>>> 3. Refreshing must be done from the client
>>>
>>> Can we agree on that?
>>
>>
>> Yes.
>>
>> Server originating refreshing is in the wrong direction. It is the
>> client utilizing the service; if it wishes to make use of that 
>> service,
>> it has to be responsible for refreshing the connection.
>>
>> -Jonathan R.
>>
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ 
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.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: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] openssl for IPv6  & ARES library availability
Date: Tue, 23 Nov 2004 15:32:14 -0600
To: kkotha@hssworld.com
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Work has been done on ARES in resiprocate/contrib to support v6 DNS 
servers
(and AAAA records from both v4 and v6 servers).

See http://www.sipfoundry.org/reSIProcate/index.html

RjS

On Nov 23, 2004, at 3:01 AM, kkotha@hssworld.com wrote:

> Hi All,
>
>  Is there is any TLS version which will support IPv6 ??
>  [ TLS (SSL) version 9.8 which supports Ipv6, is anybody knows where 
> we can download the same ]
>
>  Is there is any ARES Library (C / C++) which supports IPv6 ??
>
>  With Regards,
>  K.Kalyana Chakravarthy,
>  Software Engineer,
>  Hughes Software Systems,
>  Ph : +91 - 80 - 51067085.
>
>
>  *********************** HSS-Unclassified ***********************
>  "DISCLAIMER: This message is proprietary to Hughes Software Systems 
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> _______________________________________________
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Subject: Re: [Sip] Does retargeting belong on the list of Evil SIP Ideas?
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Dean,

Rather that the "list of Evil SIP Ideas", I propose this be called the 
"axis of Evil SIP Ideas".

	Paul

Dean Willis wrote:
> 
> I try to keep a running list of things  that turned out to be a royal 
> pain that needs (or needed) fixing. Some of these things sounded cool at 
> first, and I championed them.  Others I never did like. Some of these 
> things I'm not allowed to suggest deprecating anymore -- we took a hum 
> on forking a few meetings back. One of these days I'm going to get this 
> list better organized and  post it someplace more persistent in the hope 
> that it may assist in avoiding repeptition of mistakes. And someday, 
> perhaps we can do something about some of these quirks.
> 
> To-date, I believe the Evil List includes:
> 
> 1) Replacing the Request-URI at each hop while routing was Evil. We 
> fixed this with Loose Routing mode, which is still slightly Evil.
> 
> 2) Abusing To and From to indicate branches is Evil. We should have left 
> these as real endpoint identifiers.
> 
> 3) Forking is Evil.
> 
> 4)  Allowing (and requiring)  responses to carry a payload with more 
> meaning than "I acknowledge having received that request" is Evil. Since 
> we can't reject a response, we can't deal with big responses and can't 
> negotiate what they carry. Acknowledgement and response are different 
> beasties -- one is transport, the other is application, and we need to 
> understand the difference.
> 
> 5) The three-message INVITE transaction model and its dependence on 
> payload in the response is Evil.
> 
> 6) The two-message non-INVITE transaction model and its interaction with 
> retransmission and hop-by-hop cascade (See Sparks'  non-invite analysis) 
> is Evil.
> 
> 7) Building transport state and reliability into the application state 
> and requiring a transport protocol extension for every new application 
> semantic is Evil.
> 
> 8) Using z9hG4bK on via fields rather than a negotiated mechanism (like 
> maybe a version) is Evil.
> 
> 9) Jon argues that not requiring SIPS to be end-to-end is Evil. I think 
> the jury is still out on this one, but we're still deliberating.
> 
> 10) Provisional responses, as currently constructed, whether reliable or 
> non-relaible, are Evil. This may be redundant with the NIT discussion.
> 
> 11)  I believe I'm coming to believe that "retargeting",  i.e. changing 
> a request in a proxy such that the entity to which it is delivered  has 
> a different assertable identity than the identity to which the request 
> was originally addressed,  is Evil. Perhaps it could be mitigated by a 
> assertion of transitivity: "I, the proxy identified herein, do assert 
> that I have altered the target of this request according to local 
> policy, and you can either deal with it or go away." Oh wait, that's the 
> 302 response . . . Perhaps this is redundant with "Forking is evil" as 
> forking is just a special case of retargeting. Or perhaps forking and 
> retargeting are both special cases of "exploders" and could both be 
> dealt with using the consent framework . . .
> 
> Sometimes I wonder if  the use of UDP was fundamentally Evil.
> 
> If I've missed something blatantly Evil, please comment back, and I may 
> add it to the list I hope to post someplace.
> 
> _______________________________________________
> 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: David R Oran <oran@cisco.com>
Subject: Re: [Sip] Does retargeting belong on the list of Evil SIP Ideas?
Date: Tue, 23 Nov 2004 17:28:34 -0500
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Cc: "'sip@ietf.org'" <sip@ietf.org>, Dean Willis <dean.willis@softarmor.com>
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Boy do you guys ever have axis to grind!

On Nov 23, 2004, at 4:34 PM, Paul Kyzivat wrote:

> Dean,
>
> Rather that the "list of Evil SIP Ideas", I propose this be called the 
> "axis of Evil SIP Ideas".
>
> 	Paul
>
> Dean Willis wrote:
>> I try to keep a running list of things  that turned out to be a royal 
>> pain that needs (or needed) fixing. Some of these things sounded cool 
>> at first, and I championed them.  Others I never did like. Some of 
>> these things I'm not allowed to suggest deprecating anymore -- we 
>> took a hum on forking a few meetings back. One of these days I'm 
>> going to get this list better organized and  post it someplace more 
>> persistent in the hope that it may assist in avoiding repeptition of 
>> mistakes. And someday, perhaps we can do something about some of 
>> these quirks.
>> To-date, I believe the Evil List includes:
>> 1) Replacing the Request-URI at each hop while routing was Evil. We 
>> fixed this with Loose Routing mode, which is still slightly Evil.
>> 2) Abusing To and From to indicate branches is Evil. We should have 
>> left these as real endpoint identifiers.
>> 3) Forking is Evil.
>> 4)  Allowing (and requiring)  responses to carry a payload with more 
>> meaning than "I acknowledge having received that request" is Evil. 
>> Since we can't reject a response, we can't deal with big responses 
>> and can't negotiate what they carry. Acknowledgement and response are 
>> different beasties -- one is transport, the other is application, and 
>> we need to understand the difference.
>> 5) The three-message INVITE transaction model and its dependence on 
>> payload in the response is Evil.
>> 6) The two-message non-INVITE transaction model and its interaction 
>> with retransmission and hop-by-hop cascade (See Sparks'  non-invite 
>> analysis) is Evil.
>> 7) Building transport state and reliability into the application 
>> state and requiring a transport protocol extension for every new 
>> application semantic is Evil.
>> 8) Using z9hG4bK on via fields rather than a negotiated mechanism 
>> (like maybe a version) is Evil.
>> 9) Jon argues that not requiring SIPS to be end-to-end is Evil. I 
>> think the jury is still out on this one, but we're still 
>> deliberating.
>> 10) Provisional responses, as currently constructed, whether reliable 
>> or non-relaible, are Evil. This may be redundant with the NIT 
>> discussion.
>> 11)  I believe I'm coming to believe that "retargeting",  i.e. 
>> changing a request in a proxy such that the entity to which it is 
>> delivered  has a different assertable identity than the identity to 
>> which the request was originally addressed,  is Evil. Perhaps it 
>> could be mitigated by a assertion of transitivity: "I, the proxy 
>> identified herein, do assert that I have altered the target of this 
>> request according to local policy, and you can either deal with it or 
>> go away." Oh wait, that's the 302 response . . . Perhaps this is 
>> redundant with "Forking is evil" as forking is just a special case of 
>> retargeting. Or perhaps forking and retargeting are both special 
>> cases of "exploders" and could both be dealt with using the consent 
>> framework . . .
>> Sometimes I wonder if  the use of UDP was fundamentally Evil.
>> If I've missed something blatantly Evil, please comment back, and I 
>> may add it to the list I hope to post someplace.
>> _______________________________________________
>> 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
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.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: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'David R Oran'" <oran@cisco.com>
Subject: RE: concrete proposal (was RE: third alternative (was RE: [Sip] R
	E: I dentity after reinvite))
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> > An alternative argument has been made that the notion of an
unanticipated
> > 'connected party' is inherently antithetical to security, because the
local
> > UA will be forced to accept a request from any party, as the potential
set
> > of valid connected parties in unbounded.
> >
> This is true only in the absence of explicit delegation. 

Of course, and the only reason the text above does not say so is because I
was intentionally trying to phrase this in a way that is silent about the
whole 'connected party' issue.

As an aside, I think "explicit delegation" is a very good term for what we
need here, when we get around to addressing the 'connected party' problem.

> At the risk of 
> skipping ahead of your well-done chain of logic here, I think that we 
> have in SIP a large number of scenarios (retargeting, REFER, 
> Third-party control) where in the absence of explicit delegation we 
> either (a) have no security, (b) force cooperating parties to use 
> impersonation, or (c) in some cases, can rely on transitivity 
> properties. Despite the complexity, we may be better off tackling the 
> delegation problem now rather than allowing half-assed things to 
> use/abuse the identity stuff because it solves a part of the problem 
> (albeit a large part), but not enough.

Well... that really depends on your value of "now". I think we can do
request identity, frankly, without solving the explicit delegation problem.
When explicit delegation is resolved, then we can describe its relevance to
identity. In the interim, we can just say that the Identity header isn't
supplied for requests in the backwards direction during a dialog when
retargeting has occurred. Perhaps I only think this language makes sense
because I believe that when we get through with the explicit delegation
problem, we won't be doing "retargeting" any more as such.

> > Given that RFC3261 permits UAS's to change the From header field value
> > (irregardless of whether or not the sip-identity mechanism is used), the
> > question here is how should an authentication service, as defined in
> > sip-identity, treat such requests? Should it add an Identity header,
proxy
> > them, discard them, etc? Should sip-identity recommend that the From
header
> > be changed in this manner when a request has been retargeted?
>
> It would be nice if the identity service did not have to be context 
> aware and just signed identity headers for any request from a source it 
> was authoritative for. Unless I misunderstand you I think you are 
> contemplating the identity service being authz policy aware, and my 
> intuition tells me that's a bad idea.

At this point in the note, anyway, I'm just contemplating the sort of
questions that seem to have been asked in the thread, not endorsing any
solutions. I agree with you that ideally, the auth service should just sign
the requests that it can, and not sign the requests that it can't.

> >
> > We add some text about dialogs, and about the use of the Identity header
> > during a dialog. Obviously, the use of identity for dialog-forming
requests
> > is a no-brainer (this is the standard 'caller-ID' assurance).
Futhermore,
> > point out that if the intended second party is actually the connected
party,
> > identity can be supplied normally in requests in the backwards
direction.
> > Stipulate that an auth service instantiated in a proxy MUST Record-Route
> > itself in a dialog-forming request, so that the auth service can
recognize a
> > mid-dialog request.
> >
> I think I understand what you mean here, but the words could be better. 
> I think you are saying that "if a sip server acts as an auth service 
> and proxies a request that it auth'ed itself, it MUST record route".
> 

Yes.

> > The Identity header simply would not be provided if the connected party
is
> > different from the party to which the local UA sent the request.
> Why? If the auth service can authenticate the From in the request, why 
> should it refuse to do so?
> 

My language here is imprecise, yes, I mean something more like "the Identity
header simply would not be provided if the auth service is not responsible
for the domain of the From header field of the request." If we couple this
with the further notion that changing the From header field of requests in
the backwards direction within a dialog is NOT RECOMMENDED because it may
surprising to the originator of the dialog, I'm cool with that.

> > Anyone unhappy with that?
> >
> Modulo my one issue above, I'm ok with the identity part, but as I 
> mentioned, doing just this doesn't actually solve anything other than 
> getting identity done (which is laudable and necessary). 

Right, my ambitions are no greater than that at the moment. Once we're all
on the same page here, we can start diving into the explicit delegation
issue in more detail as a prelude to tackling response identity.

> I don't want 
> to slow identity down UNLESS there's reason to believe we will have 
> backward compatibility problems when we introduce delegation, OR we 
> think that identity will get abused or ignored because too many 
> real-world situations really need delegation.
> 

I'm not sure there's much potential for abuse here, and as long as we don't
rule out any important use cases, it faces no more than the usual
probability of being ignored. The most essential problem that people want to
solve is the dialog-forming request identity problem - essentially, the
"caller-ID" problem. That has immediate pay-off in the marketplace. As long
as we can tackle this in a way that doesn't damage our prospects of doing
more work on corner cases in the future, I think we'll be okay.

Jon Peterson
NeuStar, Inc.

> Dave.
> 
> David R. Oran
> Cisco Fellow
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720 USA
> Tel: +1 978 264 2048
> Email: oran@cisco.com
> 
> > Jon Peterson
> > NeuStar, Inc.
> 

_______________________________________________
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 is a multi-part message in MIME format.

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Hi,
=20
let me know if a following scenario is correct.
=20
Caller-A        Proxy-A      Proxy-B     Callee-B1          Callee-B2
       --INVITE-->
                      --INVITE-->
                                   --INVITE-->
                                                       (1)
                                              <--180 with To-tag1
                            <--180 with To-tag1
    <--180 with To-tag1
                                                       (2)
                                              <--302 with Contact :
Callee-B2
                                                 -----INVITE------->
            =20
                                        (3)
                              <--180 with To-tag2-------------------=20
      <--180 with To-tag2
=20
(1) Callee-B1's phone is ringing.
(2) Callee-B1 decides to forward this call to Callee-B2 without
accepting call. and send a 302 response with a Contact: Callee-B2
(3) Callee-B2's phone is ringing.
=20
As you see, at the above flow, Proxy-A and Caller-A receive two 180
responses with a different To-tag.  Caller-A created a early-dialog when
it received the first 180 response and then, it received the second 180
response with a difference To-tag from the first one. at that case, what
happens? should a early-dialog be updated with new To-tag?  or can't
Callee-B1 send a 302 response after sending a 180 response?
=20
Regards

Gautam Chaudhry

=20

------_=_NextPart_001_01C4D1DF.3639FBD4
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	charset="us-ascii"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2800.1106" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3D"Courier New"><FONT color=3D#000080><FONT =
size=3D2><SPAN=20
class=3D078513304-24112004>Hi,</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3D"Courier New"><FONT color=3D#000080><FONT =
size=3D2><SPAN=20
class=3D078513304-24112004></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New"><FONT color=3D#000080><FONT =
size=3D2><SPAN=20
class=3D078513304-24112004></SPAN>let me know if a following scenario is =

correct.<BR>&nbsp;<BR>Caller-A&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =

Proxy-A&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Proxy-B&nbsp;&nbsp;&nbsp;&nbsp;=20
Callee-B1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Callee-B2<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
--INVITE--&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
--INVITE--&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;=20
--INVITE--&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
(1)<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;--180 with=20
To-tag1<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;--180 with To-tag1<BR>&nbsp;&nbsp;&nbsp; &lt;--180 with=20
To-tag1<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
(2)<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;--302 with Contact :=20
Callee-B2<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;=20
-----INVITE-------&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;=20
(3)<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;--180 with To-tag2------------------- =
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;--180 with To-tag2<BR>&nbsp;<BR>(1) Callee-B1's phone is =
ringing.<BR>(2)=20
Callee-B1 decides to forward this call to Callee-B2 without accepting =
call. and=20
send a 302 response with a Contact: Callee-B2<BR>(3) Callee-B2's phone =
is=20
ringing.<BR>&nbsp;<BR>As you see, at the above flow, Proxy-A and =
Caller-A=20
receive two 180 responses with a different To-tag.&nbsp; Caller-A =
created a=20
early-dialog when it received the first 180 response and then, it =
received the=20
second 180 response with a difference To-tag from the first one. at that =
case,=20
what happens? should a early-dialog be updated with new To-tag?&nbsp; or =
can't=20
Callee-B1 send a 302 response after sending a 180=20
response?</FONT></FONT></FONT></DIV>
<DIV><FONT face=3D"Courier New" color=3D#000080 =
size=3D2></FONT>&nbsp;</DIV>
<DIV align=3Dleft>
<DIV class=3DSection1><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><SPAN=20
style=3D"FONT-SIZE: 8.5pt; FONT-FAMILY: 'Palatino Linotype'"><SPAN=20
style=3D"FONT-SIZE: 8.5pt; FONT-FAMILY: 'Palatino Linotype'"><FONT=20
color=3D#060606><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><SPAN=20
style=3D"FONT-SIZE: 8.5pt; FONT-FAMILY: 'Palatino Linotype'"><FONT=20
color=3D#000000><SPAN=20
style=3D"FONT-SIZE: 8.5pt; COLOR: gray; FONT-FAMILY: 'Book Antiqua'">
<P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt" align=3Dleft><SPAN=20
style=3D"FONT-SIZE: 8.5pt; COLOR: gray; FONT-FAMILY: 'Book =
Antiqua'"><EM><FONT=20
size=3D2><FONT face=3D"Courier New"><FONT =
color=3D#400080>Regards<?xml:namespace=20
prefix =3D o ns =3D "urn:schemas-microsoft-com:office:office"=20
/><o:p></o:p></FONT></FONT></FONT></EM></SPAN></P>
<P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
style=3D"FONT-SIZE: 8.5pt; COLOR: gray; FONT-FAMILY: 'Book =
Antiqua'"><EM><FONT=20
face=3D"Courier New" color=3D#400080 size=3D2>Gautam=20
Chaudhry</FONT></EM></SPAN></P></SPAN></FONT></SPAN></SPAN></FONT></SPAN>=
</SPAN></SPAN></DIV></DIV>
<DIV><FONT face=3D"Courier New" color=3D#000080=20
size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

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From: "Nataraju A B" <bnataraju@kodiaknetworks.com>
To: <gautam.chaudhry@wipro.com>, <sip@ietf.org>
Subject: RE: [Sip] Please Help
Date: Wed, 24 Nov 2004 12:01:51 +0530
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Best Regards,
Nataraju A.B.
-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
gautam.chaudhry@wipro.com
Sent: Wednesday, November 24, 2004 10:07 AM
To: sip@ietf.org
Subject: [Sip] Please Help
 
Hi,
 
let me know if a following scenario is correct.
 
Caller-A        Proxy-A      Proxy-B     Callee-B1          Callee-B2
       --INVITE-->
                      --INVITE-->
                                   --INVITE-->
                                                       (1)
                                              <--180 with To-tag1
                            <--180 with To-tag1
    <--180 with To-tag1
                                                       (2)
                                              <--302 with Contact :
Callee-B2
                                                 -----INVITE------->
             
                                        (3)
                              <--180 with To-tag2------------------- 
      <--180 with To-tag2
 
(1) Callee-B1's phone is ringing.
(2) Callee-B1 decides to forward this call to Callee-B2 without
accepting call. and send a 302 response with a Contact: Callee-B2
(3) Callee-B2's phone is ringing.
 
As you see, at the above flow, Proxy-A and Caller-A receive two 180
responses with a different To-tag.  Caller-A created a early-dialog when
it received the first 180 response and then, it received the second 180
response with a difference To-tag from the first one. at that case, what
happens? should a early-dialog be updated with new To-tag?  or can't
Callee-B1 send a 302 response after sending a 180 response?
 
 [ABN] if callee-B1 is retrying on 302 then 302 must not be sent to the
caller. Only the response for the retried invite will be sent to caller.
Caller must be prepared to receive responses with diff to-tags which
lead to multiple early dialogs as part of the session. Then its SIP
application logic to retain one of the confirmed dialogs. This is one of
the major change from 2543 to 3261 implementation.
 
 
Regards
Gautam Chaudhry
 

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<p class=3DMsoNormal><em><i><font size=3D3 color=3Dblue face=3D"Book =
Antiqua"><span
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Antiqua";color:blue;mso-no-proof:
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<p class=3DMsoNormal><em><i><font size=3D3 color=3Dblue face=3D"Book =
Antiqua"><span
style=3D'font-size:12.0pt;font-family:"Book =
Antiqua";color:blue;mso-no-proof:
yes'>Nataraju A.B.</span></font></i></em><o:p></o:p></p>

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<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'>-----Original =
Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> =
sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org] <b><span style=3D'font-weight:bold'>On =
Behalf Of </span></b>gautam.chaudhry@wipro.com<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> =
</span></font><st1:date
Month=3D"11" Day=3D"24" Year=3D"2004"><font size=3D2 face=3DTahoma><span
 style=3D'font-size:10.0pt;font-family:Tahoma'>Wednesday, November 24, =
2004</span></font></st1:date><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> </span></font><st1:time
Hour=3D"10" Minute=3D"7"><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
 font-family:Tahoma'>10:07 AM</span></font></st1:time><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'><br>
<b><span style=3D'font-weight:bold'>To:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] Please =
Help</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dnavy
face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:navy'>Hi,</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dnavy
face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:navy'>let me know if a following scenario is correct.<br>
&nbsp;<br>
Caller-A&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Proxy-A&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Proxy-B&nbsp;&nbsp;&nbsp;&nbsp;
Callee-B1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Callee-B2<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --INVITE--&gt;<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
--INVITE--&gt;<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
--INVITE--&gt;<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
(1)<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&lt;--180 with To-tag1<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;
&lt;--180 with To-tag1<br>
&nbsp;&nbsp;&nbsp; &lt;--180 with To-tag1<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
(2)<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&lt;--302 with Contact : Callee-B2<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
-----INVITE-------&gt;<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;
(3)<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;
&lt;--180 with To-tag2------------------- <br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;--180 with To-tag2<br>
&nbsp;<br>
(1) Callee-B1's phone is ringing.<br>
(2) Callee-B1 decides to forward this call to Callee-B2 without =
accepting call.
and send a 302 response with a Contact: Callee-B2<br>
(3) Callee-B2's phone is ringing.<br>
&nbsp;<br>
As you see, at the above flow, Proxy-A and Caller-A receive two 180 =
responses
with a different To-tag.&nbsp; Caller-A created a early-dialog when it =
received
the first 180 response and then, it received the second 180 response =
with a
difference To-tag from the first one. at that case, what happens? should =
a
early-dialog be updated with new To-tag?&nbsp; or can't Callee-B1 send a =
302
response after sending a 180 response?</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
color=3Dblue
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt;color:blue'><o:p>&nbsp;</o:p></span></font></p>=


<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;<b =
style=3D'mso-bidi-font-weight:normal'><i
style=3D'mso-bidi-font-style:normal'><font color=3Dblue><span =
style=3D'color:blue;
font-weight:bold;mso-bidi-font-weight:normal;font-style:italic;mso-bidi-f=
ont-style:
normal'>[ABN] </span></font></i></b><i =
style=3D'mso-bidi-font-style:normal'><font
color=3Dblue><span =
style=3D'color:blue;font-style:italic;mso-bidi-font-style:normal'>if
callee-B1 is retrying on 302 then 302 must not be sent to the caller. =
Only the response
for the retried invite will be sent to =
caller.<o:p></o:p></span></font></i></span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><i =
style=3D'mso-bidi-font-style:normal'><font
size=3D3 color=3Dblue face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt;
color:blue;font-style:italic;mso-bidi-font-style:normal'>Caller must be
prepared to receive responses with diff to-tags which lead to multiple =
early dialogs
as part of the session. <span class=3DGramE>Then its SIP application =
logic to
retain one of the confirmed dialogs.</span> This is one of the major =
<span
class=3DGramE>change</span> from 2543 to 3261 =
implementation&#8230;<o:p></o:p></span></font></i></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><b =
style=3D'mso-bidi-font-weight:
normal'><i style=3D'mso-bidi-font-style:normal'><font size=3D3 =
color=3Dblue
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt;color:blue;font-weight:
bold;mso-bidi-font-weight:normal;font-style:italic;mso-bidi-font-style:no=
rmal'><o:p>&nbsp;</o:p></span></font></i></b></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><b =
style=3D'mso-bidi-font-weight:
normal'><i style=3D'mso-bidi-font-style:normal'><font size=3D3 =
color=3Dblue
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt;color:blue;font-weight:
bold;mso-bidi-font-weight:normal;font-style:italic;mso-bidi-font-style:no=
rmal'><span
style=3D'mso-spacerun:yes'>&nbsp;</span></span></font></i></b><o:p></o:p>=
</p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><em><i><font size=3D2 =
color=3D"#400080"
face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:#400080'>Regards<o:p></o:p></span></font></i></em></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><span =
class=3DSpellE><em><i><font
size=3D2 color=3D"#400080" face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier =
New";color:#400080'>Gautam</span></font></i></em></span><em><i><font
size=3D2 color=3D"#400080" face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New";color:#400080'> =
Chaudhry</span></font></i></em><font
color=3Dgray><span style=3D'color:gray'><o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

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Robert Sparks wrote:

> This conversation just got too loose with its terminology for me.
> 
> 3261 proxies have a UAS core.

I don't think so; the term for that is "proxy core" and is quite 
distinct from the UAS and UAC cores. Note the definition of TU:

  Transaction User (TU): The layer of protocol processing that
          resides above the transaction layer.  Transaction users include
          the UAC core, UAS core, and proxy core.

3261 proxies have a server transaction (per figure 4) but thats not the 
same thing.



> 
> Endpoints behind the things we are wanting to do this refresh things to 
> can be signaling
> directly with endpoints that aren't (a service provider could let people 
> talk to voice-mail
> or conference servers directly).

I don't see how that practically works in the presence of nat. If you 
are using tcp/tls, the thing to which you have a connection open is the 
only place from which you can get subsequent requests. Thus, if I make a 
call that lands on a voicemail server, the proxy holding my tcp/tls 
connection has to be on the signaling path, or it just won't work.

> 
> While the long-lived client-to-service-provider-proxy connection is a 
> good example
> to use for motivation, I would prefer that we ensure we build something 
> that works
> in general.

I disagree here. The original connect-reuse draft tried to address both 
the client-to-proxy and proxy-to-proxy problems and ended up doing 
neither well. We found we needed to split it because these were 
sufficiently different problems. It seems like you are suggsting that 
they are not, and that we need a single mechanism for both?

> 
> The DNS solution would work for those the services I call out above.  
> I'm not yet sure
> if its the right way to go.

The other alternative I suppose is to use a URI parameter ala 
draft-ietf-sip-compression. Something like keepalive=stun. Thus, a 
preloaded route header for a proxy that supports it would look like:

sip:outbound.proxy.com;keepalive=stun

This is arguably superior for exactly the same reasons we shied away 
from putting sigcomp into dns - you get this combinatorial explosion of 
records, since sigcomp and stun as well could both be used in any transport.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Actually, Dean's list of evils is sufficiently large, and covers enough 
of sip itself, that I think he has more or less concluded that "sip is 
evil"? :)


-Jonathan R

Paul Kyzivat wrote:

> Dean,
> 
> Rather that the "list of Evil SIP Ideas", I propose this be called the 
> "axis of Evil SIP Ideas".
> 
>     Paul
> 
> Dean Willis wrote:
> 
>>
>> I try to keep a running list of things  that turned out to be a royal 
>> pain that needs (or needed) fixing. Some of these things sounded cool 
>> at first, and I championed them.  Others I never did like. Some of 
>> these things I'm not allowed to suggest deprecating anymore -- we took 
>> a hum on forking a few meetings back. One of these days I'm going to 
>> get this list better organized and  post it someplace more persistent 
>> in the hope that it may assist in avoiding repeptition of mistakes. 
>> And someday, perhaps we can do something about some of these quirks.
>>
>> To-date, I believe the Evil List includes:
>>
>> 1) Replacing the Request-URI at each hop while routing was Evil. We 
>> fixed this with Loose Routing mode, which is still slightly Evil.
>>
>> 2) Abusing To and From to indicate branches is Evil. We should have 
>> left these as real endpoint identifiers.
>>
>> 3) Forking is Evil.
>>
>> 4)  Allowing (and requiring)  responses to carry a payload with more 
>> meaning than "I acknowledge having received that request" is Evil. 
>> Since we can't reject a response, we can't deal with big responses and 
>> can't negotiate what they carry. Acknowledgement and response are 
>> different beasties -- one is transport, the other is application, and 
>> we need to understand the difference.
>>
>> 5) The three-message INVITE transaction model and its dependence on 
>> payload in the response is Evil.
>>
>> 6) The two-message non-INVITE transaction model and its interaction 
>> with retransmission and hop-by-hop cascade (See Sparks'  non-invite 
>> analysis) is Evil.
>>
>> 7) Building transport state and reliability into the application state 
>> and requiring a transport protocol extension for every new application 
>> semantic is Evil.
>>
>> 8) Using z9hG4bK on via fields rather than a negotiated mechanism 
>> (like maybe a version) is Evil.
>>
>> 9) Jon argues that not requiring SIPS to be end-to-end is Evil. I 
>> think the jury is still out on this one, but we're still deliberating.
>>
>> 10) Provisional responses, as currently constructed, whether reliable 
>> or non-relaible, are Evil. This may be redundant with the NIT discussion.
>>
>> 11)  I believe I'm coming to believe that "retargeting",  i.e. 
>> changing a request in a proxy such that the entity to which it is 
>> delivered  has a different assertable identity than the identity to 
>> which the request was originally addressed,  is Evil. Perhaps it could 
>> be mitigated by a assertion of transitivity: "I, the proxy identified 
>> herein, do assert that I have altered the target of this request 
>> according to local policy, and you can either deal with it or go 
>> away." Oh wait, that's the 302 response . . . Perhaps this is 
>> redundant with "Forking is evil" as forking is just a special case of 
>> retargeting. Or perhaps forking and retargeting are both special cases 
>> of "exploders" and could both be dealt with using the consent 
>> framework . . .
>>
>> Sometimes I wonder if  the use of UDP was fundamentally Evil.
>>
>> If I've missed something blatantly Evil, please comment back, and I 
>> may add it to the list I hope to post someplace.
>>
>> _______________________________________________
>> 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
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

_______________________________________________
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Firstly, I support Jon's modified proposal. I'm now convinced that 
providing authenticated identity in the reverse direction doesn't 
mitigate against any attacks that can't be prevented by other, better 
means. Indeed, I am less sour on sips than Jon, and using sips fully 
prevents the impersonation attacks which Jon and Dave were discussing.

I also must say that I shudder when someone says "we'll solve the 
delegation problem" to solve response identity. That problem, in its 
most general sense, is incredibly hard. I think our only hope is to 
focus on the very specific cases we need to worry about. In fact, I 
suspect the solutions for our various cases will differ. I really don't 
see the response identity and transfer/REFER problems as similar 
delegation ones, primarily because the authorization logic is different. 
But, perhaps I am missing a commonality that is more apparent to others...

-Jonathan R.


Peterson, Jon wrote:

>>>An alternative argument has been made that the notion of an
> 
> unanticipated
> 
>>>'connected party' is inherently antithetical to security, because the
> 
> local
> 
>>>UA will be forced to accept a request from any party, as the potential
> 
> set
> 
>>>of valid connected parties in unbounded.
>>>
>>
>>This is true only in the absence of explicit delegation. 
> 
> 
> Of course, and the only reason the text above does not say so is because I
> was intentionally trying to phrase this in a way that is silent about the
> whole 'connected party' issue.
> 
> As an aside, I think "explicit delegation" is a very good term for what we
> need here, when we get around to addressing the 'connected party' problem.
> 
> 
>>At the risk of 
>>skipping ahead of your well-done chain of logic here, I think that we 
>>have in SIP a large number of scenarios (retargeting, REFER, 
>>Third-party control) where in the absence of explicit delegation we 
>>either (a) have no security, (b) force cooperating parties to use 
>>impersonation, or (c) in some cases, can rely on transitivity 
>>properties. Despite the complexity, we may be better off tackling the 
>>delegation problem now rather than allowing half-assed things to 
>>use/abuse the identity stuff because it solves a part of the problem 
>>(albeit a large part), but not enough.
> 
> 
> Well... that really depends on your value of "now". I think we can do
> request identity, frankly, without solving the explicit delegation problem.
> When explicit delegation is resolved, then we can describe its relevance to
> identity. In the interim, we can just say that the Identity header isn't
> supplied for requests in the backwards direction during a dialog when
> retargeting has occurred. Perhaps I only think this language makes sense
> because I believe that when we get through with the explicit delegation
> problem, we won't be doing "retargeting" any more as such.
> 
> 
>>>Given that RFC3261 permits UAS's to change the From header field value
>>>(irregardless of whether or not the sip-identity mechanism is used), the
>>>question here is how should an authentication service, as defined in
>>>sip-identity, treat such requests? Should it add an Identity header,
> 
> proxy
> 
>>>them, discard them, etc? Should sip-identity recommend that the From
> 
> header
> 
>>>be changed in this manner when a request has been retargeted?
>>
>>It would be nice if the identity service did not have to be context 
>>aware and just signed identity headers for any request from a source it 
>>was authoritative for. Unless I misunderstand you I think you are 
>>contemplating the identity service being authz policy aware, and my 
>>intuition tells me that's a bad idea.
> 
> 
> At this point in the note, anyway, I'm just contemplating the sort of
> questions that seem to have been asked in the thread, not endorsing any
> solutions. I agree with you that ideally, the auth service should just sign
> the requests that it can, and not sign the requests that it can't.
> 
> 
>>>We add some text about dialogs, and about the use of the Identity header
>>>during a dialog. Obviously, the use of identity for dialog-forming
> 
> requests
> 
>>>is a no-brainer (this is the standard 'caller-ID' assurance).
> 
> Futhermore,
> 
>>>point out that if the intended second party is actually the connected
> 
> party,
> 
>>>identity can be supplied normally in requests in the backwards
> 
> direction.
> 
>>>Stipulate that an auth service instantiated in a proxy MUST Record-Route
>>>itself in a dialog-forming request, so that the auth service can
> 
> recognize a
> 
>>>mid-dialog request.
>>>
>>
>>I think I understand what you mean here, but the words could be better. 
>>I think you are saying that "if a sip server acts as an auth service 
>>and proxies a request that it auth'ed itself, it MUST record route".
>>
> 
> 
> Yes.
> 
> 
>>>The Identity header simply would not be provided if the connected party
> 
> is
> 
>>>different from the party to which the local UA sent the request.
>>
>>Why? If the auth service can authenticate the From in the request, why 
>>should it refuse to do so?
>>
> 
> 
> My language here is imprecise, yes, I mean something more like "the Identity
> header simply would not be provided if the auth service is not responsible
> for the domain of the From header field of the request." If we couple this
> with the further notion that changing the From header field of requests in
> the backwards direction within a dialog is NOT RECOMMENDED because it may
> surprising to the originator of the dialog, I'm cool with that.
> 
> 
>>>Anyone unhappy with that?
>>>
>>
>>Modulo my one issue above, I'm ok with the identity part, but as I 
>>mentioned, doing just this doesn't actually solve anything other than 
>>getting identity done (which is laudable and necessary). 
> 
> 
> Right, my ambitions are no greater than that at the moment. Once we're all
> on the same page here, we can start diving into the explicit delegation
> issue in more detail as a prelude to tackling response identity.
> 
> 
>>I don't want 
>>to slow identity down UNLESS there's reason to believe we will have 
>>backward compatibility problems when we introduce delegation, OR we 
>>think that identity will get abused or ignored because too many 
>>real-world situations really need delegation.
>>
> 
> 
> I'm not sure there's much potential for abuse here, and as long as we don't
> rule out any important use cases, it faces no more than the usual
> probability of being ignored. The most essential problem that people want to
> solve is the dialog-forming request identity problem - essentially, the
> "caller-ID" problem. That has immediate pay-off in the marketplace. As long
> as we can tackle this in a way that doesn't damage our prospects of doing
> more work on corner cases in the future, I think we'll be okay.
> 
> Jon Peterson
> NeuStar, Inc.
> 
> 
>>Dave.
>>
>>David R. Oran
>>Cisco Fellow
>>Cisco Systems
>>7 Ladyslipper Lane
>>Acton, MA 01720 USA
>>Tel: +1 978 264 2048
>>Email: oran@cisco.com
>>
>>
>>>Jon Peterson
>>>NeuStar, Inc.
>>
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

_______________________________________________
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Peterson, Jon wrote:

> A few notes on the "Monica property": as RFC3323 suggests, I think that some
> sort of anonymization service, which could be instantiated by a B2BUA, is an
> appropriate way to conceal the identity of a called or calling party. What
> scares me a little is the way people suggest that retargeting, as opposed to
> redirection, can inherently be expected to provide privacy. When I register
> a contact, I can't always guarantee that the location service in question
> will be used exclusively for retargeting rather than redirection; moreover,
> the revelation of the target Request-URI to the caller is not the only way
> that the caller can learn who the connected-party is: contact headers of new
> requests in the backwards direction of the dialog, SDP in a 200 OK, and so
> on, are information leaks that are not addressed at all by retargeting, but
> would be addressed by an anonymization service.
> 
> RFC3323 is about due for an update, I think, because the major functions
> that it provides can be performed without a B2BUA, thanks to new SIP
> mechanisms like GRUUs and session-policy. So while I agree that an
> anonymization service is the right approach to this problem, these days I
> think you can probably build an anonymization service without building a
> B2BUA.

I believe this is probably true. Here's how I think it'd work:

A client that wants to make an anonymous call contacts a TURN server 
providing anonymization services. That server provides the client with 
IP addresses and ports that it can use in its Via, contact, and SDP. The 
 From field is anonymized. A privacy header is inserted. The request is 
sent via the TURN server to the user's outbound proxy. That proxy can 
still authenticate the client, and even insert an identity header that 
at least vouches that the client is a legit client that is just anonymous.

The provider of the anonymization server (turn in this case) can be 
completely decoupled from the sip provider. That provides an excellent 
level of anonymity compared to even a b2bua. The b2bua approach would 
still allow the recipient to trace the IP address in the SDP and Contact 
to the particular provider; this might not be sufficient (I don't recall 
off hand if rfc3323 discusses this problme).

It also means that there is no need to signal things like header or 
media level privacy. The client simply chooses to obfuscate (using the 
learned turn addresses) whatever fields it wishes to anonymize. This has 
the impact of simplifying the privacy spec significantly. Indeed, it 
would seem that it is hardly needed at all...

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Jonathan - inline.

	Paul

Jonathan Rosenberg wrote:
> 
>> RFC3323 is about due for an update, I think, because the major functions
>> that it provides can be performed without a B2BUA, thanks to new SIP
>> mechanisms like GRUUs and session-policy. So while I agree that an
>> anonymization service is the right approach to this problem, these days I
>> think you can probably build an anonymization service without building a
>> B2BUA.
> 
> I believe this is probably true. Here's how I think it'd work:
> 
> A client that wants to make an anonymous call contacts a TURN server 
> providing anonymization services. That server provides the client with 
> IP addresses and ports that it can use in its Via, contact, and SDP. The 
> From field is anonymized. A privacy header is inserted. The request is 
> sent via the TURN server to the user's outbound proxy. That proxy can 
> still authenticate the client, and even insert an identity header that 
> at least vouches that the client is a legit client that is just anonymous.

While I buy the approach, I don't think what you are describing is a 
TURN server. It is a combination of a TURN server and a SIP 
anonymization server. And I don't even see how this can be viewed as one 
thing. The client has to contact a TURN server from the address/port it 
intends to use for media. For multiple media streams it needs to do this 
multiple times from different source ports. It also has to contact the 
sip anonymization server to set up the sip anonymization. Then it can 
proceed as you describe.

Still seems like a good approach - but it requires additional servers / 
protocols, and extra work on the part of the client. I can see how some 
might find it easier to just send a sip request to a server that does 
all the work by magic.

	Paul

> The provider of the anonymization server (turn in this case) can be 
> completely decoupled from the sip provider. That provides an excellent 
> level of anonymity compared to even a b2bua. The b2bua approach would 
> still allow the recipient to trace the IP address in the SDP and Contact 
> to the particular provider; this might not be sufficient (I don't recall 
> off hand if rfc3323 discusses this problme).
> 
> It also means that there is no need to signal things like header or 
> media level privacy. The client simply chooses to obfuscate (using the 
> learned turn addresses) whatever fields it wishes to anonymize. This has 
> the impact of simplifying the privacy spec significantly. Indeed, it 
> would seem that it is hardly needed at all...




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Use sipping@ietf.org for new developments on the application of sip


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On Nov 24, 2004, at 1:54 AM, Jonathan Rosenberg wrote:

>
>
> Robert Sparks wrote:
>
>> This conversation just got too loose with its terminology for me.
>> 3261 proxies have a UAS core.
>
> I don't think so; the term for that is "proxy core" and is quite 
> distinct from the UAS and UAC cores. Note the definition of TU:
>
>  Transaction User (TU): The layer of protocol processing that
>          resides above the transaction layer.  Transaction users 
> include
>          the UAC core, UAS core, and proxy core.
>
> 3261 proxies have a server transaction (per figure 4) but thats not 
> the same thing.

Fair enough - I was thinking of the role-switch proxies take on error 
responses, but
I'll withdraw the complaint.

>
>
>
>> Endpoints behind the things we are wanting to do this refresh things 
>> to can be signaling
>> directly with endpoints that aren't (a service provider could let 
>> people talk to voice-mail
>> or conference servers directly).
>
> I don't see how that practically works in the presence of nat. If you 
> are using tcp/tls, the thing to which you have a connection open is 
> the only place from which you can get subsequent requests. Thus, if I 
> make a call that lands on a voicemail server, the proxy holding my 
> tcp/tls connection has to be on the signaling path, or it just won't 
> work.

If I can establish a tcp/tls connection to that voicemail server, then 
subsequent requests from it can
come back down that connection assuming we can keep it alive.

>
>> While the long-lived client-to-service-provider-proxy connection is a 
>> good example
>> to use for motivation, I would prefer that we ensure we build 
>> something that works
>> in general.
>
> I disagree here. The original connect-reuse draft tried to address 
> both the client-to-proxy and proxy-to-proxy problems and ended up 
> doing neither well. We found we needed to split it because these were 
> sufficiently different problems. It seems like you are suggsting that 
> they are not, and that we need a single mechanism for both?
No, I'm not trying to push this back to arbitrary node to arbitrary 
node. I am trying to push it closer to UAC to arbitrary node.
And I'm only talking about keeping the connection alive, which is a 
different problem than figuring out what SIP thing is
on each end so you can affect routing choices (which is a big part of 
what's been making connection-reuse hard).
>
>> The DNS solution would work for those the services I call out above.  
>> I'm not yet sure
>> if its the right way to go.
>
> The other alternative I suppose is to use a URI parameter ala 
> draft-ietf-sip-compression. Something like keepalive=stun. Thus, a 
> preloaded route header for a proxy that supports it would look like:
>
> sip:outbound.proxy.com;keepalive=stun
>
> This is arguably superior for exactly the same reasons we shied away 
> from putting sigcomp into dns - you get this combinatorial explosion 
> of records, since sigcomp and stun as well could both be used in any 
> transport.

Agreed.
>
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ 
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com


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From: Adam Roach <adam@nostrum.com>
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Subject: [Sip] Event List Changes
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I'm revising the event list draft currently. In DC, we discussed at 
length the authentication model for situations in which the RLS and 
subscriber are in the same domain; I believe I've captured this pretty 
well (but encourage interested parties to double check my interpretation 
of consensus).

When I went to update the document, however, I found that there was 
still something of a hole for the other case -- the one we described as 
a corner case. I know that we agreed that (1) additional mechanism could 
be developed to solve this case, and (2) we didn't want to wait for 
these mechanisms before publishing the event list document. However, I 
don't remember discussing what RLS implementors can do when they receive 
a request for a list resource from a user outside their domain.

So, I came up with what seemed like the most logical solution, given the 
other discussions we had. Before I submit -07 of the document, I'd like 
to have a few people look over my proposed text; if you think this is 
the wrong direction to take it, speak up now. If I get no comments 
before the end of next week, I'll submit the draft with the following 
language.

> 7.1 Authentication
>
>    If back-end subscriptions are required to retrieve resource state
>    information, the end user is no longer the direct subscriber to the
>    state of the resource.  This means that direct authentication of the
>    user is no longer possible.
>
> 7.1.1 RLS and Subscriber in the Same Domain
>
>    It is expected that the most common deployment of RLSes entails the
>    subscribers to the RLS being in the same domain as the RLS.  When
>    this is the case, the RLS then has the ability to act as an
>    authenication service.  The role of authentication service is defined
>    in "Enhancements for Authenticated Identity Management in the Session
>    Initiation Protocol (SIP)" [7].
>
>    At a high level, under this system, the RLS authenticates the
>    subscriber, and then includes an "Identity" header field in all of
>    the back-end subscriptions performed on behalf of that authenticated
>    user; this "Identity" header field cryptographically asserts that the
>    request has been authorized to be made on behalf of the user
>    indicated in the "From" header field.
>
>    Because the ability to authenticate requests is central to the proper
>    functioning of the network, any RLS which uses SIP back-end
>    subscriptions to acquire information about the resources in a
>    resource list MUST be able to act as an authentication service as
>    defined in [7].
>
> 7.1.2 RLS and Subscriber in Different Domains
>
>    In the general case, the SIP Authenticated Identity extensions do not
>    provide a means for the RLS to securely assert that subscriptions are
>    being performed on the end user's behalf.  Specifically, when the
>    subscriber and the RLS are in different domains, the RLS will have no
>    means by which it can vouch for the user's identity.  Mechanisms by
>    which back-end subscriptions in such circumstances can be
>    authenticated are left for future study.
>
>    Until such general solutions are developed, RLSes which are in a
>    different domain than the subscriber on whose behalf they are
>    creating back-end susbcriptions SHOULD subscribe to the resources
>    using their own identity.  By doing so, the RLS will generally obtain
>    only the resource information which is made publicly available.
>
>    Absent such general solutions, implemenations of subscriber user
>    agents MAY attempt direct subscriptions to resources in the resouce
>    list when subscribing to an RLS outside of their domain (either
>    directly or by way of another resource list subscription).  The
>    resouces to be subscribed to will be those indicated in the the "uri"
>    attribute of the <resource> elements present in the RLMI document
>    returned by the RLS.  Directly subscribing to the resources allows
>    proper authentication of the user to take place, which will generally
>    authorize them to receive more complete state information.
>    Implementations which choose to perform such direct subscriptions
>    SHOULD use the data retrieved directly to the exclusion of any
>    information about the resource obtained via the list subscription.


/a

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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org  Wed Nov 24 22:25:49 2004
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From: Adam Roach <adam@nostrum.com>
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Subject: Re: [Sip] Re: RLS and identity
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Adam Roach wrote:

> Based on Rohan's suggestion, the text will effectively say:
>
> - Jon's Identity draft will be mandatory to implement, optional to use.
>
> - Other mechanisms that have properties such that they can adequately
>   convey the identity of the subscriber and the permission of the RLS
>   to subscribe on the user's behalf can also be used.

As a clarification, I have received specific guidance from the area 
directors that the draft cannot contain anything about such alternate 
mechanisms, even if they are not specifically mentioned by name. The use 
of e.g. P-Asserted-Identity will need to be a modification that 3GPP 
specifically calls out relative to the draft.

Of course, since the identity work is basically done and works so much 
better than P-Asserted-Identity, there could be a very valid argument 
made that 3GPP R6 should abandon the inherently insecure 
P-Asserted-Identity mechanism in favor of the cryptographically secure 
SIP Identity mechanism.

/a

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From sip-bounces@ietf.org  Thu Nov 25 03:17:22 2004
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From: Milinski Alexander <alexander.milinski@siemens.com>
To: "'Adam Roach'" <adam@nostrum.com>
Subject: RE: [Sip] Re: RLS and identity
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Dear Adam, 
3GPP Rel-6 is close to freezing, thus I would not expect any more major changes. Also, it seems that backward compatibility would need to be addressed.
In other words: I believe your proposal is not realistic.
Regards,
Alexander

P.S. Once Dean has cleaned up all SIP evils, there will certainly be the opportunity to consider also other major changes ... :-)

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Adam Roach
Sent: Thursday, November 25, 2004 4:18 AM
To: Adam Roach
Cc: SIP WG; Aki Niemi
Subject: Re: [Sip] Re: RLS and identity


Adam Roach wrote:

> Based on Rohan's suggestion, the text will effectively say:
>
> - Jon's Identity draft will be mandatory to implement, optional to use.
>
> - Other mechanisms that have properties such that they can adequately
>   convey the identity of the subscriber and the permission of the RLS
>   to subscribe on the user's behalf can also be used.

As a clarification, I have received specific guidance from the area 
directors that the draft cannot contain anything about such alternate 
mechanisms, even if they are not specifically mentioned by name. The use 
of e.g. P-Asserted-Identity will need to be a modification that 3GPP 
specifically calls out relative to the draft.

Of course, since the identity work is basically done and works so much 
better than P-Asserted-Identity, there could be a very valid argument 
made that 3GPP R6 should abandon the inherently insecure 
P-Asserted-Identity mechanism in favor of the cryptographically secure 
SIP Identity mechanism.

/a

_______________________________________________
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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From sip-bounces@ietf.org  Thu Nov 25 13:28:26 2004
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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Event List Changes
Date: Thu, 25 Nov 2004 19:22:28 +0100
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What should an RLS do when it receives a list subscription from a 
different domain it has no trust relationship with, and that 
subscription has an identity-header? I think it should remove that 
header. Is this an identity draft issue or should the event list draft 
also mention it?

Regards,
Hisham

On Nov 25, 2004, at 4:08 AM, Adam Roach wrote:

> I'm revising the event list draft currently. In DC, we discussed at 
> length the authentication model for situations in which the RLS and 
> subscriber are in the same domain; I believe I've captured this pretty 
> well (but encourage interested parties to double check my 
> interpretation of consensus).
>
> When I went to update the document, however, I found that there was 
> still something of a hole for the other case -- the one we described 
> as a corner case. I know that we agreed that (1) additional mechanism 
> could be developed to solve this case, and (2) we didn't want to wait 
> for these mechanisms before publishing the event list document. 
> However, I don't remember discussing what RLS implementors can do when 
> they receive a request for a list resource from a user outside their 
> domain.
>
> So, I came up with what seemed like the most logical solution, given 
> the other discussions we had. Before I submit -07 of the document, I'd 
> like to have a few people look over my proposed text; if you think 
> this is the wrong direction to take it, speak up now. If I get no 
> comments before the end of next week, I'll submit the draft with the 
> following language.
>
>> 7.1 Authentication
>>
>>    If back-end subscriptions are required to retrieve resource state
>>    information, the end user is no longer the direct subscriber to the
>>    state of the resource.  This means that direct authentication of 
>> the
>>    user is no longer possible.
>>
>> 7.1.1 RLS and Subscriber in the Same Domain
>>
>>    It is expected that the most common deployment of RLSes entails the
>>    subscribers to the RLS being in the same domain as the RLS.  When
>>    this is the case, the RLS then has the ability to act as an
>>    authenication service.  The role of authentication service is 
>> defined
>>    in "Enhancements for Authenticated Identity Management in the 
>> Session
>>    Initiation Protocol (SIP)" [7].
>>
>>    At a high level, under this system, the RLS authenticates the
>>    subscriber, and then includes an "Identity" header field in all of
>>    the back-end subscriptions performed on behalf of that 
>> authenticated
>>    user; this "Identity" header field cryptographically asserts that 
>> the
>>    request has been authorized to be made on behalf of the user
>>    indicated in the "From" header field.
>>
>>    Because the ability to authenticate requests is central to the 
>> proper
>>    functioning of the network, any RLS which uses SIP back-end
>>    subscriptions to acquire information about the resources in a
>>    resource list MUST be able to act as an authentication service as
>>    defined in [7].
>>
>> 7.1.2 RLS and Subscriber in Different Domains
>>
>>    In the general case, the SIP Authenticated Identity extensions do 
>> not
>>    provide a means for the RLS to securely assert that subscriptions 
>> are
>>    being performed on the end user's behalf.  Specifically, when the
>>    subscriber and the RLS are in different domains, the RLS will have 
>> no
>>    means by which it can vouch for the user's identity.  Mechanisms by
>>    which back-end subscriptions in such circumstances can be
>>    authenticated are left for future study.
>>
>>    Until such general solutions are developed, RLSes which are in a
>>    different domain than the subscriber on whose behalf they are
>>    creating back-end susbcriptions SHOULD subscribe to the resources
>>    using their own identity.  By doing so, the RLS will generally 
>> obtain
>>    only the resource information which is made publicly available.
>>
>>    Absent such general solutions, implemenations of subscriber user
>>    agents MAY attempt direct subscriptions to resources in the resouce
>>    list when subscribing to an RLS outside of their domain (either
>>    directly or by way of another resource list subscription).  The
>>    resouces to be subscribed to will be those indicated in the the 
>> "uri"
>>    attribute of the <resource> elements present in the RLMI document
>>    returned by the RLS.  Directly subscribing to the resources allows
>>    proper authentication of the user to take place, which will 
>> generally
>>    authorize them to receive more complete state information.
>>    Implementations which choose to perform such direct subscriptions
>>    SHOULD use the data retrieved directly to the exclusion of any
>>    information about the resource obtained via the list subscription.
>
>
> /a
>
> _______________________________________________
> 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 sip-bounces@ietf.org  Fri Nov 26 00:05:59 2004
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From: "Srinivasa Rao" <Srinivasa.Rao@alcatel.com>
To: <sip@ietf.org>
Subject: [SIP] RFC 3725, clarification
Date: Fri, 26 Nov 2004 10:34:32 +0530
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Hi,
   Anybody please clarify me about the difference of
   point 1 : "Offers and answers that contain a connection line
         with an address of 0.0.0.0."

   point 7 :  "SDP Connection address of zero"    of section 11, in RFC 3725
(3pcc).

   Are both really mean the same?
    I tried the answer in mail archive, but no luck :( .

   Thanks
   Srinivasa Rao




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From sip-bounces@ietf.org  Fri Nov 26 10:17:24 2004
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Srinivasa Rao <Srinivasa.Rao@alcatel.com>
Subject: Re: [SIP] RFC 3725, clarification
References: <20041126082127.40168.qmail@web54206.mail.yahoo.com>
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Yes, they are the same thing and have been listed separately by mistake.

Gonzalo

Srinivasa Rao wrote:

> If it is YES,
>     Why it has been listed separately?
> If anyone feels if it is not same, please let me know.
> I want authors also need to give their comments on this. 
> 
> Thanks
> Srinivasa Rao 
> 
> ----- Original Message ----- 
> From: "Rayees Khan" <rakhanhss@yahoo.com>
> To: "Srinivasa Rao" <Srinivasa.Rao@alcatel.com>; <sip@ietf.org>
> Sent: Friday, November 26, 2004 1:51 PM
> Subject: Re: [SIP] RFC 3725, clarification
> 
> 
> 
>>Yes,
>>
>>I guess both of them mean the same.
>>
>>
>>regards
>>Rayees
>>
>>--- Srinivasa Rao <Srinivasa.Rao@alcatel.com> wrote:
>>
>>
>>>Hi,
>>>   Anybody please clarify me about the difference of
>>>   point 1 : "Offers and answers that contain a
>>>connection line
>>>         with an address of 0.0.0.0."
>>>
>>>   point 7 :  "SDP Connection address of zero"    of
>>>section 11, in RFC 3725
>>>(3pcc).
>>>
>>>   Are both really mean the same?
>>>    I tried the answer in mail archive, but no luck
>>>:( .
>>>
>>>   Thanks
>>>   Srinivasa Rao
>>>
>>>
>>>
>>>


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From sip-bounces@ietf.org  Fri Nov 26 10:18:29 2004
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References: <20041126082127.40168.qmail@web54206.mail.yahoo.com>
Subject: Re: [SIP] RFC 3725, clarification
Date: Fri, 26 Nov 2004 14:30:59 +0530
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If it is YES,
    Why it has been listed separately?
If anyone feels if it is not same, please let me know.
I want authors also need to give their comments on this. 

Thanks
Srinivasa Rao 

----- Original Message ----- 
From: "Rayees Khan" <rakhanhss@yahoo.com>
To: "Srinivasa Rao" <Srinivasa.Rao@alcatel.com>; <sip@ietf.org>
Sent: Friday, November 26, 2004 1:51 PM
Subject: Re: [SIP] RFC 3725, clarification


> 
> Yes,
> 
> I guess both of them mean the same.
> 
> 
> regards
> Rayees
> 
> --- Srinivasa Rao <Srinivasa.Rao@alcatel.com> wrote:
> 
> > Hi,
> >    Anybody please clarify me about the difference of
> >    point 1 : "Offers and answers that contain a
> > connection line
> >          with an address of 0.0.0.0."
> > 
> >    point 7 :  "SDP Connection address of zero"    of
> > section 11, in RFC 3725
> > (3pcc).
> > 
> >    Are both really mean the same?
> >     I tried the answer in mail archive, but no luck
> > :( .
> > 
> >    Thanks
> >    Srinivasa Rao
> > 
> > 
> > 
> > 
> > _______________________________________________
> > Sip mailing list 
> > https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP
> > Protocol
> > Use sip-implementors@cs.columbia.edu for questions
> > on current sip
> > Use sipping@ietf.org for new developments on the
> > application of sip
> > 
> 
> 
> 
> 
> __________________________________ 
> Do you Yahoo!? 
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To: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Event List Changes
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Hisham Khartabil wrote:

> What should an RLS do when it receives a list subscription from a 
> different domain it has no trust relationship with, and that 
> subscription has an identity-header? I think it should remove that 
> header. Is this an identity draft issue or should the event list draft 
> also mention it?


Yes, this is very much an identity issue, but I think you're 
misunderstanding a couple of different things here.

First -- and this is why it's useful in the first place -- the 
"Identity" header has the means to assert identity across domains that 
have no trust relationship (except for by way of a certificate 
authority). That is its purpose.

Second, when an RLS forms a back-end subscription, it is doing exactly 
that: starting up a brand new subscription. The subscription is between 
the RLS and where it sends its SUBSCRIBE message, not the end-user's 
client and the resource. It is not passing through headers from the 
original request. It might very well generate outgoing header fields 
based on the incoming header fields, but it would be foolish to do so 
blindly. To be clear: an RLS, when forming SUBSCRIBE requests, is acting 
as a back-to-back user agent, not as a proxy. It has an obligation to 
understand everything it sends in a back-end subscription request.

So, to answer the question, "What should an RLS do when it receives a 
list subscription from a different domain it has no trust relationship 
with, and that subscription has an identity-header:" it verifies that 
the identity header is valid for the request (after retrieving the 
relevant certificate, if necessary); and, if it is, accepts that the 
requestor is authorized to assert the identity in the "From" header. 
 From a practical perspective, this means that it would be safe to feed 
that identity into policy rules associated with the subscription.

/a

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Hi, 

I wanted some clarification on the branch parameter in CANCEL. From what I Understand the branch parameter should be the same in the INVITE and the CANCEL(especially when using the magic cookie) so as to match the transaction being cancelled. 

>From RFC 3261 

The following procedures are used to construct a CANCEL request. The Request-URI, Call-ID, To, the numeric part of CSeq, and From header fields in the CANCEL request MUST be identical to those in the request being cancelled, including tags. A CANCEL constructed by a client MUST have only a single Via header field value matching the top Via value in the request being cancelled. Using the same values for these header fields allows the CANCEL to be matched with the request it cancels (Section 9.2 indicates how such matching occurs). However, the method part of the CSeq header field MUST have a value of CANCEL. This allows it to be identified and processed as a transaction in its own right (See Section 17). 


The branch parameter in the topmost Via header field of the request is examined. If it is present and begins with the magic cookie "z9hG4bK", the request was generated by a client transaction compliant to this specification. Therefore, the branch parameter will be unique across all transactions sent by that client. The request matches a transaction if: 

1. the branch parameter in the request is equal to the one in the top Via header field of the request that created the transaction, and 
2. the sent-by value in the top Via of the request is equal to the one in the request that created the transaction, and 
3. the method of the request matches the one that created the transaction, except for ACK, where the method of the request that created the transaction is INVITE. 

This matching rule applies to both INVITE and non-INVITE transactions alike. 


I would like to know if this is a correct CANCEL for the INVITE sent 

INVITE sip:1@172.20.22.33 SIP/2.0 
Max-Forwards: 10 
Record-Route: <sip:1@172.20.22.102;ftag=001201544d1a001b1f8360ea-1cbce160;lr=on> 
Via: SIP/2.0/UDP 172.20.22.102;branch=z9hG4bK6ee8.484c8882.0 
Via: SIP/2.0/UDP 172.20.22.75:5060;branch=z9hG4bK39e838b9 
From: "55" <sip:55@172.20.22.102>;tag=001201544d1a001b1f8360ea-1cbce160 
To: <sip:1@172.20.22.33> 
Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.22.75 
CSeq: 101 INVITE 
User-Agent: CSCO/6 
Contact: <sip:55@172.20.22.75:5060> 
Expires: 180 
Content-Type: application/sdp 
Content-Length: 247 
Accept: application/sdp 

v=0 
o=Cisco-SIPUA 14494 13084 IN IP4 172.20.22.75 
s=SIP Call 
c=IN IP4 172.20.22.75 
t=0 0 
m=audio 18990 RTP/AVP 0 8 18 101 
a=rtpmap:0 PCMU/8000 
a=rtpmap:8 PCMA/8000 
a=rtpmap:18 G729/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 


SIP/2.0 180 Ringing 
Via: SIP/2.0/UDP 172.20.22.102;branch= z9hG4bK6ee8.484c8882.0, SIP/2.0/UDP 172.20.22.75:5060;branch= z9hG4bK39e838b9 
Record-Route: <sip:1@172.20.22.102;ftag=001201544d1a001b1f8360ea-1cbce160;lr=on> 
To: <sip:1@172.20.22.33>; tag= 3e7ca296 
From: "55" <sip:55@172.20.22.102>; tag= 001201544d1a001b1f8360ea-1cbce160 
Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.22.75 
CSeq: 101 INVITE 
Contact: <sip:1@172.20.22.33;transport=UDP> 
Content-Length: 0 


CANCEL sip:1@172.20.22.33 SIP/2.0 
Max-Forwards: 10 
Record-Route: <sip:1@172.20.22.102;ftag=001201544d1a001b1f8360ea-1cbce160;lr=on> 
Via: SIP/2.0/UDP 172.20.22.102;branch=z9hG4bK6ee8.584c8882.0 
Via: SIP/2.0/UDP 172.20.22.75:5060;branch=z9hG4bK6a729507 
From: "55" <sip:55@172.20.22.102>;tag=001201544d1a001b1f8360ea-1cbce160 
To: <sip:1@172.20.22.33> 
Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.22.75 
CSeq: 101 CANCEL 
User-Agent: CSCO/6 
Content-Length: 0 

In the above case the branch parameter matching fails. 

Regards, 
Murali

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<DIV>Hi, <BR><BR>I wanted some clarification on the branch parameter in CANCEL. From what I Understand the branch parameter should be the same in the INVITE and the CANCEL(especially when using the magic cookie)&nbsp;so as to match the transaction being cancelled. <BR><BR>From RFC 3261 <BR><BR>The following procedures are used to construct a CANCEL request. The Request-URI, Call-ID, To, the numeric part of CSeq, and From header fields in the CANCEL request MUST be identical to those in the request being cancelled, including tags. <STRONG>A CANCEL constructed by a client MUST have only a single Via header field value matching the top Via value in the request being cancelled. Using the same values for these header fields allows the CANCEL to be matched with the request it cancels</STRONG> (Section 9.2 indicates how such matching occurs). However, the method part of the CSeq header field MUST have a value of CANCEL. This allows it to be identified and processed as a transaction!
  in its
 own right (See Section 17). <BR><BR><BR>The branch parameter in the topmost Via header field of the request is examined. If it is present and begins with the magic cookie "z9hG4bK", the request was generated by a client transaction compliant to this specification. Therefore, the branch parameter will be unique across all transactions sent by that client. The request matches a transaction if: <BR><BR><STRONG>1. the branch parameter in the request is equal to the one in the top Via header field of the request that created the transaction,</STRONG> and <BR>2. the sent-by value in the top Via of the request is equal to the one in the request that created the transaction, and <BR>3. the method of the request matches the one that created the transaction, except for ACK, where the method of the request that created the transaction is INVITE. <BR><BR>This matching rule applies to both INVITE and non-INVITE transactions alike. <BR><BR><BR>I would like to know if this is a correct CA!
 NCEL for
 the INVITE sent <BR><BR>INVITE sip:1@172.20.22.33 SIP/2.0 <BR>Max-Forwards: 10 <BR>Record-Route: &lt;sip:1@172.20.22.102;ftag=001201544d1a001b1f8360ea-1cbce160;lr=on&gt; <BR><STRONG>Via: SIP/2.0/UDP 172.20.22.102;branch=z9hG4bK6ee8.484c8882.0 <BR>Via: SIP/2.0/UDP 172.20.22.75:5060;branch=z9hG4bK39e838b9 <BR></STRONG>From: "55" &lt;sip:55@172.20.22.102&gt;;tag=001201544d1a001b1f8360ea-1cbce160 <BR>To: &lt;sip:1@172.20.22.33&gt; <BR>Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.22.75 <BR>CSeq: 101 INVITE <BR>User-Agent: CSCO/6 <BR>Contact: &lt;sip:55@172.20.22.75:5060&gt; <BR>Expires: 180 <BR>Content-Type: application/sdp <BR>Content-Length: 247 <BR>Accept: application/sdp <BR><BR>v=0 <BR>o=Cisco-SIPUA 14494 13084 IN IP4 172.20.22.75 <BR>s=SIP Call <BR>c=IN IP4 172.20.22.75 <BR>t=0 0 <BR>m=audio 18990 RTP/AVP 0 8 18 101 <BR>a=rtpmap:0 PCMU/8000 <BR>a=rtpmap:8 PCMA/8000 <BR>a=rtpmap:18 G729/8000 <BR>a=rtpmap:101 telephone-event/8000 <BR>a=fmtp:101 0-15 <BR><BR><BR>SIP/2.!
 0 180
 Ringing <BR>Via: SIP/2.0/UDP 172.20.22.102;branch= z9hG4bK6ee8.484c8882.0, SIP/2.0/UDP 172.20.22.75:5060;branch= z9hG4bK39e838b9 <BR>Record-Route: &lt;sip:1@172.20.22.102;ftag=001201544d1a001b1f8360ea-1cbce160;lr=on&gt; <BR>To: &lt;sip:1@172.20.22.33&gt;; tag= 3e7ca296 <BR>From: "55" &lt;sip:55@172.20.22.102&gt;; tag= 001201544d1a001b1f8360ea-1cbce160 <BR>Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.22.75 <BR>CSeq: 101 INVITE <BR>Contact: &lt;sip:1@172.20.22.33;transport=UDP&gt; <BR>Content-Length: 0 <BR><BR><BR>CANCEL sip:1@172.20.22.33 SIP/2.0 <BR>Max-Forwards: 10 <BR>Record-Route: &lt;sip:1@172.20.22.102;ftag=001201544d1a001b1f8360ea-1cbce160;lr=on&gt; <BR><STRONG>Via: SIP/2.0/UDP 172.20.22.102;branch=z9hG4bK6ee8.584c8882.0 <BR>Via: SIP/2.0/UDP 172.20.22.75:5060;branch=z9hG4bK6a729507 <BR></STRONG>From: "55" &lt;sip:55@172.20.22.102&gt;;tag=001201544d1a001b1f8360ea-1cbce160 <BR>To: &lt;sip:1@172.20.22.33&gt; <BR>Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.!
 22.75
 <BR>CSeq: 101 CANCEL <BR>User-Agent: CSCO/6 <BR>Content-Length: 0 <BR><BR>In the above case the branch parameter matching fails. <BR><BR>Regards, <BR>Murali</DIV><p>__________________________________________________<br>Do You Yahoo!?<br>Tired of spam?  Yahoo! Mail has the best spam protection around <br>http://mail.yahoo.com 
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Thread-Topic: Mistake in PRES-URI BNF
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> Hi,
>=20
> I found a problem in the BNF of PRES-URI.
>=20
> RFC 2396 defined the generic syntax of URI. It explicitly says that =
the following (delims) characters are disallowed within URI:
> delims=3D "<" | ">" | "#" | "%" | <">=20
>=20
> However, in RFCs 3859 and 2822 the definition for PRES-URI is:
> 	PRES URI =3D "pres:" [ to ] [ headers ]=20
> 	to =3D mailbox=20
> 	mailbox =3D name-addr / addr-spec=20
> 	name-addr =3D [display-name] angle-addr=20
> 	angle-addr =3D [CFWS] "<" addr-spec ">" [CFWS] / obs-angle-addr=20
> 	addr-spec =3D local-part "@" domain=20
> 	display-name =3D phrase=20
> Which means that pres addresses can look like pres:"name"<user@host>, =
hence may contain " and <>.
>=20
> Therefore I believe that the PRES-URI BNF is mistaken.=20
>=20
> Thanks,
> Tamar.
>=20

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> Hi,
>=20
> I found a problem in the BNF of PRES-URI.
>=20
> RFC 2396 defined the generic syntax of URI. It explicitly says that =
the following (delims) characters are disallowed within URI:
> delims=3D "<" | ">" | "#" | "%" | <">=20
>=20
> However, in RFCs 3859 and 2822 the definition for PRES-URI is:
> 	PRES URI =3D "pres:" [ to ] [ headers ]=20
> 	to =3D mailbox=20
> 	mailbox =3D name-addr / addr-spec=20
> 	name-addr =3D [display-name] angle-addr=20
> 	angle-addr =3D [CFWS] "<" addr-spec ">" [CFWS] / obs-angle-addr=20
> 	addr-spec =3D local-part "@" domain=20
> 	display-name =3D phrase=20
> Which means that pres addresses can look like pres:"name"<user@host>, =
hence may contain " and <>.
>=20
> Therefore I believe that the PRES-URI BNF is mistaken.=20
>=20
> Thanks,
> Tamar.
>=20

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Subject: RE: [Sip] branch parameter in CANCEL
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Murali,

=20

You are correct in your interpretation. The branch parameter for a
CANCEL does need to be the same as the request it is canceling. In fact,
it appears you are using a particular version of Cisco SIP firmware that
has already been the subject of a TAC case I logged for exactly this
problem.

I suggest that if you upgrade the firmware on that phone the issue will
be resolved.

Also, this type of post probably belongs on SIP-IMPLEMENTORS rather than
the IETF list.

=20

HTH,

=20

B

=20

Ben Gatewood

Lead Architect - IP Telephony

Streamdoor Ltd.

114b Power Rd

London=20

W4 5PY

+44 20 7422 0439

+44 7835 136 187

bgatewood@streamdoor.net

=20

=20

=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
murali foru
Sent: 29 November 2004 04:32
To: sip@ietf.org
Subject: [Sip] branch parameter in CANCEL

=20

Hi,=20

I wanted some clarification on the branch parameter in CANCEL. From what
I Understand the branch parameter should be the same in the INVITE and
the CANCEL(especially when using the magic cookie) so as to match the
transaction being cancelled.=20

>From RFC 3261=20

The following procedures are used to construct a CANCEL request. The
Request-URI, Call-ID, To, the numeric part of CSeq, and From header
fields in the CANCEL request MUST be identical to those in the request
being cancelled, including tags. A CANCEL constructed by a client MUST
have only a single Via header field value matching the top Via value in
the request being cancelled. Using the same values for these header
fields allows the CANCEL to be matched with the request it cancels
(Section 9.2 indicates how such matching occurs). However, the method
part of the CSeq header field MUST have a value of CANCEL. This allows
it to be identified and processed as a transaction! in its own right
(See Section 17).=20


The branch parameter in the topmost Via header field of the request is
examined. If it is present and begins with the magic cookie "z9hG4bK",
the request was generated by a client transaction compliant to this
specification. Therefore, the branch parameter will be unique across all
transactions sent by that client. The request matches a transaction if:=20=


1. the branch parameter in the request is equal to the one in the top
Via header field of the request that created the transaction, and=20
2. the sent-by value in the top Via of the request is equal to the one
in the request that created the transaction, and=20
3. the method of the request matches the one that created the
transaction, except for ACK, where the method of the request that
created the transaction is INVITE.=20

This matching rule applies to both INVITE and non-INVITE transactions
alike.=20


I would like to know if this is a correct CA! NCEL for the INVITE sent=20=


INVITE sip:1@172.20.22.33 SIP/2.0=20
Max-Forwards: 10=20
Record-Route:
<sip:1@172.20.22.102;ftag=3D001201544d1a001b1f8360ea-1cbce160;lr=3Don>=20=

Via: SIP/2.0/UDP 172.20.22.102;branch=3Dz9hG4bK6ee8.484c8882.0=20
Via: SIP/2.0/UDP 172.20.22.75:5060;branch=3Dz9hG4bK39e838b9=20
From: "55" <sip:55@172.20.22.102>;tag=3D001201544d1a001b1f8360ea-1cbce160=
=20
To: <sip:1@172.20.22.33>=20
Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.22.75=20
CSeq: 101 INVITE=20
User-Agent: CSCO/6=20
Contact: <sip:55@172.20.22.75:5060>=20
Expires: 180=20
Content-Type: application/sdp=20
Content-Length: 247=20
Accept: application/sdp=20

v=3D0=20
o=3DCisco-SIPUA 14494 13084 IN IP4 172.20.22.75=20
s=3DSIP Call=20
c=3DIN IP4 172.20.22.75=20
t=3D0 0=20
m=3Daudio 18990 RTP/AVP 0 8 18 101=20
a=3Drtpmap:0 PCMU/8000=20
a=3Drtpmap:8 PCMA/8000=20
a=3Drtpmap:18 G729/8000=20
a=3Drtpmap:101 telephone-event/8000=20
a=3Dfmtp:101 0-15=20


SIP/2.! 0 180 Ringing=20
Via: SIP/2.0/UDP 172.20.22.102;branch=3D z9hG4bK6ee8.484c8882.0,
SIP/2.0/UDP 172.20.22.75:5060;branch=3D z9hG4bK39e838b9=20
Record-Route:
<sip:1@172.20.22.102;ftag=3D001201544d1a001b1f8360ea-1cbce160;lr=3Don>=20=

To: <sip:1@172.20.22.33>; tag=3D 3e7ca296=20
From: "55" <sip:55@172.20.22.102>; tag=3D
001201544d1a001b1f8360ea-1cbce160=20
Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.22.75=20
CSeq: 101 INVITE=20
Contact: <sip:1@172.20.22.33;transport=3DUDP>=20
Content-Length: 0=20


CANCEL sip:1@172.20.22.33 SIP/2.0=20
Max-Forwards: 10=20
Record-Route:
<sip:1@172.20.22.102;ftag=3D001201544d1a001b1f8360ea-1cbce160;lr=3Don>=20=

Via: SIP/2.0/UDP 172.20.22.102;branch=3Dz9hG4bK6ee8.584c8882.0=20
Via: SIP/2.0/UDP 172.20.22.75:5060;branch=3Dz9hG4bK6a729507=20
From: "55" <sip:55@172.20.22.102>;tag=3D001201544d1a001b1f8360ea-1cbce160=
=20
To: <sip:1@172.20.22.33>=20
Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.! 22.75=20
CSeq: 101 CANCEL=20
User-Agent: CSCO/6=20
Content-Length: 0=20

In the above case the branch parameter matching fails.=20

Regards,=20
Murali

__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around=20
http://mail.yahoo.com
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<div class=3DSection1>

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


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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>You are correct in your interpretati=
on.
The branch parameter for a CANCEL does need to be the same as the request=
 it is
canceling. In fact, it appears you are using a particular version of Cisc=
o SIP firmware
that has already been the subject of a TAC case I logged for exactly this=
 problem.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I suggest that if you upgrade the fi=
rmware
on that phone the issue will be resolved.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Also, this type of post probably bel=
ongs
on SIP-IMPLEMENTORS rather than the IETF list.<o:p></o:p></span></font></=
p>

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

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

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

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

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

<p class=3DMsoNormal><st1:PersonName w:st=3D"on"><font size=3D2 color=3Dn=
avy
 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:nav=
y'>Ben
 Gatewood</span></font></st1:PersonName><font size=3D2 color=3Dnavy face=3D=
Arial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p></o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Lead Architect &#8211; IP Telephony<=
o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>114b <st1:Street w:st=3D"on"><st1:ad=
dress
 w:st=3D"on">Power Rd</st1:address></st1:Street><o:p></o:p></span></font>=
</p>

<p class=3DMsoNormal><st1:City w:st=3D"on"><st1:place w:st=3D"on"><font s=
ize=3D2
  color=3Dnavy face=3DArial><span style=3D'font-size:10.0pt;font-family:A=
rial;
  color:navy'>London</span></font></st1:place></st1:City><font size=3D2
color=3Dnavy face=3DArial><span style=3D'font-size:10.0pt;font-family:Ari=
al;
color:navy'> <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>W4 5PY<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>+44 20 7422 0439<o:p></o:p></span></=
font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>+44 7835 136 187<o:p></o:p></span></=
font></p>

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

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

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

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

<div>

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

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

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

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-=
size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D=
2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b><span style=3D'font=
-weight:
bold'>On Behalf Of </span></b>murali foru<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 29 November 2004 04:=
32<br>
<b><span style=3D'font-weight:bold'>To:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] branch para=
meter in
CANCEL</span></font><o:p></o:p></p>

</div>

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

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Hi, <br>
<br>
I wanted some clarification on the branch parameter in CANCEL. From what =
I
Understand the branch parameter should be the same in the INVITE and the
CANCEL(especially when using the magic cookie)&nbsp;so as to match the
transaction being cancelled. <br>
<br>
>From RFC 3261 <br>
<br>
The following procedures are used to construct a CANCEL request. The
Request-URI, Call-ID, To, the numeric part of CSeq, and From header field=
s in
the CANCEL request MUST be identical to those in the request being cancel=
led,
including tags. <strong><b><font face=3D"Times New Roman">A CANCEL constr=
ucted by
a client MUST have only a single Via header field value matching the top =
Via
value in the request being cancelled. Using the same values for these hea=
der
fields allows the CANCEL to be matched with the request it cancels</font>=
</b></strong>
(Section 9.2 indicates how such matching occurs). However, the method par=
t of
the CSeq header field MUST have a value of CANCEL. This allows it to be i=
dentified
and processed as a transaction! in its own right (See Section 17). <br>
<br>
<br>
The branch parameter in the topmost Via header field of the request is
examined. If it is present and begins with the magic cookie
&quot;z9hG4bK&quot;, the request was generated by a client transaction
compliant to this specification. Therefore, the branch parameter will be =
unique
across all transactions sent by that client. The request matches a transa=
ction
if: <br>
<br>
<strong><b><font face=3D"Times New Roman">1. the branch parameter in the =
request
is equal to the one in the top Via header field of the request that creat=
ed the
transaction,</font></b></strong> and <br>
2. the sent-by value in the top Via of the request is equal to the one in=
 the
request that created the transaction, and <br>
3. the method of the request matches the one that created the transaction=
, except
for ACK, where the method of the request that created the transaction is
INVITE. <br>
<br>
This matching rule applies to both INVITE and non-INVITE transactions ali=
ke. <br>
<br>
<br>
I would like to know if this is a correct CA! NCEL for the INVITE sent <b=
r>
<br>
INVITE sip:1@172.20.22.33 SIP/2.0 <br>
Max-Forwards: 10 <br>
Record-Route:
&lt;sip:1@172.20.22.102;ftag=3D001201544d1a001b1f8360ea-1cbce160;lr=3Don&=
gt; <br>
<strong><b><font face=3D"Times New Roman">Via: SIP/2.0/UDP
172.20.22.102;branch=3Dz9hG4bK6ee8.484c8882.0 </font></b></strong><b><spa=
n
style=3D'font-weight:bold'><br>
<strong><b><font face=3D"Times New Roman">Via: SIP/2.0/UDP
172.20.22.75:5060;branch=3Dz9hG4bK39e838b9 </font></b></strong><br>
</span></b>From: &quot;55&quot; &lt;sip:55@172.20.22.102&gt;;tag=3D001201=
544d1a001b1f8360ea-1cbce160
<br>
To: &lt;sip:1@172.20.22.33&gt; <br>
Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.22.75 <br>
CSeq: 101 INVITE <br>
User-Agent: CSCO/6 <br>
Contact: &lt;sip:55@172.20.22.75:5060&gt; <br>
Expires: 180 <br>
Content-Type: application/sdp <br>
Content-Length: 247 <br>
Accept: application/sdp <br>
<br>
v=3D0 <br>
o=3DCisco-SIPUA 14494 13084 IN IP4 172.20.22.75 <br>
s=3DSIP Call <br>
c=3DIN IP4 172.20.22.75 <br>
t=3D0 0 <br>
m=3Daudio 18990 RTP/AVP 0 8 18 101 <br>
a=3Drtpmap:0 PCMU/8000 <br>
a=3Drtpmap:8 PCMA/8000 <br>
a=3Drtpmap:18 G729/8000 <br>
a=3Drtpmap:101 telephone-event/8000 <br>
a=3Dfmtp:101 0-15 <br>
<br>
<br>
SIP/2.! 0 180 Ringing <br>
Via: SIP/2.0/UDP 172.20.22.102;branch=3D z9hG4bK6ee8.484c8882.0, SIP/2.0/=
UDP
172.20.22.75:5060;branch=3D z9hG4bK39e838b9 <br>
Record-Route: &lt;sip:1@172.20.22.102;ftag=3D001201544d1a001b1f8360ea-1cb=
ce160;lr=3Don&gt;
<br>
To: &lt;sip:1@172.20.22.33&gt;; tag=3D 3e7ca296 <br>
From: &quot;55&quot; &lt;sip:55@172.20.22.102&gt;; tag=3D
001201544d1a001b1f8360ea-1cbce160 <br>
Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.22.75 <br>
CSeq: 101 INVITE <br>
Contact: &lt;sip:1@172.20.22.33;transport=3DUDP&gt; <br>
Content-Length: 0 <br>
<br>
<br>
CANCEL sip:1@172.20.22.33 SIP/2.0 <br>
Max-Forwards: 10 <br>
Record-Route:
&lt;sip:1@172.20.22.102;ftag=3D001201544d1a001b1f8360ea-1cbce160;lr=3Don&=
gt; <br>
<strong><b><font face=3D"Times New Roman">Via: SIP/2.0/UDP
172.20.22.102;branch=3Dz9hG4bK6ee8.584c8882.0 </font></b></strong><b><spa=
n
style=3D'font-weight:bold'><br>
<strong><b><font face=3D"Times New Roman">Via: SIP/2.0/UDP
172.20.22.75:5060;branch=3Dz9hG4bK6a729507 </font></b></strong><br>
</span></b>From: &quot;55&quot;
&lt;sip:55@172.20.22.102&gt;;tag=3D001201544d1a001b1f8360ea-1cbce160 <br>=

To: &lt;sip:1@172.20.22.33&gt; <br>
Call-ID: 00120154-4d1a001b-54ba3444-4e6b98f1@172.20.! 22.75 <br>
CSeq: 101 CANCEL <br>
User-Agent: CSCO/6 <br>
Content-Length: 0 <br>
<br>
In the above case the branch parameter matching fails. <br>
<br>
Regards, <br>
Murali<o:p></o:p></span></font></p>

</div>

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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 Stream Control Transmission Protocol as a 
                          Transport for  for the Session Initiation Protocol
	Author(s)	: J. Rosenberg, et al.
	Filename	: draft-ietf-sip-sctp-05.txt
	Pages		: 9
	Date		: 2004-11-29
	
This document specifies a mechanism for usage of SCTP (the Stream
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   Initiation Protocol) entities.  SCTP is a new protocol which provides
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From sip-bounces@ietf.org  Tue Nov 30 01:05:28 2004
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From: "Jesske, R" <R.Jesske@t-com.net>
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Date: Tue, 30 Nov 2004 06:52:58 +0100
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Dear all,
in the past there was a discussion regarding the specification of a Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt)
What is the actual status of this activity. 
We are still interested in such kind of originating indication if the call/communication is coming from a normal SIP user or a SIP -Payphone or SIP-Hotelphone. 
Is there a interest from other parties in this issue?

Best Regards

Roland

Deutsche Telekom AG
T-Com Zentrale
Roland Jesske, TE332-2
Section TE33; Signalling, Gateways and Switching Systems 
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940 
Fax:      +49 6151 83-4577 
email:   r.jesske@t-com.net




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From sip-bounces@ietf.org  Tue Nov 30 02:55:28 2004
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Subject: RE: [Sip] Mistake in PRES-URI BNF
Date: Tue, 30 Nov 2004 09:41:42 +0200
Message-ID: <10DA2C035FE3BC4FA8D73EC55FC43D9B5A3A71@rvil-mail.radvision.com>
Thread-Topic: Mistake in PRES-URI BNF
Thread-Index: AcTWCaDjM4FIQsWjTuS7Fjes8AVjBwAAo34QACjqB0A=
From: "Sarit Galanos" <Sarit@radvision.com>
To: "Tamar Barzuza" <TamarB@radvision.com>, <sip@ietf.org>
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Hi,
I also think that there is a critical problem with the BNF.
According to the BNF the following From header is valid:

From: Bob<pres:Bob<bob@home.com>>

This means that the pres address has the "<" in it (and not around it).
This is not valid according the generic URI syntax.

How should such and address be parsed???

Basically, I think that the problem is the fact that the pres URI =
contains a mailbox and not an addr-spec in it.

Thanks,
Sarit.

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Tamar Barzuza
Sent: Monday, November 29, 2004 2:09 PM
To: sip@ietf.org
Subject: [Sip] Mistake in PRES-URI BNF


> Hi,
>=20
> I found a problem in the BNF of PRES-URI.
>=20
> RFC 2396 defined the generic syntax of URI. It explicitly says that =
the following (delims) characters are disallowed within URI:
> delims=3D "<" | ">" | "#" | "%" | <">=20
>=20
> However, in RFCs 3859 and 2822 the definition for PRES-URI is:
> 	PRES URI =3D "pres:" [ to ] [ headers ]=20
> 	to =3D mailbox=20
> 	mailbox =3D name-addr / addr-spec=20
> 	name-addr =3D [display-name] angle-addr=20
> 	angle-addr =3D [CFWS] "<" addr-spec ">" [CFWS] / obs-angle-addr=20
> 	addr-spec =3D local-part "@" domain=20
> 	display-name =3D phrase=20
> Which means that pres addresses can look like pres:"name"<user@host>, =
hence may contain " and <>.
>=20
> Therefore I believe that the PRES-URI BNF is mistaken.=20
>=20
> Thanks,
> Tamar.
>=20

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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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Yes,

I guess both of them mean the same.


regards
Rayees

--- Srinivasa Rao <Srinivasa.Rao@alcatel.com> wrote:

> Hi,
>    Anybody please clarify me about the difference of
>    point 1 : "Offers and answers that contain a
> connection line
>          with an address of 0.0.0.0."
> 
>    point 7 :  "SDP Connection address of zero"    of
> section 11, in RFC 3725
> (3pcc).
> 
>    Are both really mean the same?
>     I tried the answer in mail archive, but no luck
> :( .
> 
>    Thanks
>    Srinivasa Rao
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list 
> https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP
> Protocol
> Use sip-implementors@cs.columbia.edu for questions
> on current sip
> Use sipping@ietf.org for new developments on the
> application of sip
> 



		
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From sip-bounces@ietf.org  Tue Nov 30 09:55:45 2004
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Adam,

In general I think this is a reasonable compromise in order to make 
progress in the near term. However, I think you have been a bit heavy 
handed in the restrictions. You say:

      "any RLS which uses SIP back-end
       subscriptions to acquire information about the resources in a
       resource list MUST be able to act as an authentication service"

This means that any RLS implementation that does back-end subscriptions 
MUST be a privileged member of the domain, so that it may act as an 
authentication service.

While this may be a reasonable strategy in many cases, it still ought to 
be possible for someone to build an RLS that isn't so priviliged. One 
that isn't just won't be about to use the original subscriber's identity 
in the back-end subscriptions, and instead will have to either generate 
subscriptions with no identity, or with its own identity - same as when 
handling subscribers from outside its domain.

	Paul

Adam Roach wrote:
> I'm revising the event list draft currently. In DC, we discussed at 
> length the authentication model for situations in which the RLS and 
> subscriber are in the same domain; I believe I've captured this pretty 
> well (but encourage interested parties to double check my interpretation 
> of consensus).
> 
> When I went to update the document, however, I found that there was 
> still something of a hole for the other case -- the one we described as 
> a corner case. I know that we agreed that (1) additional mechanism could 
> be developed to solve this case, and (2) we didn't want to wait for 
> these mechanisms before publishing the event list document. However, I 
> don't remember discussing what RLS implementors can do when they receive 
> a request for a list resource from a user outside their domain.
> 
> So, I came up with what seemed like the most logical solution, given the 
> other discussions we had. Before I submit -07 of the document, I'd like 
> to have a few people look over my proposed text; if you think this is 
> the wrong direction to take it, speak up now. If I get no comments 
> before the end of next week, I'll submit the draft with the following 
> language.
> 
>> 7.1 Authentication
>>
>>    If back-end subscriptions are required to retrieve resource state
>>    information, the end user is no longer the direct subscriber to the
>>    state of the resource.  This means that direct authentication of the
>>    user is no longer possible.
>>
>> 7.1.1 RLS and Subscriber in the Same Domain
>>
>>    It is expected that the most common deployment of RLSes entails the
>>    subscribers to the RLS being in the same domain as the RLS.  When
>>    this is the case, the RLS then has the ability to act as an
>>    authenication service.  The role of authentication service is defined
>>    in "Enhancements for Authenticated Identity Management in the Session
>>    Initiation Protocol (SIP)" [7].
>>
>>    At a high level, under this system, the RLS authenticates the
>>    subscriber, and then includes an "Identity" header field in all of
>>    the back-end subscriptions performed on behalf of that authenticated
>>    user; this "Identity" header field cryptographically asserts that the
>>    request has been authorized to be made on behalf of the user
>>    indicated in the "From" header field.
>>
>>    Because the ability to authenticate requests is central to the proper
>>    functioning of the network, any RLS which uses SIP back-end
>>    subscriptions to acquire information about the resources in a
>>    resource list MUST be able to act as an authentication service as
>>    defined in [7].
>>
>> 7.1.2 RLS and Subscriber in Different Domains
>>
>>    In the general case, the SIP Authenticated Identity extensions do not
>>    provide a means for the RLS to securely assert that subscriptions are
>>    being performed on the end user's behalf.  Specifically, when the
>>    subscriber and the RLS are in different domains, the RLS will have no
>>    means by which it can vouch for the user's identity.  Mechanisms by
>>    which back-end subscriptions in such circumstances can be
>>    authenticated are left for future study.
>>
>>    Until such general solutions are developed, RLSes which are in a
>>    different domain than the subscriber on whose behalf they are
>>    creating back-end susbcriptions SHOULD subscribe to the resources
>>    using their own identity.  By doing so, the RLS will generally obtain
>>    only the resource information which is made publicly available.
>>
>>    Absent such general solutions, implemenations of subscriber user
>>    agents MAY attempt direct subscriptions to resources in the resouce
>>    list when subscribing to an RLS outside of their domain (either
>>    directly or by way of another resource list subscription).  The
>>    resouces to be subscribed to will be those indicated in the the "uri"
>>    attribute of the <resource> elements present in the RLMI document
>>    returned by the RLS.  Directly subscribing to the resources allows
>>    proper authentication of the user to take place, which will generally
>>    authorize them to receive more complete state information.
>>    Implementations which choose to perform such direct subscriptions
>>    SHOULD use the data retrieved directly to the exclusion of any
>>    information about the resource obtained via the list subscription.
> 
> 
> 
> /a
> 
> _______________________________________________
> 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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> Use sipping@ietf.org for new developments on the application of sip
> 


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Subject: [Sip] Re: [Iptel] Calling Party's Category
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The question was what the requirement is for this sort of indication.  How is it used?

Jesske, R wrote:
> Dear all,
> in the past there was a discussion regarding the specification of a Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt)
> What is the actual status of this activity. 
> We are still interested in such kind of originating indication if the call/communication is coming from a normal SIP user or a SIP -Payphone or SIP-Hotelphone. 
> Is there a interest from other parties in this issue?
> 
> Best Regards
> 
> Roland
> 
> Deutsche Telekom AG
> T-Com Zentrale
> Roland Jesske, TE332-2
> Section TE33; Signalling, Gateways and Switching Systems 
> Am Kavalleriesand 3, 64295 Darmstadt, Germany
> Phone:  +49 6151 83-5940 
> Fax:      +49 6151 83-4577 
> email:   r.jesske@t-com.net
> 
> 
> 
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
> 
> 

-- 
Tom Taylor
Carrier VoIP Standards Development
Nortel Networks
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortelnetworks.com

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From: "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>
To: "Tom Taylor" <taylor@nortelnetworks.com>, "Jesske, R" <R.Jesske@t-com.net>
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Hi,

As an fixed line operator we also are interested in this feature and =
would be happy that the IETF finally moves forward with this draft. This =
type of parameter is widely signalled in ISDN/PSTN networks. In SIP (for =
example) it is required in context of both:

- pure SIP networks (e.g. in case of a SIP payphone), and
- interworking with legacy networks=20

I think the uses case for this parameter will depend on the operator =
service portfolio. For example, in France, the calling party's category =
parameter is used for the Call Return service.

Best regards,
sebastien=20

-----Message d'origine-----
De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] De la part =
de Tom Taylor
Envoy=E9 : mardi 30 novembre 2004 16:07
=C0 : Jesske, R
Cc : iptel@ietf.org; sip@ietf.org
Objet : Re: [Iptel] Calling Party's Category

The question was what the requirement is for this sort of indication.  =
How is it used?

Jesske, R wrote:
> Dear all,
> in the past there was a discussion regarding the specification of a=20
> Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) What =
is the actual status of this activity.
> We are still interested in such kind of originating indication if the =
call/communication is coming from a normal SIP user or a SIP -Payphone =
or SIP-Hotelphone.=20
> Is there a interest from other parties in this issue?
>=20
> Best Regards
>=20
> Roland
>=20
> Deutsche Telekom AG
> T-Com Zentrale
> Roland Jesske, TE332-2
> Section TE33; Signalling, Gateways and Switching Systems Am=20
> Kavalleriesand 3, 64295 Darmstadt, Germany
> Phone:  +49 6151 83-5940=20
> Fax:      +49 6151 83-4577=20
> email:   r.jesske@t-com.net
>=20
>=20
>=20
>=20
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>=20
>=20

--
Tom Taylor
Carrier VoIP Standards Development
Nortel Networks
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortelnetworks.com

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Paul Kyzivat wrote:

> In general I think this is a reasonable compromise in order to make 
> progress in the near term. However, I think you have been a bit heavy 
> handed in the restrictions. You say:
>
>      "any RLS which uses SIP back-end
>       subscriptions to acquire information about the resources in a
>       resource list MUST be able to act as an authentication service"
>
> This means that any RLS implementation that does back-end 
> subscriptions MUST be a privileged member of the domain, so that it 
> may act as an authentication service.


Good point. I was trying to capture the apparent consensus in DC that 
the SIP Identity draft is mandatory to implement (optional to use) in 
RLSes. As you point out, I have probably gone too far, since my phrasing 
implies certain administrative details instead of stopping at 
implementation details.

I'll re-work the language so that it applies only to implementation, and 
not to deployment.

/a

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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
Date: Tue, 30 Nov 2004 11:23:16 -0500
To: "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>
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On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:

> Hi,
>
> As an fixed line operator we also are interested in this feature and=20=

> would be happy that the IETF finally moves forward with this draft.=20
> This type of parameter is widely signalled in ISDN/PSTN networks. In=20=

> SIP (for example) it is required in context of both:
>
> - pure SIP networks (e.g. in case of a SIP payphone), and
> - interworking with legacy networks
>
> I think the uses case for this parameter will depend on the operator=20=

> service portfolio. For example, in France, the calling party's=20
> category parameter is used for the Call Return service.
>
Could you explain a bit about how this is used? I, for one would find=20
it helpful and possibly a motivating use case.

Also note that if this information is used for any security or=20
billing-sensitive operations, we have to deal with the issue of how to=20=

secure it against spoofing (SIP UAs are generally not trusted). Would=20
you believe a call made by Robert Mitnick from a prison phone would=20
actually have a prison CPC in it's SIP headers :-) ?

(Mitnick is a widely known phone freak and network hacker who served a=20=

number of years in US prisons).

> Best regards,
> sebastien
>
> -----Message d'origine-----
> De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] De la part=20=

> de Tom Taylor
> Envoy=E9 : mardi 30 novembre 2004 16:07
> =C0 : Jesske, R
> Cc : iptel@ietf.org; sip@ietf.org
> Objet : Re: [Iptel] Calling Party's Category
>
> The question was what the requirement is for this sort of indication. =20=

> How is it used?
>
> Jesske, R wrote:
>> Dear all,
>> in the past there was a discussion regarding the specification of a
>> Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) What=20=

>> is the actual status of this activity.
>> We are still interested in such kind of originating indication if the=20=

>> call/communication is coming from a normal SIP user or a SIP=20
>> -Payphone or SIP-Hotelphone.
>> Is there a interest from other parties in this issue?
>>
>> Best Regards
>>
>> Roland
>>
>> Deutsche Telekom AG
>> T-Com Zentrale
>> Roland Jesske, TE332-2
>> Section TE33; Signalling, Gateways and Switching Systems Am
>> Kavalleriesand 3, 64295 Darmstadt, Germany
>> Phone:  +49 6151 83-5940
>> Fax:      +49 6151 83-4577
>> email:   r.jesske@t-com.net
>>
>>
>>
>>
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>>
>>
>
> --
> Tom Taylor
> Carrier VoIP Standards Development
> Nortel Networks
> Phone +1 613 763 1496  (ESN 393-1496)
> E-mail: taylor@nortelnetworks.com
>
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>
> _______________________________________________
> 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
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.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


From sip-bounces@ietf.org  Tue Nov 30 12:06:25 2004
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Subject: RE: [Sip] RE: [Iptel] Calling Party's Category
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To: "David R Oran" <oran@cisco.com>
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In the Call Return service, a destination exchange memories an incoming =
call which failed due to non-reply. If the call was orginated by a =
payphone then the destination exchange does not memories the number =
since it makes no sense in calling back a payphone.=20

BR,
sebastien
 =20

-----Message d'origine-----
De : David R Oran [mailto:oran@cisco.com]=20
Envoy=E9 : mardi 30 novembre 2004 17:23
=C0 : GARCIN Sebastien RD-CORE-ISS
Cc : Jesske, R; Tom Taylor; iptel@ietf.org; sip@ietf.org
Objet : Re: [Sip] RE: [Iptel] Calling Party's Category


On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:

> Hi,
>
> As an fixed line operator we also are interested in this feature and=20
> would be happy that the IETF finally moves forward with this draft.
> This type of parameter is widely signalled in ISDN/PSTN networks. In=20
> SIP (for example) it is required in context of both:
>
> - pure SIP networks (e.g. in case of a SIP payphone), and
> - interworking with legacy networks
>
> I think the uses case for this parameter will depend on the operator=20
> service portfolio. For example, in France, the calling party's=20
> category parameter is used for the Call Return service.
>
Could you explain a bit about how this is used? I, for one would find it =
helpful and possibly a motivating use case.

Also note that if this information is used for any security or =
billing-sensitive operations, we have to deal with the issue of how to =
secure it against spoofing (SIP UAs are generally not trusted). Would =
you believe a call made by Robert Mitnick from a prison phone would =
actually have a prison CPC in it's SIP headers :-) ?

(Mitnick is a widely known phone freak and network hacker who served a =
number of years in US prisons).

> Best regards,
> sebastien
>
> -----Message d'origine-----
> De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] De la part =

> de Tom Taylor Envoy=E9 : mardi 30 novembre 2004 16:07 =C0 : Jesske, R =
Cc :=20
> iptel@ietf.org; sip@ietf.org Objet : Re: [Iptel] Calling Party's=20
> Category
>
> The question was what the requirement is for this sort of indication.  =

> How is it used?
>
> Jesske, R wrote:
>> Dear all,
>> in the past there was a discussion regarding the specification of a=20
>> Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) What=20
>> is the actual status of this activity.
>> We are still interested in such kind of originating indication if the =

>> call/communication is coming from a normal SIP user or a SIP=20
>> -Payphone or SIP-Hotelphone.
>> Is there a interest from other parties in this issue?
>>
>> Best Regards
>>
>> Roland
>>
>> Deutsche Telekom AG
>> T-Com Zentrale
>> Roland Jesske, TE332-2
>> Section TE33; Signalling, Gateways and Switching Systems Am=20
>> Kavalleriesand 3, 64295 Darmstadt, Germany
>> Phone:  +49 6151 83-5940
>> Fax:      +49 6151 83-4577
>> email:   r.jesske@t-com.net
>>
>>
>>
>>
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>>
>>
>
> --
> Tom Taylor
> Carrier VoIP Standards Development
> Nortel Networks
> Phone +1 613 763 1496  (ESN 393-1496)
> E-mail: taylor@nortelnetworks.com
>
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sipping@ietf.org for new developments on the application of sip
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.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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His name is Kevin. Other than that I agree with you.

I'm interested in this area as well as I might be able to make use of thi=
s type of functionality in a hospitality industry scenario.=20

Regards,

B

http://www.freekevin.com/

Ben Gatewood
Streamdoor Ltd
114b Power Rd
London
W4 5PY
+44 20 7422 0439
+44 7835 136 187
bgatewood@streamdoor.net



-----Original Message-----
From: sip-bounces@ietf.org on behalf of David R Oran
Sent: Tue 11/30/2004 4:23 PM
To: GARCIN Sebastien RD-CORE-ISS
Cc: Jesske, R; iptel@ietf.org; Tom Taylor; sip@ietf.org
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
=20

On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:

> Hi,
>
> As an fixed line operator we also are interested in this feature and=20=

> would be happy that the IETF finally moves forward with this draft.=20
> This type of parameter is widely signalled in ISDN/PSTN networks. In=20=

> SIP (for example) it is required in context of both:
>
> - pure SIP networks (e.g. in case of a SIP payphone), and
> - interworking with legacy networks
>
> I think the uses case for this parameter will depend on the operator=20=

> service portfolio. For example, in France, the calling party's=20
> category parameter is used for the Call Return service.
>
Could you explain a bit about how this is used? I, for one would find=20
it helpful and possibly a motivating use case.

Also note that if this information is used for any security or=20
billing-sensitive operations, we have to deal with the issue of how to=20=

secure it against spoofing (SIP UAs are generally not trusted). Would=20
you believe a call made by Robert Mitnick from a prison phone would=20
actually have a prison CPC in it's SIP headers :-) ?

(Mitnick is a widely known phone freak and network hacker who served a=20=

number of years in US prisons).

> Best regards,
> sebastien
>
> -----Message d'origine-----
> De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] De la part=20=

> de Tom Taylor
> Envoy=E9 : mardi 30 novembre 2004 16:07
> =C0 : Jesske, R
> Cc : iptel@ietf.org; sip@ietf.org
> Objet : Re: [Iptel] Calling Party's Category
>
> The question was what the requirement is for this sort of indication. =20=

> How is it used?
>
> Jesske, R wrote:
>> Dear all,
>> in the past there was a discussion regarding the specification of a
>> Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) What=20=

>> is the actual status of this activity.
>> We are still interested in such kind of originating indication if the=20=

>> call/communication is coming from a normal SIP user or a SIP=20
>> -Payphone or SIP-Hotelphone.
>> Is there a interest from other parties in this issue?
>>
>> Best Regards
>>
>> Roland
>>
>> Deutsche Telekom AG
>> T-Com Zentrale
>> Roland Jesske, TE332-2
>> Section TE33; Signalling, Gateways and Switching Systems Am
>> Kavalleriesand 3, 64295 Darmstadt, Germany
>> Phone:  +49 6151 83-5940
>> Fax:      +49 6151 83-4577
>> email:   r.jesske@t-com.net
>>
>>
>>
>>
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>>
>>
>
> --
> Tom Taylor
> Carrier VoIP Standards Development
> Nortel Networks
> Phone +1 613 763 1496  (ESN 393-1496)
> E-mail: taylor@nortelnetworks.com
>
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>
> _______________________________________________
> 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
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.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
*************************************************************************=
*************
Disclaimer:

This electronic message may contain privileged or confidential informatio=
n. If you are not the intended recipient, be advised that any disclosure,=
 copying, distribution or use of this information is strictly prohibited.=
 If you are not the intended recipient, please notify the sender and dele=
te this message. Views expressed in this message are those of the individ=
ual sender, and are not necessarily the views of the Streamdoor Limited, =
unless otherwise stated.

Although Streamdoor Limited has taken reasonable precautions to ensure no=
 viruses are present in this email, the company cannot accept responsibil=
ity for any loss or damage arising from the use of this email or attachem=
ent

Employees of Streamdoor Limited and its affiliates are expressly required=
 not to make defamatory statements and not to infringe or authorise any i=
nfringements of copyrights or any other legal right by email communicatio=
ns. Any such communication is contrary to company policy and outside the =
scope of the employment of the individual concerned.

For further assistance on email policy, or if you have received this emai=
l in error, please contact  Streamdoor Group IT&C Helpdesk by email at it=
&c_helpdesk@streamdoor.net or write to  Streamdoor Ltd , Head of IT&C Dep=
artment , 114 Power Road , Chiswick, London , W4 5PY.

www.streamdoor.com
*************************************************************************=
*************
*************************************************************************=
*************
Disclaimer:

This electronic message may contain privileged or confidential informatio=
n. If you are not the intended recipient, be advised that any disclosure,=
 copying, distribution or use of this information is strictly prohibited.=
 If you are not the intended recipient, please notify the sender and dele=
te this message. Views expressed in this message are those of the individ=
ual sender, and are not necessarily the views of the Streamdoor Limited, =
unless otherwise stated.

Although Streamdoor Limited has taken reasonable precautions to ensure no=
 viruses are present in this email, the company cannot accept responsibil=
ity for any loss or damage arising from the use of this email or attachem=
ent

Employees of Streamdoor Limited and its affiliates are expressly required=
 not to make defamatory statements and not to infringe or authorise any i=
nfringements of copyrights or any other legal right by email communicatio=
ns. Any such communication is contrary to company policy and outside the =
scope of the employment of the individual concerned.

For further assistance on email policy, or if you have received this emai=
l in error, please contact  Streamdoor Group IT&C Helpdesk by email at it=
&c_helpdesk@streamdoor.net or write to  Streamdoor Ltd , Head of IT&C Dep=
artment , 114 Power Road , Chiswick, London , W4 5PY.

www.streamdoor.com
*************************************************************************=
*************

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<P><FONT SIZE=3D2>His name is Kevin. Other than that I agree with you.<BR=
>
<BR>
I'm interested in this area as well as I might be able to make use of thi=
s type of functionality in a hospitality industry scenario.<BR>
<BR>
Regards,<BR>
<BR>
B<BR>
<BR>
<A HREF=3D"http://www.freekevin.com/">http://www.freekevin.com/</A><BR>
<BR>
Ben Gatewood<BR>
Streamdoor Ltd<BR>
114b Power Rd<BR>
London<BR>
W4 5PY<BR>
+44 20 7422 0439<BR>
+44 7835 136 187<BR>
bgatewood@streamdoor.net<BR>
<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: sip-bounces@ietf.org on behalf of David R Oran<BR>
Sent: Tue 11/30/2004 4:23 PM<BR>
To: GARCIN Sebastien RD-CORE-ISS<BR>
Cc: Jesske, R; iptel@ietf.org; Tom Taylor; sip@ietf.org<BR>
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category<BR>
<BR>
<BR>
On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:<BR>
<BR>
&gt; Hi,<BR>
&gt;<BR>
&gt; As an fixed line operator we also are interested in this feature and=
<BR>
&gt; would be happy that the IETF finally moves forward with this draft.<=
BR>
&gt; This type of parameter is widely signalled in ISDN/PSTN networks. In=
<BR>
&gt; SIP (for example) it is required in context of both:<BR>
&gt;<BR>
&gt; - pure SIP networks (e.g. in case of a SIP payphone), and<BR>
&gt; - interworking with legacy networks<BR>
&gt;<BR>
&gt; I think the uses case for this parameter will depend on the operator=
<BR>
&gt; service portfolio. For example, in France, the calling party's<BR>
&gt; category parameter is used for the Call Return service.<BR>
&gt;<BR>
Could you explain a bit about how this is used? I, for one would find<BR>=

it helpful and possibly a motivating use case.<BR>
<BR>
Also note that if this information is used for any security or<BR>
billing-sensitive operations, we have to deal with the issue of how to<BR=
>
secure it against spoofing (SIP UAs are generally not trusted). Would<BR>=

you believe a call made by Robert Mitnick from a prison phone would<BR>
actually have a prison CPC in it's SIP headers :-) ?<BR>
<BR>
(Mitnick is a widely known phone freak and network hacker who served a<BR=
>
number of years in US prisons).<BR>
<BR>
&gt; Best regards,<BR>
&gt; sebastien<BR>
&gt;<BR>
&gt; -----Message d'origine-----<BR>
&gt; De : iptel-bounces@ietf.org [<A HREF=3D"mailto:iptel-bounces@ietf.or=
g">mailto:iptel-bounces@ietf.org</A>] De la part<BR>
&gt; de Tom Taylor<BR>
&gt; Envoy=E9 : mardi 30 novembre 2004 16:07<BR>
&gt; =C0 : Jesske, R<BR>
&gt; Cc : iptel@ietf.org; sip@ietf.org<BR>
&gt; Objet : Re: [Iptel] Calling Party's Category<BR>
&gt;<BR>
&gt; The question was what the requirement is for this sort of indication=
=2E&nbsp;<BR>
&gt; How is it used?<BR>
&gt;<BR>
&gt; Jesske, R wrote:<BR>
&gt;&gt; Dear all,<BR>
&gt;&gt; in the past there was a discussion regarding the specification o=
f a<BR>
&gt;&gt; Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) =
What<BR>
&gt;&gt; is the actual status of this activity.<BR>
&gt;&gt; We are still interested in such kind of originating indication i=
f the<BR>
&gt;&gt; call/communication is coming from a normal SIP user or a SIP<BR>=

&gt;&gt; -Payphone or SIP-Hotelphone.<BR>
&gt;&gt; Is there a interest from other parties in this issue?<BR>
&gt;&gt;<BR>
&gt;&gt; Best Regards<BR>
&gt;&gt;<BR>
&gt;&gt; Roland<BR>
&gt;&gt;<BR>
&gt;&gt; Deutsche Telekom AG<BR>
&gt;&gt; T-Com Zentrale<BR>
&gt;&gt; Roland Jesske, TE332-2<BR>
&gt;&gt; Section TE33; Signalling, Gateways and Switching Systems Am<BR>
&gt;&gt; Kavalleriesand 3, 64295 Darmstadt, Germany<BR>
&gt;&gt; Phone:&nbsp; +49 6151 83-5940<BR>
&gt;&gt; Fax:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; +49 6151 83-4577<BR>
&gt;&gt; email:&nbsp;&nbsp; r.jesske@t-com.net<BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;&gt; _______________________________________________<BR>
&gt;&gt; Iptel mailing list<BR>
&gt;&gt; Iptel@ietf.org<BR>
&gt;&gt; <A HREF=3D"https://www1.ietf.org/mailman/listinfo/iptel">https:/=
/www1.ietf.org/mailman/listinfo/iptel</A><BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;<BR>
&gt; --<BR>
&gt; Tom Taylor<BR>
&gt; Carrier VoIP Standards Development<BR>
&gt; Nortel Networks<BR>
&gt; Phone +1 613 763 1496&nbsp; (ESN 393-1496)<BR>
&gt; E-mail: taylor@nortelnetworks.com<BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; Iptel mailing list<BR>
&gt; Iptel@ietf.org<BR>
&gt; <A HREF=3D"https://www1.ietf.org/mailman/listinfo/iptel">https://www=
1.ietf.org/mailman/listinfo/iptel</A><BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; Sip mailing list&nbsp; <A HREF=3D"https://www1.ietf.org/mailman/list=
info/sip">https://www1.ietf.org/mailman/listinfo/sip</A><BR>
&gt; This list is for NEW development of the core SIP Protocol<BR>
&gt; Use sip-implementors@cs.columbia.edu for questions on current sip<BR=
>
&gt; Use sipping@ietf.org for new developments on the application of sip<=
BR>
&gt;<BR>
David R. Oran<BR>
Cisco Fellow<BR>
Cisco Systems<BR>
7 Ladyslipper Lane<BR>
Acton, MA 01720 USA<BR>
Tel: +1 978 264 2048<BR>
Email: oran@cisco.com<BR>
<BR>
<BR>
_______________________________________________<BR>
Sip mailing list&nbsp; <A HREF=3D"https://www1.ietf.org/mailman/listinfo/=
sip">https://www1.ietf.org/mailman/listinfo/sip</A><BR>
This list is for NEW development of the core SIP Protocol<BR>
Use sip-implementors@cs.columbia.edu for questions on current sip<BR>
Use sipping@ietf.org for new developments on the application of sip<BR>
*************************************************************************=
*************<BR>
Disclaimer:<BR>
<BR>
This electronic message may contain privileged or confidential informatio=
n. If you are not the intended recipient, be advised that any disclosure,=
 copying, distribution or use of this information is strictly prohibited.=
 If you are not the intended recipient, please notify the sender and dele=
te this message. Views expressed in this message are those of the individ=
ual sender, and are not necessarily the views of the Streamdoor Limited, =
unless otherwise stated.<BR>
<BR>
Although Streamdoor Limited has taken reasonable precautions to ensure no=
 viruses are present in this email, the company cannot accept responsibil=
ity for any loss or damage arising from the use of this email or attachem=
ent<BR>
<BR>
Employees of Streamdoor Limited and its affiliates are expressly required=
 not to make defamatory statements and not to infringe or authorise any i=
nfringements of copyrights or any other legal right by email communicatio=
ns. Any such communication is contrary to company policy and outside the =
scope of the employment of the individual concerned.<BR>
<BR>
For further assistance on email policy, or if you have received this emai=
l in error, please contact&nbsp; Streamdoor Group IT&amp;C Helpdesk by em=
ail at it&amp;c_helpdesk@streamdoor.net or write to&nbsp; Streamdoor Ltd =
, Head of IT&amp;C Department , 114 Power Road , Chiswick, London , W4 5P=
Y.<BR>
<BR>
www.streamdoor.com<BR>
*************************************************************************=
*************<BR>
</FONT>
</P>

*************************************************************************=
*************<br>Disclaimer:<br><br>This electronic message may contain p=
rivileged or confidential information. If you are not the intended recipi=
ent, be advised that any disclosure, copying, distribution or use of this=
 information is strictly prohibited. If you are not the intended recipien=
t, please notify the sender and delete this message. Views expressed in t=
his message are those of the individual sender, and are not necessarily t=
he views of the Streamdoor Limited, unless otherwise stated.<br><br>Altho=
ugh Streamdoor Limited has taken reasonable precautions to ensure no viru=
ses are present in this email, the company cannot accept responsibility f=
or any loss or damage arising from the use of this email or attachement<b=
r><br>Employees of Streamdoor Limited and its affiliates are expressly re=
quired not to make defamatory statements and not to infringe or authorise=
 any infringements of copyrights or any other legal right by email commun=
ications. Any such communication is contrary to company policy and outsid=
e the scope of the employment of the individual concerned.<br><br>For fur=
ther assistance on email policy, or if you have received this email in er=
ror, please contact Streamdoor Group IT&C Helpdesk by email at it&c_helpd=
esk@streamdoor.net or write to Streamdoor Ltd , Head of IT&C Department ,=
 114 Power Road , Chiswick, London , W4 5PY.<br><br>www.streamdoor.com<br=
>************************************************************************=
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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
--===============1756058855==--



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From: "Johnston, Alan" <Alan.Johnston@mci.com>
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Subject: [Sip] FW: XCON Interim Attendance Poll - please respond
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FYI

-----Original Message-----
From: xcon-bounces@ietf.org [mailto:xcon-bounces@ietf.org] On Behalf Of
Johnston, Alan
Sent: Tuesday, November 23, 2004 2:21 PM
To: xcon@ietf.org
Cc: 'Adam@nostrum.com'
Subject: [XCON] XCON Interim Attendance Poll - please respond


All,

The XCON working group interim meeting is being planned for January 5 and
January 6, 2005 in Boston, MA.

The exact location and agenda with hotel information will be posted in about
a week, as will the official announcement on the IETF Announce list.

For the moment, if you plan to attend, or know of people in your
organization who plan to attend, please send an email directly to me
(mailto:alan.johnston@mci.com) (do *not* send to the list) so we can get an
accurate count to finalize the planning.

Please respond in the next week, by November 30.

Thanks,
Alan Johnston
co-chair XCON

_______________________________________________
XCON mailing list
XCON@ietf.org
https://www1.ietf.org/mailman/listinfo/xcon

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Adam Roach'" <adam@nostrum.com>
Subject: RE: [Sip] Re: RLS and identity
Date: Tue, 30 Nov 2004 18:04:33 -0000
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This creates a number of problems for 3GPP which I think need to be discussed further. I also think the current IESG diktat needs fleshing out with more words on the issues they have with a wider specification of available methods, as I don't believe I have seen that on the list yet. 

Adam is also making generalisations when he states "inherently insecure P-Asserted-Identity mechanism". Noone has made any justification of this statement. I accept that P-Asserted-Identity has a restricted applicability, in that it needs a trust domain, and needs to fulfil a number of criteria, as listed by the template at the back of RFC 3325, but that does not make it insecure. Any mechanism is insecure if you do not follow the rules.

Back to the problems. Identity is not ready yet. Yet 3GPP needs RLS in release 6. Creating a normative dependency on identity means that RLS cannot get out in due time.

Moreover, to use identity in 3GPP, we need to examine the interdependence of P-Asserted-Identity and the identity draft, as they need to coexist. We are not in a position to just rip one out and replace it with another, and any suggestion that this should be done is an expression of bad faith on behalf of IETF from when RFC 3325 was created. Identity still only solves half the issues that 3GPP use P-Asserted-Identity to solve. The author of the identity draft appears to have accepted the need to study and document this interdependency, as it is one of the identified open issues for completion of that draft.

Once we have done that, I am perfectly prepared to look at specifying use of identity in 3GPP release 7, but we need to solve the problem of allowing release 6 implementations to have an RLS and that are conformant with IETF RFCs.

I do not want to have to get to the position of 3GPP having to specify something that is not IETF compliant, but that seems to be where they are being driven at the moment.

regards

Keith

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


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
> Adam Roach
> Sent: 25 November 2004 03:18
> To: Adam Roach
> Cc: SIP WG; Aki Niemi
> Subject: Re: [Sip] Re: RLS and identity
> 
> 
> Adam Roach wrote:
> 
> > Based on Rohan's suggestion, the text will effectively say:
> >
> > - Jon's Identity draft will be mandatory to implement, 
> optional to use.
> >
> > - Other mechanisms that have properties such that they can 
> adequately
> >   convey the identity of the subscriber and the permission 
> of the RLS
> >   to subscribe on the user's behalf can also be used.
> 
> As a clarification, I have received specific guidance from the area 
> directors that the draft cannot contain anything about such alternate 
> mechanisms, even if they are not specifically mentioned by 
> name. The use 
> of e.g. P-Asserted-Identity will need to be a modification that 3GPP 
> specifically calls out relative to the draft.
> 
> Of course, since the identity work is basically done and 
> works so much 
> better than P-Asserted-Identity, there could be a very valid argument 
> made that 3GPP R6 should abandon the inherently insecure 
> P-Asserted-Identity mechanism in favor of the 
> cryptographically secure 
> SIP Identity mechanism.
> 
> /a
> 
> _______________________________________________
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> 

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Drage, Keith (Keith) wrote:

>Adam is also making generalisations when he states "inherently insecure P-Asserted-Identity mechanism". Noone has made any justification of this statement. I accept that P-Asserted-Identity has a restricted applicability, in that it needs a trust domain, and needs to fulfil a number of criteria, as listed by the template at the back of RFC 3325, but that does not make it insecure. Any mechanism is insecure if you do not follow the rules.
>  
>

I consider any network that has security properties of "crunchy security 
shell with soft, easily compromised filling" to be inherently insecure 
-- a single compromise anywhere in the system can be used to exploit the 
entire system; in 3GPP, this weakness can even be exploited across 
domains. Any network architecture with such a design guarantees that the 
security of any node in the system is no better than the security of the 
least secure node in the system. For sufficiently large systems, this 
degenerates to the same properties as little or no security, since the 
probability of at least one node being compromisable goes up 
geometrically with the number of nodes in the system [1].

Since P-Asserted-Identity works only in the framework of such systems, 
and relies exclusively on the crunchy shell for protection, I consider 
it inherently insecure.

/a

[1] For example, if you have a system with 2,000 nodes, each
    of which has a 99.9% chance of having been designed and
    configured to have appropriate security properties, there
    is still an 86.5% chance that at least one of those nodes
    will be exploitable, meaning that there is an 86.5% chance
    that the entire system can be exploited.
    (99.9% ^ 2000 = 13.5%)

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Subject: [Sip] Any talk of an Interim for SIP or SIPPING?
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Hey

Is there any serious talk about having an interim meeting for SIP or SIPPING?

Just trying to plan travel schedules in advance, and say the XCON 
announcement (knowing it,SIP, SIPPING and SIMPLE were thrown in together 
for one super meeting last May).

cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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To: David R Oran <oran@cisco.com>
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
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David R Oran wrote:

>
> On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:
>
>> As an fixed line operator we also are interested in this feature and 
>> would be happy that the IETF finally moves forward with this draft. 
>> This type of parameter is widely signalled in ISDN/PSTN networks. In 
>> SIP (for example) it is required in context of both:
>>
>> - pure SIP networks (e.g. in case of a SIP payphone), and
>> - interworking with legacy networks
>>
>> I think the uses case for this parameter will depend on the operator 
>> service portfolio. For example, in France, the calling party's 
>> category parameter is used for the Call Return service.
>>
> Could you explain a bit about how this is used? I, for one would find 
> it helpful and possibly a motivating use case.


I'm equally confused. I'm familiar with the calling party category in 
ISUP, which works only because of the walled-garden nature of the SS7 
network. Without that, there are really two major factors around this 
which need to be characterized:

   1. Who is authorized to assert a particular calling party category?
      This is much trickier than the identity problem; with identity, we
      can trust that a domain is the authority for the use of user names
      in that domain. For calling party category, there is no such
      relationship. For example, it would probably not be valid for a
      domain of "adamroach.com" to assert a calling party category of
      "priority," "operator," or "police."
   2. Under which circumstances is the calling party category required
      to be present? Is there some architectural mechanism that can be
      used to enforce this? (We don't need to define it, but there
      should at least be some idea about whether it's possible). If not,
      then there is no purpose in coming up with a protocol mechanism
      for conveying such information.

/a

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In a message dated 11/30/2004 11:49:18 AM Eastern Standard Time, 
oran@cisco.com writes:


> Also note that if this information is used for any security or 
> billing-sensitive operations, we have to deal with the issue of how to 
> secure it against spoofing (SIP UAs are generally not trusted). Would 
> you believe a call made by Robert Mitnick from a prison phone would 
> actually have a prison CPC in it's SIP headers :-) ?
> 

Of course, this shows a basic difference in the way this capability is 
handled in SIP vs today's telephone network. It's an important thing to deal with, 
since many of these things are mandated by law. For example, I believe the 
requirement to identify calls from a prison as such is mandated.

In today's network, in order to provide assurance that the call is properly 
marked, not just for CPC but calling party ID as well, the phone in the prison 
is rnot esponsible for the marking. The switch outside the prison marks any 
call that comes over that pair of wires. Robert Mitnick had no opportunity to 
change that marking.

The issue for SIP is to provide the same level of assurance for such 
functions. I'm sure anxious to hear a resolution to this problem, since it keeps 
coming back. CPC is not unique in this regard, so a general solution is needed.

Mike Pierce
Artel


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/30/2004 11:49:18 AM Eastern Standard Time, oran@cisco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Also note that if this info=
rmation is used for any security or=20
<BR>billing-sensitive operations, we have to deal with the issue of how to=20
<BR>secure it against spoofing (SIP UAs are generally not trusted). Would=20
<BR>you believe a call made by Robert Mitnick from a prison phone would=20
<BR>actually have a prison CPC in it's SIP headers :-) ?
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>Of course, this shows a basic difference in the way this capability is h=
andled in SIP vs today's telephone network. It's an important thing to deal=20=
with, since many of these things are mandated by law. For example, I believe=
 the requirement to identify calls from a prison as such is mandated.
<BR>
<BR>In today's network, in order to provide assurance that the call is prope=
rly marked, not just for CPC but calling party ID as well, the phone in the=20=
prison is rnot esponsible for the marking. The switch outside the prison mar=
ks any call that comes over that pair of wires. Robert Mitnick had no opport=
unity to change that marking.
<BR>
<BR>The issue for SIP is to provide the same level of assurance for such fun=
ctions. I'm sure anxious to hear a resolution to this problem, since it keep=
s coming back. CPC is not unique in this regard, so a general solution is ne=
eded.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR></FONT></HTML>

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Thread-Topic: OPTIONS outside dialog
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From: "Anil Bollineni" <ABollineni@juniper.net>
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Hi,

    Assume OPTIONS comes outside dialog, what are the possible responses
that are for OPTIONS, assume these are the same responses for any
request as defined in RFC 3261. Please clarify.

=20

Thanks in advance,

Anil

=20


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp; Assume OPTIONS comes outside =
dialog, what
are the possible responses that are for OPTIONS, assume these are the =
same
responses for any request as defined in RFC 3261. Please =
clarify.</span></font></p>

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

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
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font-family:Arial'>Anil</span></font></p>

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From sip-bounces@ietf.org  Tue Nov 30 19:56:03 2004
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Adam Roach'" <adam@nostrum.com>, David R Oran <oran@cisco.com>
Subject: RE: [Sip] RE: [Iptel] Calling Party's Category
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I view these sorts of security attributes as elements of trait-based
authorization. The calling party's category requires the sort of
pre-arrangement and federation to which something like SAML naturally lends
itself.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Tuesday, November 30, 2004 9:27 PM
> To: David R Oran
> Cc: Jesske, R; iptel@ietf.org; Tom Taylor; sip@ietf.org
> Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
> 
> 
> David R Oran wrote:
> 
> >
> > On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:
> >
> >> As an fixed line operator we also are interested in this 
> feature and 
> >> would be happy that the IETF finally moves forward with 
> this draft. 
> >> This type of parameter is widely signalled in ISDN/PSTN 
> networks. In 
> >> SIP (for example) it is required in context of both:
> >>
> >> - pure SIP networks (e.g. in case of a SIP payphone), and
> >> - interworking with legacy networks
> >>
> >> I think the uses case for this parameter will depend on 
> the operator 
> >> service portfolio. For example, in France, the calling party's 
> >> category parameter is used for the Call Return service.
> >>
> > Could you explain a bit about how this is used? I, for one 
> would find 
> > it helpful and possibly a motivating use case.
> 
> 
> I'm equally confused. I'm familiar with the calling party category in 
> ISUP, which works only because of the walled-garden nature of the SS7 
> network. Without that, there are really two major factors around this 
> which need to be characterized:
> 
>    1. Who is authorized to assert a particular calling party category?
>       This is much trickier than the identity problem; with 
> identity, we
>       can trust that a domain is the authority for the use of 
> user names
>       in that domain. For calling party category, there is no such
>       relationship. For example, it would probably not be valid for a
>       domain of "adamroach.com" to assert a calling party category of
>       "priority," "operator," or "police."
>    2. Under which circumstances is the calling party category required
>       to be present? Is there some architectural mechanism that can be
>       used to enforce this? (We don't need to define it, but there
>       should at least be some idea about whether it's 
> possible). If not,
>       then there is no purpose in coming up with a protocol mechanism
>       for conveying such information.
> 
> /a
> 
> _______________________________________________
> 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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