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Date: Mon, 02 Jun 2003 10:03:51 +0800
From: Prasanna Venkatesh <prasanna@huawei.com>
Subject: RE: [Sip] Why do we need 3-way handshake for INVITE transaction....
In-reply-to: <006501c326e8$f7622b60$5d02120a@in.huawei.com>
To: "Nataraju A.B." <natarajuab@huawei.com>,
        Arjun Roychowdhury <aroychow@hns.com>,
        Vishal Phirke <vishal_phirke@nmss.com>
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Hi,
	One of the simplest policies could be accept the first 200.  After the UA
receives the first 200, it sends a ACK+BYE for all subsequent 200 responses.
	However for a very intelligent UA (C&s) pair, it could use Call-Info to
determine the selection, the call-info being a part of the 200 response. The
call-info could either containing a link to a URL on reaching which the UAC
could determine the type of callee or it could be a business-card using
which it could determien the type of callee.
	There are crude ways of doing this also, like using say the URL or display
name of the Contact, though it is not advisable.

But I do not know a clean way where the callee mentions to the caller the
kind of device it is or the identity of the person answering the call (as
the display name of the To is also fixed by the UAC).  Any solution to do
this?

Cheers,
Prasanna

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
Nataraju A.B.
Sent: Saturday, May 31, 2003 4:21 AM
To: Arjun Roychowdhury; Vishal Phirke
Cc: sip@ietf.org; sip-admin@ietf.org; Sip-implementors@cs.columbia.edu
Subject: Re: [Sip] Why do we need 3-way handshake for INVITE
transaction....



----- Original Message -----
From: "Arjun Roychowdhury" <aroychow@hns.com>
To: "Vishal Phirke" <vishal_phirke@nmss.com>
Cc: "Nataraju A.B." <natarajuab@huawei.com>; <sip@ietf.org>;
<sip-admin@ietf.org>
Sent: Saturday, May 31, 2003 1:27 AM
Subject: Re: [Sip] Why do we need 3-way handshake for INVITE transaction....


> On 05/30/2003 03:03:53 PM, "Vishal Phirke" wrote:
>
> > 2. Invites can be forked. And if multiple destinations send
> > 200 OK, you
> > don't want to end up with multiple connections. Instead, 3-way
> > handshake
> > allows the decision of accepting one of them delayed till Ack
> > is sent by
> > the originator of the request.
> >
>
> I dont think this is a valid reason for 3-way handshakes. By definition,
> all 200 OKs are proxied back. Its upto the caller UA
> to then send BYE/OK to callees that it does not want to talk to due to
> forking. Therefore, the ACK is sent to all of them and doesnt
> help in delaying the decision.

[ABN] what could be the logic which the application might use to reject the
responses,
I mean which one out of 'n' 200-OK response for INVITE will be accepted and
rest all will be rejected.
The application only intended to contact the calle, but it does not have any
information about the callee information (which one to select).

will this be driven by a local policy ????

Regards,
-Nataraju A.B.

>
> regds
> arjun
>
> _______________________________________________
> 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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_______________________________________________
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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: "Christer Holmberg (LMF)" <Christer.Holmberg@lmf.ericsson.se>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: Sanjay Sinha <sanjsinh@cisco.com>, sip@ietf.org
Subject: RE: [Sip] Question on RFC 3263
Date: Mon, 2 Jun 2003 07:33:27 +0200 
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Hi Jonathan,

>>>but in any case, the UAS shouldnt terminate the call just because
>>>the 18x never got a PRACK.
>> 
>>What if the UAS includes an SDP in the reliable 18x? Shall it
>>assume that the UAC will terminate the call if it doesn't receive
>>an SDP when it expect to do so?
> 
>Why would the UAC expect an SDP?

Eg if the INVITE did not contain SDP.

Also, assume we have a pre-condition scenario, where multiple offer/answer transactions are needed before the called party is even alerted. The pre-condition setup will be "halted" if SDPs aren't received. Now, I don't know if pre-condition setups really require SDPs to be sent in specific messages, but at least the nodes expect to receive SDPs, so then the question is how long they shall "wait" for them before taking actions (ie releasing the setup).

>In any case, as I said genreally a call is not terminated because a 18x never gets a prack.

I didn't see the word "generally" :) Anyway, I think (and I guess that's what you meant too) nodes have make that decission case-by-case. After all, the major reason PRACK is used in the first place is to make sure that 18x provisional responses really DO reach the UAC. If one didn't "care", or if the information (SDP, encapslulated ISUP... whatever) carried in the 18x response had no meaning for the session, why use PRACK in the first place?

Regards,

Christer Holmberg
Ericsson Finland




> 
> -Jonathan R.
> 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Scientist                             Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From: "Shane McCormack" <smccormack@tssg.org>
To: <sip@ietf.org>, <sipping@ietf.org>
Date: Tue, 27 May 2003 14:50:08 +0100
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Subject: [Sip] Number of registered UAs
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HI all,

Is it possible to calculate how many UA's a particular user in the network
has either through registration details or otherwise?

Regards
Shane

Shane McCormack

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

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




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



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Date: Tue, 27 May 2003 10:49:39 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sip] RE: [Sipping] RE: SIP mobility in WLAN
In-reply-to: <Pine.GSO.4.53.0305270656150.1065@uni03du.unity.ncsu.edu>
To: "'Prabhas Ranjan Sinha'" <prsinha@unity.ncsu.edu>,
        "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>
Cc: "'Santharam, Arun [NTWK SVCS]'" <asanth01@sprintspectrum.com>,
        "'Rajarshi Chakraborty'" <rajarshi_chakraborty@hotmail.com>,
        drage@lucent.com, ashishn@mahindrabt.com, sip@ietf.org,
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Please check www.research.telcordia.com/sip-mobile and

 http://www.cs.columbia.edu/~hgs/papers/Schu0007_Application.pdf

Thanks, Henry

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Prabhas Ranjan Sinha
> Sent: Tuesday, May 27, 2003 6:00 AM
> To: Henry Sinnreich
> Cc: 'Santharam, Arun [NTWK SVCS]'; 'Rajarshi Chakraborty'; 
> drage@lucent.com; ashishn@mahindrabt.com; sip@ietf.org; 
> sipping@ietf.org
> Subject: Re: [Sip] RE: [Sipping] RE: SIP mobility in WLAN
> 
> 
> 
> >
> > The other alternative is to have application layer mobility 
> for SIP as 
> > published in several papers at Columbia University and Telcordia. I 
> > don't believe you need mobile IP for all scenarios or that 
> is the best 
> > option under all circumstances.
> >
> > The last word on this may not be over yet.
> >
> > Henry
> 
> Henry,
> Could you please elaborate on it more w.r.t pros and cons or 
> refer me to to any publication which can talk about it ? I 
> have read that people forsee that Mobile IP could be the most 
> suitable mechanism for mobility across multiple subnets of 
> WLAN network.
> 
> Prabhas
> 
> 
> 
> 
> 
> 
> >
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On 
> > > Behalf Of Santharam, Arun [NTWK SVCS]
> > > Sent: Monday, May 19, 2003 1:35 PM
> > > To: Rajarshi Chakraborty; drage@lucent.com; 
> ashishn@mahindrabt.com; 
> > > sip@ietf.org
> > > Cc: sipping@ietf.org
> > > Subject: [Sipping] RE: SIP mobility in WLAN
> > >
> > >
> > >
> > > If we follow the SIP implementation on Mobile IP then the 
> SIP stack 
> > > should not have to bother about the mobility aspect. The 
> SIP stack 
> > > should be notified of IP changes/handoff information (if relevant 
> > > for the SIP stack) by the underlying stack. This may prompt a SIP 
> > > registration and subsequent call flows.
> > >
> > > My 2 cents ...
> > > arun
> > >
> > >
> > > > -----Original Message-----
> > > > From: Rajarshi Chakraborty 
> > > > [mailto:rajarshi_chakraborty@hotmail.com]
> > > > Sent: Sunday, May 18, 2003 11:42 PM
> > > > To: drage@lucent.com; ashishn@mahindrabt.com; Santharam, Arun
> > > > [NTWK SVCS]; sip@ietf.org
> > > > Cc: sipping@ietf.org
> > > > Subject: SIP mobility in WLAN
> > > >
> > > >
> > > > Hi,
> > > >
> > > > Where can I get call-flows for SIP session management in a 
> > > > wireless LAN (802.11)? 3GPP only addresses the issues 
> for cellular
> > > > networks. Or are they
> > > > too trivial to be addressed specially? I am specially 
> interested in
> > > > situations of handoffs (in this case, the IP changes as soon
> > > > as the host
> > > > shifts from the zone around one hotspot to another). Should
> > > > SIP bother at
> > > > all to detect IP changes or should the control be completely
> > > > outside SIP? I
> > > > mean should it be some other module that has the
> > > > responsibility of informing
> > > > the SIP stack in the mobile host about the IP changes, so
> > > > that the host may
> > > > then send a re-INVITE with a new SDP but the call still
> > > > remains active?
> > > >
> > > > regards
> > > > Rajarshi
> > > >
> > > >
> > > > Rajarshi Chakraborty
> > > > B-196, IIT Campus,
> > > > Kharagpur,
> > > > West Bengal, India
> > > > PIN-721302
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > ----Original Message Follows----
> > > > From: "Drage, Keith (Keith)" <drage@lucent.com>
> > > > To: "'Ashish Naik'" <ashishn@mahindrabt.com>,   "Santharam,
> > > > Arun [CC]"
> > > > <asanth01@sprintspectrum.com>,   "Sip@Ietf. Org" <sip@ietf.org>
> > > > Subject: RE: [Sip] Session Mobility in SIP
> > > > Date: Thu, 8 May 2003 14:42:24 +0100
> > > >
> > > > Your best starting point to see the architecture is 3GPP TS 
> > > > 23.228, with reference to 3GPP TS 23.002 to see how 
> these entities 
> > > > fit together into the
> > > > entire 3GPP architecture.
> > > >
> > > > The SIP protocol application is covered in 3GPP TS 24.229 with 
> > > > example flows in 3GPP TS 24.228.
> > > >
> > > > These documents may be found at:
> > > >
> > > > ftp://ftp.3gpp.org/Specs/latest/Rel-5/
> > > > <ftp://ftp.3gpp.org/Specs/latest/Rel-5/>
> > > >
> > > > 3GPP uses these specifications over GPRS to provide mobility.
> > > >
> > > > There is a parallel set of 3GPP2 specifications, 
> substantially of 
> > > > common text, that uses the same architecture over mobile IP, 
> > > > instead of GPRS. These
> > > > documents are currently in 3GPP2 comment and review.
> > > >
> > > > regards
> > > >
> > > > Keith
> > > >
> > > >
> > > >
> > > >
> > > > Keith Drage
> > > > Lucent Technologies
> > > > Tel: +44 1793 776249
> > > > Email: drage@lucent.com
> > > >
> > > > -----Original Message-----
> > > > From: Ashish Naik [mailto:ashishn@mahindrabt.com]
> > > > Sent: 08 May 2003 06:23
> > > > To: Santharam, Arun [CC]; Sip@Ietf. Org
> > > > Subject: RE: [Sip] Session Mobility in SIP
> > > >
> > > >
> > > > ok. Is there any specific document on IETF and 3GPP that can 
> > > > explain how it is being addressed  ?
> > > >
> > > >
> > > > Ragards,
> > > > Ashish Naik
> > > > CoE Mobile Computing
> > > > -----------------------------------
> > > > Phone 4018100 Ext 3070
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On 
> Behalf Of 
> > > > Santharam, Arun [CC]
> > > > Sent: Wednesday, April 09, 2003 9:42 PM
> > > > To: Ashish Naik; Sip@Ietf. Org
> > > > Subject: RE: [Sip] Session Mobility in SIP
> > > >
> > > >
> > > >
> > > > Ashish,
> > > >
> > > >
> > > >
> > > > Please find comments inline.
> > > >
> > > >
> > > >
> > > > Thank you
> > > >
> > > > arun
> > > >
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: Ashish Naik [mailto:ashishn@mahindrabt.com]
> > > > Sent: Wednesday, April 09, 2003 4:01 AM
> > > > To: Sip@Ietf. Org
> > > > Subject: [Sip] Session Mobility in SIP
> > > >
> > > >
> > > >
> > > > How is SIP going to address session mobility ? Is there any 
> > > > initiative taken up for this ?
> > > >
> > > >
> > > >
> > > >  > This has been addressed in 3GPP standards. Please 
> refer to the 
> > > > 3GPP All IP architecture specifications.
> > > >
> > > >
> > > >
> > > > SIP is said to be a better alternative than Mobile IP. 
> Can I have 
> > > > some latest development document references ?
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >  >  SIP is not going to replace Mobile IP. It is going 
> to leverage 
> > > > Mobile IP for session management in 3G wireless networks.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Regards,
> > > > Ashish Naik
> > > > -------------------
> > > > CoE Mobile Computing
> > > > Phone: 4018100 Ext 3070
> > > >
> > > >
> > > >
> > > >
> > > > *********************************************************
> > > > Disclaimer
> > > >
> > > > This message (including any attachments) contains confidential 
> > > > information intended for a specific individual and 
> purpose, and is 
> > > > protected by law. If you are not the intended recipient, you 
> > > > should delete this message and are hereby notified that any 
> > > > disclosure, copying, or distribution of this message, or the 
> > > > taking of any action based on it, is strictly prohibited.
> > > >
> > > > *********************************************************
> > > > Visit us at http://www.mahindrabt.com
> > > >
> > > >
> > > >
> > > >
> > > > *********************************************************
> > > > Disclaimer
> > > >
> > > > This message (including any attachments) contains confidential 
> > > > information intended for a specific individual and 
> purpose, and is 
> > > > protected by law. If you are not the intended recipient, you 
> > > > should delete this message and are hereby notified that any 
> > > > disclosure, copying, or distribution of this message, or the 
> > > > taking of any action based on it, is strictly prohibited.
> > > >
> > > > *********************************************************
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> > > >
> > > > 
> _________________________________________________________________
> > > > Mobile, masti, magic! Cool ringtones & logos. 
> > > > http://www.msn.co.in/mobile/ Get noticed
> > > >
> > > >
> > > _______________________________________________
> > > Sipping mailing list  
> https://www1.ietf.org/mailman/listi> nfo/sipping
> > > This list 
> is for NEW development of the 
> application of SIP Use 
> > > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > > sip@ietf.org for new developments of core 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
> >
> 
> Prabhas
> +++++++++++++
> CSEL Technology Enterprise Programme
> London Business School
> Baker Street, London NW1 6DD
> *****************************************
> There is no wisdom greater than kindness.
> *****************************************
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Mon Jun  2 11:29:42 2003
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From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Question on RFC 3263
Date: Mon, 2 Jun 2003 11:25:28 -0400
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> >> Actually, before that even happens the client 
> >> transaction will simply terminate at the UAC. See:
> >> 
> >> 
> http://www.ietf.org/internet-drafts/draft-sparks-sip-noninvite-00.txt
> >> 
> >> 
> >> but in any case, the UAS shouldnt terminate the 
> >> call just because the 18x never got a PRACK.
> > 
> > 
> > What if the UAS includes an SDP in the reliable 18x? 
> > Shall it assume that the UAC will terminate the call 
> > if it doesn't receive an SDP when it expect to do so?
> 
> Why would the UAC expect an SDP? 

The following is from RFC 3262 section 5:

  "If the UAC receives a reliable provisional 
   response with an offer (this would occur if 
   the UAC sent an INVITE without an offer, in
   which case the first reliable provisional 
   response will contain the offer), it MUST 
   generate an answer in the PRACK.  If the 
   UAC receives a reliable provisional response 
   with an answer, it MAY generate an additional 
   offer in the PRACK.  If the UAS receives a 
   PRACK with an offer, it MUST place the answer 
   in the 2xx to the PRACK."

> In any case, as I said genreally a call is not 
> terminated because a 18x never gets a prack.

The following is from RFC 3262 section 3:

  "If a reliable provisional response is 
   retransmitted for 64*T1 seconds without 
   reception of a corresponding PRACK, the 
   UAS SHOULD reject the original request 
   with a 5xx response."

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From mailnull@www1.ietf.org  Mon Jun  2 12:55:42 2003
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Prabhas Ranjan Sinha'"
	 <prsinha@unity.ncsu.edu>,
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Subject: RE: [Sip] WLAN & SIP
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To clarify what 3GPP is currently doing.

1)	3GPP is looking at the integration of Wireless LAN and 3GPP systems. These do not change the wireless LAN specifications, or give indications on how thet are put together. That work is divided into a number of scenarios, each of which has a different degree of integration. They are more fully described in ftp://ftp.3gpp.org/Specs/latest/Rel-6/22_series/22934-610.zip
Scenario 1 requires no specification work, and otherwise only scenario 2 and 3 will be supported at release 6.

2)	3GPP is taking its existing IM CN subsystem specifications, which include its SIP documents, and making some of them more access independent in release 6. This means principally changing terminology rather than technical requirements. There will need to be some extensions for the 3GPP defined bits of some P-headers to cover other access mechanisms.

In conclusion 3GPP is not really defining SIP over wireless LAN, rather assuming that it will work, and if it is used, it can be used to access the IM CN subsystem of 3GPP.

regards

Keith

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

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 27 May 2003 17:09
> To: 'Prabhas Ranjan Sinha'; sip@ietf.org
> Subject: RE: [Sip] WLAN & SIP
> 
> 
> 
> Prabhas asked:
> > 
> > Is there any draft on SIP usage in WLAN arena ?
> > Any help would be appreciated.
> > 
> 
> I suppose "Just use it -- we don't really care whether the IP 
> traffic is
> over WLAN, ethernet, frame relay, barbed wired, or avian 
> carrier" isn't the
> answer you're looking for. But really, the IETF (outside of 
> task specific
> areas like the Performance Implications of Link Characteistics working
> group) isn't generally concerned with specific lower-layer aspects.
> 
> The short answer is that SIP works just fine over 802.11b 
> (I'm using it that
> way right now) and should work over "a" and "g" just fine too.
> 
> 3GPP (www.3gpp.org) is doing some work on extending IMS 
> Release 6 (Internet
> Multimedia Subsystem) which uses SIP onto WLAN. However, I'm not yet
> completely certain that what they mean by WLAN and what you 
> mean by WLAN are
> the same thing -- they seem to be mostly worred about the 
> sort of WLAN that
> would be installed by mobile operators in hotspots and tied 
> to one's mobile
> phone account. Consequently, they're interested in things 
> like using the
> mobile's SIM card for authentication over WLAN, differential 
> access pricing
> by service, Q0S interworking with UMTS, and all of the usual 
> 3G-mobile-phone
> type things.
> 
> --
> Dean
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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--NextPart

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

	Title		: A Mechanism for Content Indirection in Session 
                          Initiation Protocol (SIP) Messages
	Author(s)	: S. Olson
	Filename	: draft-ietf-sip-content-indirect-mech-03.txt
	Pages		: 19
	Date		: 2003-6-3
	
This document proposes an extension to the URL MIME External- Body
Access-Type to satisfy the content indirection requirements for SIP.
These extensions are aimed at allowing any MIME part in a SIP message
to be referred to indirectly via a URI.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-content-indirect-mech-03.txt

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Internet-Drafts are also available by anonymous FTP. Login with the username
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	"get draft-ietf-sip-content-indirect-mech-03.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:
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NOTE:	The mail server at ietf.org can return the document in
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--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
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	access-type="mail-server";
	server="mailserv@ietf.org"

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--OtherAccess
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	name="draft-ietf-sip-content-indirect-mech-03.txt";
	site="ftp.ietf.org";
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From: "Rajarshi Chakraborty" <rajarshi_chakraborty@hotmail.com>
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Date: Thu, 05 Jun 2003 04:42:53 +0000
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Subject: [Sip] reference to papers on SIP mobility
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Hi,

Would any sort of discussion on mobility using SIP, in reference to external 
papers, be appreciated in the SIP or SIPPING group? By external, I mean 
papers which are not simply IETF drafts n RFCs, but those maintained in 
sites like Telcordia, universities, etc.

regards
Rajarshi


Rajarshi Chakraborty
B-196, IIT Campus,
Kharagpur,
West Bengal, India
PIN-721302

_________________________________________________________________
Staying fit. It's about being happy! 
http://server1.msn.co.in/features/stayingfit/index.asp Check out the new 
mantra.

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Hi,
I have a question regarding RFC3263.

from the RFC (4):
Because the ACK request for 2xx responses to INVITE constitutes a
different transaction, there is no requirement that it be delivered
to the same server that received the original request (indeed, if
that server did not record-route, it will not get the ACK).

Does that mean that for ACK on 2xx responses, I need to re-query the DNS
server (assuming a non IP uri)?
Do I need to query the DNS even if the URI of the ACK is the same as the URI
of the initial INVITE request?

thanks,
Udi J. Tirosh.

_______________________________________________
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: Thu, 05 Jun 2003 13:15:49 +0530
From: "Nataraju A.B." <natarajuab@huawei.com>
Subject: Re: [Sip] Question on rfc3263 - locating sip servers
To: Udi Weintrob <udiw@radvision.com>, sip@ietf.org
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The basic requirement for ACK for 2xx of INVOTE is to make sure the UAS that 2xx been received by UAC. 

Also ACK may not pass through the same route which the INVITE passed through to UAS.

UAC while sending the ACK request will use the route header in the 2xx(INVITE) response and the Req-URI will be set with teh contact address in teh 2xx response.

more comments inline..

Regards,
-Nataraju A.B.
May you live as long as you wish and love as long as you live. 
--Robert A. Heinlein Time Enough for Love.

----- Original Message ----- 
  From: Udi Weintrob 
  To: sip@ietf.org 
  Sent: Thursday, June 05, 2003 12:31 PM
  Subject: [Sip] Question on rfc3263 - locating sip servers


  Hi,
  I have a question regarding RFC3263.

  from the RFC (4):
  Because the ACK request for 2xx responses to INVITE constitutes a
  different transaction, there is no requirement that it be delivered
  to the same server that received the original request (indeed, if
  that server did not record-route, it will not get the ACK).

  Does that mean that for ACK on 2xx responses, I need to re-query the DNS
  server (assuming a non IP uri)?
  Do I need to query the DNS even if the URI of the ACK is the same as the URI
  of the initial INVITE request?

  [Nataraju A.B.]
  U may need to query the DNS server if u have not cached the domain name and FQDN mapping.

  [Nataraju A.B.]
  In the first place There is no point in sending ACK to some one else also. 

  thanks,
  Udi J. Tirosh.

  _______________________________________________
  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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<DIV><FONT size=2>The basic requirement for ACK for 2xx of INVOTE is to make 
sure the UAS that 2xx been received by UAC. </FONT></DIV>
<DIV><FONT size=2></FONT>&nbsp;</DIV>
<DIV><FONT size=2>Also ACK may not pass through the same route which the 
INVITE&nbsp;passed through to UAS.</FONT></DIV>
<DIV><FONT size=2></FONT>&nbsp;</DIV>
<DIV><FONT size=2>UAC while sending the ACK request will use the route header in 
the 2xx(INVITE) response and the Req-URI will be set with teh contact address in 
teh 2xx response.</FONT></DIV>
<DIV><FONT size=2></FONT>&nbsp;</DIV>
<DIV><FONT size=2>more comments inline..</FONT></DIV>
<DIV><FONT size=2></FONT>&nbsp;</DIV>
<DIV>Regards,<BR>-Nataraju A.B.<BR>May you live as long as you wish and love as 
long as you live. <BR>--Robert A. Heinlein Time Enough for Love.</DIV>
<DIV><FONT size=2></FONT>&nbsp;</DIV>
<DIV>----- Original Message ----- </DIV>
<BLOCKQUOTE 
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
  <DIV 
  style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B> 
  <A title=udiw@radvision.com href="mailto:udiw@radvision.com">Udi Weintrob</A> 
  </DIV>
  <DIV style="FONT: 10pt arial"><B>To:</B> <A title=sip@ietf.org 
  href="mailto:sip@ietf.org">sip@ietf.org</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, June 05, 2003 12:31 
  PM</DIV>
  <DIV style="FONT: 10pt arial"><B>Subject:</B> [Sip] Question on rfc3263 - 
  locating sip servers</DIV>
  <DIV><FONT size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><BR></DIV>
  <DIV>Hi,<BR>I have a question regarding RFC3263.<BR><BR>from the RFC 
  (4):<BR>Because the ACK request for 2xx responses to INVITE constitutes 
  a<BR>different transaction, there is no requirement that it be delivered<BR>to 
  the same server that received the original request (indeed, if<BR>that server 
  did not record-route, it will not get the ACK).<BR><BR>Does that mean that for 
  ACK on 2xx responses, I need to re-query the DNS<BR>server (assuming a non IP 
  uri)?<BR>Do I need to query the DNS even if the URI of the ACK is the same as 
  the URI<BR>of the initial INVITE request?</DIV>
  <DIV><FONT size=2></FONT>&nbsp;</DIV><FONT size=2>
  <DIV><FONT size=2>
  <DIV><FONT size=2>[Nataraju A.B.]</FONT></DIV>U may need to query the DNS 
  server if u have not cached the domain name and FQDN mapping.</FONT></DIV>
  <DIV>&nbsp;</DIV></FONT>
  <DIV><FONT size=2>[Nataraju A.B.]</FONT></DIV>
  <DIV><FONT size=2>In the first place There is no point in sending ACK to some 
  one else also. </FONT></DIV>
  <DIV><FONT size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><FONT 
  size=2></FONT><FONT size=2></FONT><FONT size=2></FONT><BR>thanks,<BR>Udi J. 
  Tirosh.<BR><BR>_______________________________________________<BR>Sip mailing 
  list&nbsp; <A 
  href="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 <A 
  href="mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu</A> 
  for questions on current sip<BR>Use <A 
  href="mailto:sipping@ietf.org">sipping@ietf.org</A> for new developments on 
  the application of sip</DIV></BLOCKQUOTE></BODY></HTML>

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Hi all, I=B4m new to SIP so I was wondering is there some official =
document (draft or RFC) towards the behaviour of a SIP UA after =
receiving a FLASH from the end user??? In other words, how do SIP =
interacts with digitally services such as "transfer call" that needs a =
FLASH+number undestanding???
 Alejandro Islas
 Kb/TEL=20
MEXICO

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<DIV><FONT face=3DArial size=3D2>Hi all, I=B4m new to SIP so I was =
wondering is there=20
some official document (draft or RFC) towards the behaviour of a SIP UA =
after=20
receiving a FLASH from the end user??? In other words, how do SIP =
interacts with=20
digitally services such as "transfer call" that&nbsp;needs a =
FLASH+number=20
undestanding???</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;Alejandro Islas<BR>&nbsp;Kb/TEL =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>MEXICO</FONT></DIV></BODY></HTML>

------=_NextPart_000_0016_01C32906.C461EB40--

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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Alejandro Islas'" <aislas@kbtel.com>, sip@ietf.org
Subject: RE: [Sip] Sip and flash function
Date: Thu, 5 Jun 2003 15:50:35 +0100 
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FLASH + number is stimulus signalling, i.e. the sending terminal does not understand the service that is being requested. This type of signalling is inappropriate to SIP entities (although potentially could in some limited circumstances be passed transparently through the SIP system to some remote entity, e.g. a voice mail server).

You need some entity in the way (before SIP is reached) that interprets that FLASH + number as the call transfer service, or whatever and implements the rest of call transfer in a stimulus fashion back to the non-SIP user, and implements call transfer towards the SIP entities. This is exactly the same function as is performed in the PSTN between the user access and SS7.

As this entity is not in SIP, it is out of scope of the SIP/SIPPING groups.

You will find example descriptions of a number of services provided by SIP in SIPPING group drafts e.g.

http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-transfer-01.txt
http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-04.txt 
http://www.ietf.org/internet-drafts/draft-ietf-sipping-basic-call-flows-02.txt
http://www.ietf.org/internet-drafts/draft-ietf-sipping-pstn-call-flows-02.txt
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-offer-answer-examples-00.txt

Some of these are obviously more relevent to you than others.

regards

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 
-----Original Message-----
From: Alejandro Islas [mailto:aislas@kbtel.com]
Sent: 02 June 2003 18:59
To: sip@ietf.org
Subject: [Sip] Sip and flash function


Hi all, I´m new to SIP so I was wondering is there some official document (draft or RFC) towards the behaviour of a SIP UA after receiving a FLASH from the end user??? In other words, how do SIP interacts with digitally services such as "transfer call" that needs a FLASH+number undestanding???
 Alejandro Islas
 Kb/TEL 
MEXICO
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Use sipping@ietf.org for new developments on the application of sip



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From: "Bang Du" <bangdu@ccpu.com>
To: "Udi Weintrob" <udiw@radvision.com>, <sip@ietf.org>
Subject: RE: [Sip] Question on rfc3263 - locating sip servers
Date: Thu, 5 Jun 2003 12:02:01 -0700
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Hi Udi,
In either case, you need to do the dns query.

Regards,
Bang

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Udi
Weintrob
Sent: Thursday, June 05, 2003 12:02 AM
To: sip@ietf.org
Subject: [Sip] Question on rfc3263 - locating sip servers


Hi,
I have a question regarding RFC3263.

from the RFC (4):
Because the ACK request for 2xx responses to INVITE constitutes a
different transaction, there is no requirement that it be delivered
to the same server that received the original request (indeed, if
that server did not record-route, it will not get the ACK).

Does that mean that for ACK on 2xx responses, I need to re-query the DNS
server (assuming a non IP uri)?
Do I need to query the DNS even if the URI of the ACK is the same as the URI
of the initial INVITE request?

thanks,
Udi J. Tirosh.

_______________________________________________
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 mailnull@www1.ietf.org  Thu Jun  5 16:35:39 2003
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From: Amit Bhadoria <bhadoria@cisco.com>
To: Udi Weintrob <udiw@radvision.com>
cc: sip@ietf.org
Subject: Re: [Sip] Question on rfc3263 - locating sip servers
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udi,

its not a MUST to do a DNS querry for subsequent messages, you could cache
the results from last successfull querry and use it. however, in case of
failover, or to handle some non-trivial case, you may have to do DNS
querry again -- and thats exactly what these guidelines are for, to give a
standard/open procedure.

as a side note, the Record-route inserted by each intermediate hop gives a
little hint on how it wants to be resolved for subsequent messages. for
example, it may choose to insert its domain only, in which case you may
have to do NAPTR/SRV querries.

regards,
-amit.
--
Amit Bhadoria
Software Engineer
http://www.cisco.com

On Thu, 5 Jun 2003, Udi Weintrob wrote:

> Hi,
> I have a question regarding RFC3263.
> 
> from the RFC (4):
> Because the ACK request for 2xx responses to INVITE constitutes a
> different transaction, there is no requirement that it be delivered
> to the same server that received the original request (indeed, if
> that server did not record-route, it will not get the ACK).
> 
> Does that mean that for ACK on 2xx responses, I need to re-query the DNS
> server (assuming a non IP uri)?
> Do I need to query the DNS even if the URI of the ACK is the same as the URI
> of the initial INVITE request?
> 
> thanks,
> Udi J. Tirosh.
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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



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Date: Fri, 6 Jun 2003 17:28:34 -0700
Subject: Re: [Sip] UPDATE and P-Asserted-ID
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Cc: "'Michael Thomas'" <mat@cisco.com>,
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        "Elwell, John" <john.elwell@siemens.com>,
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On Friday, May 23, 2003, at 03:21 AM, Mark Watson wrote:

> Jonathan's approach seems reasonable, and since P-Asserted-Id just 
> represents the identity of the originator of the *message*

i think you are correct, that the spec means the originator of the 
message, but I do not this is especially clear. i was not sure about 
this.  i thought it was just the initiator of the dialog. After reading 
through RFC 3325 three times, the only thing that strongly indicated 
this was the case was the list of which methods could contain the 
header.  It is unfortunate that UPDATE was not included, since this is 
one of the places using P-Asserted-Id makes the most sense.

thanks,
-rohan


> we could expect it to change during a dialog - I suppose there is a 
> philosphical question that if the identity of the SIP UAS changes, 
> does that mean it is now a logically different UAS and so a new dialog 
> should be established ? even if it's the same physical device so that 
> all the state needed to maintain a single dialog is still available ?? 
> I guess this is the case when call forward/transfer happens downstream 
> of a SIP->X gateway.
>
> Equally, were a device to de-register and re-register with a new 
> Address Of Record, does this represent an 'identity change' for that 
> UA ? Could it meaningfully update its identity in any ongoing dialogs 
> > ?
>
> I think there should be scope for individual headers to specify that 
> their state cannot be updated mid-dialog. Not having looked through 
> all the 'such headers' referred to by Jonathan (not even knowing what 
> this list is!) we can't rule out that the meaning of a mid-dialog 
> update for one of these might not be well-defined i.e. it has no 
> sensible meaning or is ambiguous. In those cases we would need to rule 
> out mid-dialog updates, or define the meaning of them for that header.
>
> ...Mark
>
>
>
> > -----Original Message-----
> > From: Michael Thomas [mailto:mat@cisco.com]
> > Sent: 19 May 2003 21:18
> > To: Jonathan Rosenberg
> > Cc: Elwell, John; 'Rohan Mahy'; 'Drage, Keith (Keith)'; 
> 'sip@ietf.org'
> > Subject: Re: [Sip] UPDATE and P-Asserted-ID
> >
> >
> > Jonathan Rosenberg writes:
> >  > 1. Each header field that conveys dialog-associated state would
> >  > indicate whether or not that state is updated by a target
> > refresh request.
> >  >
> >  > 2. We specify a blanket rule for all such headers that
> > their state is
> >  > updated by a target refresh request (absence of a header
> > would imply
> >  > no update).
> >  >
> >  > 3. We specify a blanket rule for all such headers that
> > their state is
> >  > never updated by a target refresh request.
> >  >
> >  > 4. We specify a default rule for all such headers, but allow
> >  > individual headers to override it.
> >  >
> >  >
> >  > In the interests of simplicity I would tend to lean
> > towards (2). The
> >  > best place for this, of course, is rfc 3261, in which case
> > we should
> >  > treat this as a bug against the spec.
> >
> > This seems quite reasonable to me as well --
> > special casing "initial" from subsequent dialog
> > seems pretty fraught. The only reason that I can
> > think of why you might want to do that is when the
> > state assembly/teardown of a mid-session change
> > was so onerous and/or ambiguous to other state
> > that it was prudent to set up the state once and
> > leave it be. It doesn't seem to me that that's the
> > case here, but I'm hardly an expert.
> >
> >        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
> >
>

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This list is for NEW development of the core SIP Protocol
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From: "Christer Holmberg (LMF)" <Christer.Holmberg@lmf.ericsson.se>
To: "'Rohan Mahy'" <rohan@cisco.com>,
        Mark Watson
	 <mwatson@nortelnetworks.com>
Cc: "'Michael Thomas'" <mat@cisco.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>,
        "Elwell, John" <john.elwell@siemens.com>,
        "'Drage, Keith (Keith)'" <drage@lucent.com>,
        "'sip@ietf.org'"
	 <sip@ietf.org>
Subject: RE: [Sip] UPDATE and P-Asserted-ID
Date: Sat, 7 Jun 2003 09:47:23 +0200 
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Hi,

There currently are a number of headers where the sender or a request, and the initiator of a dialog, can insert some kind of identity of themselves. They are:

Contact
From
In-Reply-To
Reply-To
P-Asserted-Identity

etc

In most cases the spec(s) do define what they can be used for, but not really what they identify (user, device, etc). Maybe some clarification is needed for that?

I also agree that we need to define which headers are allowed to be changed/updated during a session, and what it really means from a dialog point of view.

Regards,

Christer Holmberg
Ericsson Finland


> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: 7. kesäkuuta 2003 3:29
> To: Mark Watson
> Cc: 'Michael Thomas'; Jonathan Rosenberg; Elwell, John; 'Drage, Keith
> (Keith)'; 'sip@ietf.org'
> Subject: Re: [Sip] UPDATE and P-Asserted-ID
> 
> 
> 
> On Friday, May 23, 2003, at 03:21 AM, Mark Watson wrote:
> 
> > Jonathan's approach seems reasonable, and since P-Asserted-Id just 
> > represents the identity of the originator of the *message*
> 
> i think you are correct, that the spec means the originator of the 
> message, but I do not this is especially clear. i was not sure about 
> this.  i thought it was just the initiator of the dialog. 
> After reading 
> through RFC 3325 three times, the only thing that strongly indicated 
> this was the case was the list of which methods could contain the 
> header.  It is unfortunate that UPDATE was not included, 
> since this is 
> one of the places using P-Asserted-Id makes the most sense.
> 
> thanks,
> -rohan
> 
> 
> > we could expect it to change during a dialog - I suppose there is a 
> > philosphical question that if the identity of the SIP UAS changes, 
> > does that mean it is now a logically different UAS and so a 
> new dialog 
> > should be established ? even if it's the same physical 
> device so that 
> > all the state needed to maintain a single dialog is still 
> available ?? 
> > I guess this is the case when call forward/transfer happens 
> downstream 
> > of a SIP->X gateway.
> >
> > Equally, were a device to de-register and re-register with a new 
> > Address Of Record, does this represent an 'identity change' 
> for that 
> > UA ? Could it meaningfully update its identity in any 
> ongoing dialogs 
> > > ?
> >
> > I think there should be scope for individual headers to 
> specify that 
> > their state cannot be updated mid-dialog. Not having looked through 
> > all the 'such headers' referred to by Jonathan (not even 
> knowing what 
> > this list is!) we can't rule out that the meaning of a mid-dialog 
> > update for one of these might not be well-defined i.e. it has no 
> > sensible meaning or is ambiguous. In those cases we would 
> need to rule 
> > out mid-dialog updates, or define the meaning of them for 
> that header.
> >
> > ...Mark
> >
> >
> >
> > > -----Original Message-----
> > > From: Michael Thomas [mailto:mat@cisco.com]
> > > Sent: 19 May 2003 21:18
> > > To: Jonathan Rosenberg
> > > Cc: Elwell, John; 'Rohan Mahy'; 'Drage, Keith (Keith)'; 
> > 'sip@ietf.org'
> > > Subject: Re: [Sip] UPDATE and P-Asserted-ID
> > >
> > >
> > > Jonathan Rosenberg writes:
> > >  > 1. Each header field that conveys dialog-associated state would
> > >  > indicate whether or not that state is updated by a target
> > > refresh request.
> > >  >
> > >  > 2. We specify a blanket rule for all such headers that
> > > their state is
> > >  > updated by a target refresh request (absence of a header
> > > would imply
> > >  > no update).
> > >  >
> > >  > 3. We specify a blanket rule for all such headers that
> > > their state is
> > >  > never updated by a target refresh request.
> > >  >
> > >  > 4. We specify a default rule for all such headers, but allow
> > >  > individual headers to override it.
> > >  >
> > >  >
> > >  > In the interests of simplicity I would tend to lean
> > > towards (2). The
> > >  > best place for this, of course, is rfc 3261, in which case
> > > we should
> > >  > treat this as a bug against the spec.
> > >
> > > This seems quite reasonable to me as well --
> > > special casing "initial" from subsequent dialog
> > > seems pretty fraught. The only reason that I can
> > > think of why you might want to do that is when the
> > > state assembly/teardown of a mid-session change
> > > was so onerous and/or ambiguous to other state
> > > that it was prudent to set up the state once and
> > > leave it be. It doesn't seem to me that that's the
> > > case here, but I'm hardly an expert.
> > >
> > >        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
> > >
> >
> 
> _______________________________________________
> 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 mailnull@www1.ietf.org  Sat Jun  7 23:06:16 2003
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From: "Venkatesh Venkataramanan" <Venkatesh.Venkataramanan@sylantro.com>
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Subject: [Sip] P-Asserted-Identity and Called Party Name Delivery
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Hi:

There has been some discussion in the mailing list about using the
P-Asserted-Identity header in requests like UPDATE to enable updating the
identity of the caller mid-call. I was thinking anyone has considered
enabling this header in SIP response messages as well? There are a few
features that would need updating the calling party, the identity of the
target while the called target is still being alerted. While 3261 allows the
UAS to use the Contact header, there are a couple scenarios that need
something more than the contact, a header a proxy in the middle can update
in responses. 

1) In the case of calls to numbers like hunt-groups and/or ACD, every UAS
that receives the INVITE will place its local contact information in the
response to this request. All that may be expected for these calls is that
the calling party see information they called a name corresponding to the
hunt group or ACD number they called; like "United Airlines" and not "Agent
John Doe". Arguably this may be achieved by local configuration of some sort
at the UAS that is a part of the ACD group; but there can be call flows
where the UAS receiving an INVITE might not even be aware that it is being
reached as a part of some application. For example an implementation might
choose to store hunt group information in a proxy and have the proxy fork
sequentially or round-robin or whatever, and not having the UAS know about
it. Another example might be a find me-follow me type service where the
calling party may not need to know the list of targets that is being looked
up or which of these targets answered the call.

2) A proxy providing called name lookup services for a user community
whereby it looks up a centralized directory database of some sort and update
the calling UA of the called party's name; this is useful if the called
number is a PSTN number. 

3) An UAS might not even place it's display name in a SIP response. Enabling
a proxy to add this information gurantees a end user of this feature
regardless of the capabilities and/or preferences of the UAS.

4) Display language preferences: A end user might have preferences to
display called names in English. The UAS called might have preferences that
makes it send the called display name in different language. A proxy might
want to provide a directory look up service based on the called number and
provide the information to the end user in a language preferred by the
calling UA. 

Is it worth considering enabling proxies to insert the P-Asserted-Identity
header in SIP response message for the same?? 

regards,
Venkatesh

_______________________________________________
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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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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] UPDATE and P-Asserted-ID
To: "'Rohan Mahy'" <rohan@cisco.com>, Mark Watson <mwatson@nortelnetworks.com>
Cc: "'Michael Thomas'" <mat@cisco.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        "'Drage, Keith (Keith)'" <drage@lucent.com>,
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Section 5 of RFC3325 states:

   "If the proxy receives a message (request or response) from a node
   that it trusts, it can use the information in the P-Asserted-Identity
   header field, if any, as if it had authenticated the user itself.

   If there is no P-Asserted-Identity header field present, a proxy MAY
   add one containing...."

I take this as meaning that P-Asserted-Id can be sent in requests or
responses. This capability is made use of in draft-ietf-sipping-qsig2sip-02
sections 9.1.3 and 9.2.3.

John Elwell (john.elwell@siemens.com)

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: 07 June 2003 01:29
> To: Mark Watson
> Cc: 'Michael Thomas'; Jonathan Rosenberg; Elwell, John; 'Drage, Keith
> (Keith)'; 'sip@ietf.org'
> Subject: Re: [Sip] UPDATE and P-Asserted-ID
> 
> 
> 
> On Friday, May 23, 2003, at 03:21 AM, Mark Watson wrote:
> 
> > Jonathan's approach seems reasonable, and since P-Asserted-Id just 
> > represents the identity of the originator of the *message*
> 
> i think you are correct, that the spec means the originator of the 
> message, but I do not this is especially clear. i was not sure about 
> this.  i thought it was just the initiator of the dialog. 
> After reading 
> through RFC 3325 three times, the only thing that strongly indicated 
> this was the case was the list of which methods could contain the 
> header.  It is unfortunate that UPDATE was not included, 
> since this is 
> one of the places using P-Asserted-Id makes the most sense.
> 
> thanks,
> -rohan
> 
> 
> > we could expect it to change during a dialog - I suppose there is a 
> > philosphical question that if the identity of the SIP UAS changes, 
> > does that mean it is now a logically different UAS and so a 
> new dialog 
> > should be established ? even if it's the same physical 
> device so that 
> > all the state needed to maintain a single dialog is still 
> available ?? 
> > I guess this is the case when call forward/transfer happens 
> downstream 
> > of a SIP->X gateway.
> >
> > Equally, were a device to de-register and re-register with a new 
> > Address Of Record, does this represent an 'identity change' 
> for that 
> > UA ? Could it meaningfully update its identity in any 
> ongoing dialogs 
> > > ?
> >
> > I think there should be scope for individual headers to 
> specify that 
> > their state cannot be updated mid-dialog. Not having looked through 
> > all the 'such headers' referred to by Jonathan (not even 
> knowing what 
> > this list is!) we can't rule out that the meaning of a mid-dialog 
> > update for one of these might not be well-defined i.e. it has no 
> > sensible meaning or is ambiguous. In those cases we would 
> need to rule 
> > out mid-dialog updates, or define the meaning of them for 
> that header.
> >
> > ...Mark
> >
> >
> >
> > > -----Original Message-----
> > > From: Michael Thomas [mailto:mat@cisco.com]
> > > Sent: 19 May 2003 21:18
> > > To: Jonathan Rosenberg
> > > Cc: Elwell, John; 'Rohan Mahy'; 'Drage, Keith (Keith)'; 
> > 'sip@ietf.org'
> > > Subject: Re: [Sip] UPDATE and P-Asserted-ID
> > >
> > >
> > > Jonathan Rosenberg writes:
> > >  > 1. Each header field that conveys dialog-associated state would
> > >  > indicate whether or not that state is updated by a target
> > > refresh request.
> > >  >
> > >  > 2. We specify a blanket rule for all such headers that
> > > their state is
> > >  > updated by a target refresh request (absence of a header
> > > would imply
> > >  > no update).
> > >  >
> > >  > 3. We specify a blanket rule for all such headers that
> > > their state is
> > >  > never updated by a target refresh request.
> > >  >
> > >  > 4. We specify a default rule for all such headers, but allow
> > >  > individual headers to override it.
> > >  >
> > >  >
> > >  > In the interests of simplicity I would tend to lean
> > > towards (2). The
> > >  > best place for this, of course, is rfc 3261, in which case
> > > we should
> > >  > treat this as a bug against the spec.
> > >
> > > This seems quite reasonable to me as well --
> > > special casing "initial" from subsequent dialog
> > > seems pretty fraught. The only reason that I can
> > > think of why you might want to do that is when the
> > > state assembly/teardown of a mid-session change
> > > was so onerous and/or ambiguous to other state
> > > that it was prudent to set up the state once and
> > > leave it be. It doesn't seem to me that that's the
> > > case here, but I'm hardly an expert.
> > >
> > >        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
> > >
> >
> 
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Hi all, I=B4m new to SIP. Where can i found some official documents (draft=
 or=20
RFC) for use Java (and applets) SIP methods?

thanks.

Andrea (andrea@giandrea.com)=20
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<html>
<body>
<br>
<font face=3D"arial" size=3D2>Hi all, I=B4m new to SIP. Where can i found so=
me
official documents (draft or RFC) for use Java (and applets) SIP
methods?<br><br>
</font>thanks. <br><br>
Andrea (andrea@giandrea.com)</body>
</html>

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Date: Mon, 09 Jun 2003 19:59:58 +0530
From: "Nataraju A.B." <natarajuab@huawei.com>
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Regards,
-Nataraju A.B.
May you live as long as you wish and love as long as you live. 
--Robert A. Heinlein Time Enough for Love.
----- Original Message ----- 
From: Nataraju A.B. 
To: SIP Implementors 
Sent: Monday, June 09, 2003 5:04 PM
Subject: Is there a freeeware available for conversion between feature predicate and feature parameters...........

 
Hi All, 

is there a freeware or general guidelines available for converion between feature predicate and feature parameters. 

Thanx in advance, 

Regards,
-Nataraju A.B.

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<DIV><FONT size=2></FONT>&nbsp;</DIV>
<DIV>Regards,<BR>-Nataraju A.B.<BR>May you live as long as you wish and love as 
long as you live. <BR>--Robert A. Heinlein Time Enough for Love.</DIV>
<DIV style="FONT: 10pt arial">----- Original Message ----- 
<DIV style="BACKGROUND: #e4e4e4; font-color: black"><B>From:</B> <A 
title=natarajuab@huawei.com href="mailto:natarajuab@huawei.com">Nataraju 
A.B.</A> </DIV>
<DIV><B>To:</B> <A title=sip-implementors@cs.columbia.edu 
href="mailto:sip-implementors@cs.columbia.edu">SIP Implementors</A> </DIV>
<DIV><B>Sent:</B> Monday, June 09, 2003 5:04 PM</DIV>
<DIV><B>Subject:</B> Is there a freeeware available for conversion between 
feature predicate and feature parameters...........</DIV></DIV>
<DIV><BR>&nbsp;</DIV>
<DIV><FONT size=2>Hi All, </FONT></DIV>
<DIV><FONT size=2></FONT>&nbsp;</DIV>
<DIV><FONT size=2>is there a freeware or general guidelines available for 
converion between feature predicate and feature parameters. </FONT></DIV>
<DIV><FONT size=2></FONT>&nbsp;</DIV>
<DIV><FONT size=2>Thanx in advance, </FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face="Courier New" color=#008080 size=2>Regards,<BR>-Nataraju 
A.B.</FONT></DIV></BODY></HTML>

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giAndrea wrote:
> 
> Hi all, I´m new to SIP. Where can i found some official documents (draft 
> or RFC) for use Java (and applets) SIP methods?

http://java.sun.com/products/jain/api_specs.html

> 
> thanks.
> 
> Andrea (andrea@giandrea.com)



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From mailnull@www1.ietf.org  Mon Jun  9 15:44:33 2003
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Subject: Re: [Sip] WGLC for auth-id body
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Rohan Mahy <rohan@cisco.com>
Cc: sip@ietf.org, "'Dean Willis'" <dean.willis@softarmor.com>,
        Gonzalo.Camarillo@ericsson.com,
        Jon Peterson <jon.peterson@neustar.biz>
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I've reviewed this document (with revision of referredby
particularly in mind) and have the following minor comments:

- It would help section 4 to more explicitly note that it
  is discussing using AIBs to do something beyond(besides?)
  providing integrity protection/authentication for the 
  request it appears in.  It would also help to capture that
  the presence of these other things don't preclude the section
  2 use of an AIB. Perhaps this text at end of the first paragraph
  of section 4:

     Such information might be carried in one or more supplemental
     AIBs. The presence of these supplemental AIBs does not preclude
     the use of AIB as specified in this document to protect the
     message in which they appear.

- Instead of a special case of INVITE (see the heading of section 4), I
  think this simply a different use of an AIB. This document constrains
  the use of AIBs for providing integrity protection of messages and
  authenticating their sender regardless of method type. Section 4 is
  trying to note that other AIBs might appear that do something 
  different. Perhaps this section could be titled "Potential additional
  uses of AIBs"?

- Section 10 (Security Considerations): The first sentence needs to
  be scoped to the section 2 use of AIBs, not all uses of AIBs with
  message/sipfrag in them.

- Spelling Nits:
   * Second sentence, last paragraph Page 4: 
       s/SHOULD be added it to/SHOULD be added to/
   * Second sentence, last paragraph, section 4, Page 6:
       s/harder to correlated an AIB/harder to correlate an AIB/

Other than that, I believe this document is ready to go.

RjS

On Tue, 2003-05-13 at 19:39, Rohan Mahy wrote:
> Hello Everyone,
> 
> I would like to begin Working Group Last Call on
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sip-authid-body-01.txt
> 
> WGLC will end on Friday, June 13, 2003.
> 
> thanks,
> -rohan
> 
> _______________________________________________
> 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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Several weeks ago, Pekka made a suggestion that I would like to
follow up on now (sorry for the very long delay Pekka). His 
suggestion was to use a different Content-Disposition
disposition-type for referredby-tokens.

This draws attention to what the "aib" value means. 

Is it merely a synonym for "signed message/sipfrag"? If so, the
authid-body draft defines the value _and_ specifies one particular
use of a body with that disposition type (providing integrity and
authentication of sender for a single message). Other uses are valid,
so we should reuse it and not add noise to the IANA registry.

If, instead, "aib" means "signed message/sipfrag with these
particular security implications", then our referredby token
is _not_ an aib. It's something _like_ an aib, but with 
slightly different requirements, and artifacts in the protocol
(like the disposition-type) should probably reflect that. If this
is the case, referredby could register something like "RBIB"
(ReferredBy Identity Body), or, treading much more dangerous ground,
"TPIB" (Third Party Identity Body).

So, which of the above meanings for AIB is auth-id body establishing?

RjS


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Subject: [Sip] Question on draft-ietf-sip-identity-01
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Hi Jon,

regarding draft-ietf-sip-identity-01, I'm wondering what can be the 
problems trying to place the authentication token into a SIP header 
rather than the body: has the consequences of this option been 
investigated already?

Thanks in advance
Bye
Marco Aime

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Dipartimento di Automatica e Informatica
Politecnico di Torino
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Tel: +39 011 22102-44
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From: Vijay Kamath <vijay.kamath@wipro.com>
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Hi all,

As per "draft-ietf-sip-replaces-03.txt", the Replaces header can only be 
present in an INVITE request. I was wondering if this invite request 
MUST always be a new request or can it also be a re-invite. Re-invites 
are mentioned nowhere in the draft.

Regards... Vijay

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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To: "'Vijay Kamath'" <vijay.kamath@wipro.com>, <sip@ietf.org>
Subject: RE: [Sip] Replaces header query
Date: Wed, 11 Jun 2003 23:20:49 -0700
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A Replaces: header in a re-INVITE would be a bit odd, but I don't see any
reason to disallow it.

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Vijay
Kamath
Sent: Wednesday, June 11, 2003 11:00 PM
To: sip@ietf.org
Subject: [Sip] Replaces header query


Hi all,

As per "draft-ietf-sip-replaces-03.txt", the Replaces header can only be 
present in an INVITE request. I was wondering if this invite request 
MUST always be a new request or can it also be a re-invite. Re-invites 
are mentioned nowhere in the draft.

Regards... Vijay

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

_______________________________________________
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From mailnull@www1.ietf.org  Thu Jun 12 06:48:59 2003
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From: "Shailendra" <shail@mahindrabt.com>
To: "Sip@Ietf. Org" <sip@ietf.org>
Cc: "Tasos Dagiouklas" <ntan@intracom.gr>
References: <NGBBLHKPJKCDNEIOOKKOKELPCLAA.ssal@intracom.gr>
Subject: Re: [Sip] Of-hook and On-hook @ sip?
Date: Thu, 12 Jun 2003 16:16:20 +0530
Organization: Mahibdra - British Telecom
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Hi,

INVITE will map to SETUP ( Q-931).
but if you ask precisely about PSTN then
INVITE is equivalent of (OFFHOOK + Sizure + dialed digits )

BYE will map to ONHOOK in PSTN and REL in Q-931

/Shailendra

----- Original Message -----
From: "F.S.Salloum" <ssal@intracom.gr>
To: "Sip@Ietf. Org" <sip@ietf.org>
Cc: "Tasos Dagiouklas" <ntan@intracom.gr>
Sent: Wednesday, February 26, 2003 6:01 PM
Subject: [Sip] Of-hook and On-hook @ sip?

> Dear all,
>
> One simple question is there a similar message
> in SIP which works like on-hook and off-hook on
> the PSTN world (Q-931)?
>
> I know that this can be implemented playing with
> the state machines of the hard-sip-phone but we were
> wondering if this has been investigated from you guys.
>
> Thanks
> F.S.Salloum
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

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Hi,
    Anyone has any idea regarding what should be message body of SIP =
Info method that carries the DTMF digits. Is there any standard RFC or =
draft implementation.
    Any suggestions are welcome.

    Regards
    Debanjan
   =20
    D-Link India Ltd
    Software and R & D Center

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<META content=3D"text/html; charset=3Diso-8859-1" =
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<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; Anyone has any idea =
regarding=20
what should be message body of SIP Info method that carries the DTMF =
digits. Is=20
there any standard RFC or draft implementation.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; Any suggestions are=20
welcome.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; Regards</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; =
Debanjan</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; D-Link India =
Ltd</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; Software and R &amp; =
D=20
Center</FONT></DIV></BODY></HTML>

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Subject: [Sip] Question on draft-schulzrinne-sip-register-01
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Hi,

The draft in reference to which I have a doubt has already expired long back 
(October 2001). So are questions in reference to outdated drafts entertained 
in these mailing lists? It would help me a lot if someone responds to this 
email.
My doubt is regarding the second model of registration called "Outbound 
proxy intercept"...

It has been proposed in this model that the foreign outbound proxy must 
maintain a temporary ID for the visiting mobile host. But who is supposed to 
store this ID?

Also I understand that the Local proxy will only forward the register msg to 
the Local registrar only after it receives a 200 OK from the home proxy, 
either with local proxy's contact or with the real local contact. But what 
about the SIP URI in the To and From headers of the regiater msg that the 
local proxy will forward to the local registrar?

If A is the home proxy's domain and B is the local proxy's domain and 1000 
is the user name, would the SIP URI in the To and From header of this 
register msg be
1000%40A@B ? Ofcourse the SIP URI in the register msg that is forwarded to 
the home proxy should be 1000@A. I guess the canonical visitor name is 
useful for the local proxy to handle incoming requests meant for the 
visiting mobile host. But for the Register I think the ID corresponding to 
the visiting host should be 1000%40A, i.e. the first part. This has not been 
mentioned in the draft. Or is it
supposed to be understood from the way the model has been proposed in the 
draft?

regards,
Rajarshi

Rajarshi Chakraborty
B-196, IIT Campus,
Kharagpur,
West Bengal, India
PIN-721302

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From: "Umesh Sharma" <umsharma@techie.com>
To: "debanjan" <debanjan@dlink.co.in>, sip@ietf.org, sipping@ietf.org
Date: Fri, 13 Jun 2003 01:01:18 -0500
Subject: Re: [Sip] Message Body format for INFO method carrying DTMF Digits
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<P>There is this&nbsp;draft "draft-choudhuri-sip-info-digit-00.txt"&nbsp; but it seems it has expired &amp; other one is "RFC 2833"&nbsp; on DTMF transmission. </P>
<P>Umesh<BR><BR>----- Original Message -----<BR>From: "debanjan" <DEBANJAN@DLINK.CO.IN><BR>Date: Thu, 12 Jun 2003 19:05:25 +0530<BR>To: <SIP@IETF.ORG>, <SIPPING@IETF.ORG><BR>Subject: [Sip] Message Body format for INFO method carrying DTMF Digits<BR><BR><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<DIV><FONT face=Arial size=2>Hi,</FONT></DIV>
<DIV><FONT face=Arial size=2>&nbsp;&nbsp;&nbsp; Anyone has any idea regarding what should be message body of SIP Info method that carries the DTMF digits. Is there any standard RFC or draft implementation.</FONT></DIV>
<DIV><FONT face=Arial size=2>&nbsp;&nbsp;&nbsp; Any suggestions are welcome.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>&nbsp;&nbsp;&nbsp; Regards</FONT></DIV>
<DIV><FONT face=Arial size=2>&nbsp;&nbsp;&nbsp; Debanjan</FONT></DIV>
<DIV><FONT face=Arial size=2>&nbsp;&nbsp;&nbsp; </FONT></DIV>
<DIV><FONT face=Arial size=2>&nbsp;&nbsp;&nbsp; D-Link India Ltd</FONT></DIV>
<DIV><FONT face=Arial size=2>&nbsp;&nbsp;&nbsp; Software and R &amp; D Center</FONT></DIV>
-- 
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From mailnull@www1.ietf.org  Fri Jun 13 09:51:42 2003
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Robert Sparks'" <rsparks@dynamicsoft.com>, sip@ietf.org
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Subject: [Sip] RE: Use of AIB in referredby
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I look at the AIB as the application of sipfrags to the problem described in
RFC3261 23.4 - the tunneling of integrity and authentication properties
within the MIME bodies of SIP messages. In the AIB, these properties are
intended to provide identity, hence the name 'authenticated identity body'.
I think that the usage in referredby is sufficiently close to that purpose,
providing authenticated identity within a body, that I don't think we have a
problem with reuse of the term.

The real question, I think, is whether or not there is any value in
differentiating an AIB resulting from a REFER from any other AIB (such as a
'normal' one representing the sender of an INVITE) that might be in a
request. Surely these need to be differentiated somehow - the signature on
the AIB itself would be one indicator, as would the headers in the body, but
Content-Disposition is another point at which the two could be
distinguished. 

I don't have a strong intuition about whether or not the use of a different
Content-Disposition would be valuable.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
> Sent: Monday, June 09, 2003 1:00 PM
> To: sip@ietf.org
> Cc: Jon Peterson
> Subject: Use of AIB in referredby
> 
> 
> Several weeks ago, Pekka made a suggestion that I would like to
> follow up on now (sorry for the very long delay Pekka). His 
> suggestion was to use a different Content-Disposition
> disposition-type for referredby-tokens.
> 
> This draws attention to what the "aib" value means. 
> 
> Is it merely a synonym for "signed message/sipfrag"? If so, the
> authid-body draft defines the value _and_ specifies one particular
> use of a body with that disposition type (providing integrity and
> authentication of sender for a single message). Other uses are valid,
> so we should reuse it and not add noise to the IANA registry.
> 
> If, instead, "aib" means "signed message/sipfrag with these
> particular security implications", then our referredby token
> is _not_ an aib. It's something _like_ an aib, but with 
> slightly different requirements, and artifacts in the protocol
> (like the disposition-type) should probably reflect that. If this
> is the case, referredby could register something like "RBIB"
> (ReferredBy Identity Body), or, treading much more dangerous ground,
> "TPIB" (Third Party Identity Body).
> 
> So, which of the above meanings for AIB is auth-id body establishing?
> 
> RjS
> 
> 
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From mailnull@www1.ietf.org  Fri Jun 13 10:51:46 2003
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Marco Aime'" <m.aime@polito.it>
Cc: sip@ietf.org
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Subject: [Sip] RE: Question on draft-ietf-sip-identity-01
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Hi,

Yes, the consequences of placing an authentication token into a SIP header
have been considered in the past. There were really three significant
factors that led to the architecture used in draft-ietf-sip-identity (in
which the authentication token appears in a body instead):

- The tokens themselves can be quite large and complicated. Following the
recommendation in the AIB draft (draft-ietf-sip-authid-body), a token might
contain numerous headers which are required for reference integrity.
Compound that with a digital signature. Compound that with an appended
certificate for verifying the signature (common in CMS applications). All
together, the size of one of these tokens would be considerably larger that
conventional SIP headers. While technically, headers are unbounded in size,
from a practical perspective chunks of data over a certain threshhold are
more suitable for the body of a message than a header. A proxy handling a
header that was, say, over 1K in size could conceivably hiccup. Encoding a
header that contained multiple SIP headers and a digital signature and a
certificate could also be challenging.

- For additional reference integrity, some tokens may want to place
signatures around actual message bodies, notably SDP. This is a general
motivation for the use of S/MIME for identity. Replicating bodies in headers
would be... silly.

- Headers are frequently manipulated by proxy servers - bodies, however,
MUST NOT be modified by proxy servers (per RFC3261 16.6). A great deal of
emphasis in the SIP security work was placed on end-to-end security
properties. While headers can be marked unmodifiable (by omitting the 'amd's
in the famous Table 2 of RFC3261), the guidelines for bodies are more
strict.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Marco Aime [mailto:m.aime@polito.it]
> Sent: Tuesday, June 10, 2003 9:21 AM
> To: jon.peterson@neustar.biz
> Cc: sip@ietf.org
> Subject: Question on draft-ietf-sip-identity-01
> 
> 
> Hi Jon,
> 
> regarding draft-ietf-sip-identity-01, I'm wondering what can be the 
> problems trying to place the authentication token into a SIP header 
> rather than the body: has the consequences of this option been 
> investigated already?
> 
> Thanks in advance
> Bye
> Marco Aime
> 
> -- 
> ------------------------------------------------------------------
> Marco AIME
> Dipartimento di Automatica e Informatica
> Politecnico di Torino
> Addr: Via Cardinal Massaia 83, Torino, Italy
> Tel: +39 011 22102-44
> Fax: +39 011 22102-29
> Mail: m.aime@polito.it (marcodomenico.aime@polito.it)
> ------------------------------------------------------------------
> 
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Robert Sparks'" <rsparks@dynamicsoft.com>,
        Rohan Mahy
	 <rohan@cisco.com>
Cc: sip@ietf.org, "'Dean Willis'" <dean.willis@softarmor.com>,
        Gonzalo.Camarillo@ericsson.com
Subject: RE: [Sip] WGLC for auth-id body
Date: Fri, 13 Jun 2003 03:03:16 -0400
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Thanks for these comments. A few notes below.

Jon Peterson
NeuStar, Inc.

[snip]
> I've reviewed this document (with revision of referredby
> particularly in mind) and have the following minor comments:
> 
> - It would help section 4 to more explicitly note that it
>   is discussing using AIBs to do something beyond(besides?)
>   providing integrity protection/authentication for the 
>   request it appears in.  It would also help to capture that
>   the presence of these other things don't preclude the section
>   2 use of an AIB. Perhaps this text at end of the first paragraph
>   of section 4:
> 
>      Such information might be carried in one or more supplemental
>      AIBs. The presence of these supplemental AIBs does not preclude
>      the use of AIB as specified in this document to protect the
>      message in which they appear.
> 

Yes, I do think it is important to capture the idea that there may be
multiple AIBs in a SIP message that identify different parties. The text
given above looks good to me.

> - Instead of a special case of INVITE (see the heading of section 4), I
>   think this simply a different use of an AIB. This document constrains
>   the use of AIBs for providing integrity protection of messages and
>   authenticating their sender regardless of method type. Section 4 is
>   trying to note that other AIBs might appear that do something 
>   different. Perhaps this section could be titled "Potential additional
>   uses of AIBs"?
> 

Well, although REFER-instigated INVITEs might not be a special case as such,
most would probably say that 3PCC is a special case. But still, we need a
section title that accommodates both, so I think that's fair; a better title
for this section is probably in order. I think we need something with the
sense "Use of AIBs to express the identity of someone other than the sender
of the INVITE". I'll try to figure out a way to compress that to
header-size.

> - Section 10 (Security Considerations): The first sentence needs to
>   be scoped to the section 2 use of AIBs, not all uses of AIBs with
>   message/sipfrag in them.
> 

Okay, fine.

> - Spelling Nits:
>    * Second sentence, last paragraph Page 4: 
>        s/SHOULD be added it to/SHOULD be added to/
>    * Second sentence, last paragraph, section 4, Page 6:
>        s/harder to correlated an AIB/harder to correlate an AIB/
> 

Thanks, I'll take care of all of those.

> Other than that, I believe this document is ready to go.
> 
> RjS
> 
> On Tue, 2003-05-13 at 19:39, Rohan Mahy wrote:
> > Hello Everyone,
> > 
> > I would like to begin Working Group Last Call on
> > 
> > 
> http://www.ietf.org/internet-drafts/draft-ietf-sip-authid-body-01.txt
> > 
> > WGLC will end on Friday, June 13, 2003.
> > 
> > thanks,
> > -rohan
> > 
> > _______________________________________________
> > 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 mailnull@www1.ietf.org  Fri Jun 13 12:56:42 2003
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From: "Rajarshi Chakraborty" <rajarshi_chakraborty@hotmail.com>
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Date: Fri, 13 Jun 2003 14:27:02 +0000
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Subject: [Sip] Using SUBSCRIBE/NOTIFY in mobility
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Hi,

This mail is kind of a follow-up to the earlier mail where I posted a couple 
of doubts regarding registration models in draft-schulzrinne-register-01. In 
the second design called "Outbound proxy intercept", could we not use 
SUBSCRIBE and NOTIFY between the mobile user agent and the outbound proxy 
server in the foriegn domain to indicate the former's departure from that 
domain into some other domain? And this would be followed by the local 
registrar unbinding the temporary canonical ID of the mobile host just upon 
receiving the NOTIFY, if it is co-located with the local proxy or the latter 
could send a de-register to the registrar if not co-located with it. 
According to this design the mobile host doesn't register with the foreign 
proxy/registrar. Hence I think SUBSCRIBE/NOTIFY would be of good use. The 
local proxy would SUBSCRIBE to the mobile user agent, and just before the 
latter departs it sends a NOTIFY back to the local proxy. Is this feasible?

regards,
Rajarshi



Rajarshi Chakraborty
B-196, IIT Campus,
Kharagpur,
West Bengal, India
PIN-721302

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From mailnull@www1.ietf.org  Fri Jun 13 13:07:27 2003
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From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'Robert Sparks'"
	 <rsparks@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] RE: Use of AIB in referredby
Date: Fri, 13 Jun 2003 10:58:39 -0500
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My general preference is to be more explicit and towards that end I would
think a different Content-Disposition disposition-type could be useful.
That said, if AIB were as general as Robert suggested initially, then it
could possibly be reused even for the Inserted headers (like History-Info),
where I'm currently modeling the solution after AIB, but was planning on a
new Content-Disposition disposition-type.  However, even I would agree that
the Inserted Headers are more different than the original intent of AIB than
the Referred-by use and again I do prefer to be more explicit.

Mary.

-----Original Message-----
From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
Sent: Friday, June 13, 2003 2:17 AM
To: 'Robert Sparks'; sip@ietf.org
Subject: [Sip] RE: Use of AIB in referredby



I look at the AIB as the application of sipfrags to the problem described in
RFC3261 23.4 - the tunneling of integrity and authentication properties
within the MIME bodies of SIP messages. In the AIB, these properties are
intended to provide identity, hence the name 'authenticated identity body'.
I think that the usage in referredby is sufficiently close to that purpose,
providing authenticated identity within a body, that I don't think we have a
problem with reuse of the term.

The real question, I think, is whether or not there is any value in
differentiating an AIB resulting from a REFER from any other AIB (such as a
'normal' one representing the sender of an INVITE) that might be in a
request. Surely these need to be differentiated somehow - the signature on
the AIB itself would be one indicator, as would the headers in the body, but
Content-Disposition is another point at which the two could be
distinguished. 

I don't have a strong intuition about whether or not the use of a different
Content-Disposition would be valuable.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
> Sent: Monday, June 09, 2003 1:00 PM
> To: sip@ietf.org
> Cc: Jon Peterson
> Subject: Use of AIB in referredby
> 
> 
> Several weeks ago, Pekka made a suggestion that I would like to
> follow up on now (sorry for the very long delay Pekka). His 
> suggestion was to use a different Content-Disposition
> disposition-type for referredby-tokens.
> 
> This draws attention to what the "aib" value means. 
> 
> Is it merely a synonym for "signed message/sipfrag"? If so, the
> authid-body draft defines the value _and_ specifies one particular
> use of a body with that disposition type (providing integrity and
> authentication of sender for a single message). Other uses are valid,
> so we should reuse it and not add noise to the IANA registry.
> 
> If, instead, "aib" means "signed message/sipfrag with these
> particular security implications", then our referredby token
> is _not_ an aib. It's something _like_ an aib, but with 
> slightly different requirements, and artifacts in the protocol
> (like the disposition-type) should probably reflect that. If this
> is the case, referredby could register something like "RBIB"
> (ReferredBy Identity Body), or, treading much more dangerous ground,
> "TPIB" (Third Party Identity Body).
> 
> So, which of the above meanings for AIB is auth-id body establishing?
> 
> RjS
> 
> 
_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Fri Jun 13 13:17:18 2003
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From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'Robert Sparks'"
	 <rsparks@dynamicsoft.com>,
        Rohan Mahy <rohan@cisco.com>
Cc: sip@ietf.org, "'Dean Willis'" <dean.willis@softarmor.com>,
        Gonzalo.Camarillo@ericsson.com
Subject: RE: [Sip] WGLC for auth-id body
Date: Fri, 13 Jun 2003 11:09:38 -0500
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There's one more nit that I noticed that doesn't appear to be mentioned by
Robert. 

In the second paragraph of section 4, the 2nd clause of the last sentence
needs rewording.  I think it should read:
   " ..., the From header
   of the INVITE would indicate the referee, whereas a separate header
   would indicate the referrer."

Regards,
Mary.

-----Original Message-----
From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
Sent: Friday, June 13, 2003 2:03 AM
To: 'Robert Sparks'; Rohan Mahy
Cc: sip@ietf.org; 'Dean Willis'; Gonzalo.Camarillo@ericsson.com
Subject: RE: [Sip] WGLC for auth-id body



Thanks for these comments. A few notes below.

Jon Peterson
NeuStar, Inc.

[snip]
> I've reviewed this document (with revision of referredby
> particularly in mind) and have the following minor comments:
> 
> - It would help section 4 to more explicitly note that it
>   is discussing using AIBs to do something beyond(besides?)
>   providing integrity protection/authentication for the 
>   request it appears in.  It would also help to capture that
>   the presence of these other things don't preclude the section
>   2 use of an AIB. Perhaps this text at end of the first paragraph
>   of section 4:
> 
>      Such information might be carried in one or more supplemental
>      AIBs. The presence of these supplemental AIBs does not preclude
>      the use of AIB as specified in this document to protect the
>      message in which they appear.
> 

Yes, I do think it is important to capture the idea that there may be
multiple AIBs in a SIP message that identify different parties. The text
given above looks good to me.

> - Instead of a special case of INVITE (see the heading of section 4), I
>   think this simply a different use of an AIB. This document constrains
>   the use of AIBs for providing integrity protection of messages and
>   authenticating their sender regardless of method type. Section 4 is
>   trying to note that other AIBs might appear that do something 
>   different. Perhaps this section could be titled "Potential additional
>   uses of AIBs"?
> 

Well, although REFER-instigated INVITEs might not be a special case as such,
most would probably say that 3PCC is a special case. But still, we need a
section title that accommodates both, so I think that's fair; a better title
for this section is probably in order. I think we need something with the
sense "Use of AIBs to express the identity of someone other than the sender
of the INVITE". I'll try to figure out a way to compress that to
header-size.

> - Section 10 (Security Considerations): The first sentence needs to
>   be scoped to the section 2 use of AIBs, not all uses of AIBs with
>   message/sipfrag in them.
> 

Okay, fine.

> - Spelling Nits:
>    * Second sentence, last paragraph Page 4: 
>        s/SHOULD be added it to/SHOULD be added to/
>    * Second sentence, last paragraph, section 4, Page 6:
>        s/harder to correlated an AIB/harder to correlate an AIB/
> 

Thanks, I'll take care of all of those.

> Other than that, I believe this document is ready to go.
> 
> RjS
> 
> On Tue, 2003-05-13 at 19:39, Rohan Mahy wrote:
> > Hello Everyone,
> > 
> > I would like to begin Working Group Last Call on
> > 
> > 
> http://www.ietf.org/internet-drafts/draft-ietf-sip-authid-body-01.txt
> > 
> > WGLC will end on Friday, June 13, 2003.
> > 
> > thanks,
> > -rohan
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Mary Barnes'" <mbarnes@nortelnetworks.com>,
        "'Robert Sparks'"
	 <rsparks@dynamicsoft.com>,
        Rohan Mahy <rohan@cisco.com>
Cc: sip@ietf.org, "'Dean Willis'" <dean.willis@softarmor.com>,
        Gonzalo.Camarillo@ericsson.com
Subject: RE: [Sip] WGLC for auth-id body
Date: Fri, 13 Jun 2003 13:03:45 -0400
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Noted - yes, your text is superior. It'll be fixed. Thanks!

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Mary Barnes [mailto:mbarnes@nortelnetworks.com]
> Sent: Friday, June 13, 2003 9:10 AM
> To: 'Peterson, Jon'; 'Robert Sparks'; Rohan Mahy
> Cc: sip@ietf.org; 'Dean Willis'; Gonzalo.Camarillo@ericsson.com
> Subject: RE: [Sip] WGLC for auth-id body
> 
> 
> There's one more nit that I noticed that doesn't appear to be mentioned by
> Robert. 
> 
> In the second paragraph of section 4, the 2nd clause of the 
> last sentence
> needs rewording.  I think it should read:
>    " ..., the From header
>    of the INVITE would indicate the referee, whereas a separate header
>    would indicate the referrer."
> 
> Regards,
> Mary.
> 
> -----Original Message-----
> From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
> Sent: Friday, June 13, 2003 2:03 AM
> To: 'Robert Sparks'; Rohan Mahy
> Cc: sip@ietf.org; 'Dean Willis'; Gonzalo.Camarillo@ericsson.com
> Subject: RE: [Sip] WGLC for auth-id body
> 
> 
> 
> Thanks for these comments. A few notes below.
> 
> Jon Peterson
> NeuStar, Inc.
> 
> [snip]
> > I've reviewed this document (with revision of referredby
> > particularly in mind) and have the following minor comments:
> > 
> > - It would help section 4 to more explicitly note that it
> >   is discussing using AIBs to do something beyond(besides?)
> >   providing integrity protection/authentication for the 
> >   request it appears in.  It would also help to capture that
> >   the presence of these other things don't preclude the section
> >   2 use of an AIB. Perhaps this text at end of the first paragraph
> >   of section 4:
> > 
> >      Such information might be carried in one or more supplemental
> >      AIBs. The presence of these supplemental AIBs does not preclude
> >      the use of AIB as specified in this document to protect the
> >      message in which they appear.
> > 
> 
> Yes, I do think it is important to capture the idea that there may be
> multiple AIBs in a SIP message that identify different 
> parties. The text
> given above looks good to me.
> 
> > - Instead of a special case of INVITE (see the heading of 
> section 4), I
> >   think this simply a different use of an AIB. This 
> document constrains
> >   the use of AIBs for providing integrity protection of messages and
> >   authenticating their sender regardless of method type. 
> Section 4 is
> >   trying to note that other AIBs might appear that do something 
> >   different. Perhaps this section could be titled 
> "Potential additional
> >   uses of AIBs"?
> > 
> 
> Well, although REFER-instigated INVITEs might not be a 
> special case as such,
> most would probably say that 3PCC is a special case. But 
> still, we need a
> section title that accommodates both, so I think that's fair; 
> a better title
> for this section is probably in order. I think we need 
> something with the
> sense "Use of AIBs to express the identity of someone other 
> than the sender
> of the INVITE". I'll try to figure out a way to compress that to
> header-size.
> 
> > - Section 10 (Security Considerations): The first sentence needs to
> >   be scoped to the section 2 use of AIBs, not all uses of AIBs with
> >   message/sipfrag in them.
> > 
> 
> Okay, fine.
> 
> > - Spelling Nits:
> >    * Second sentence, last paragraph Page 4: 
> >        s/SHOULD be added it to/SHOULD be added to/
> >    * Second sentence, last paragraph, section 4, Page 6:
> >        s/harder to correlated an AIB/harder to correlate an AIB/
> > 
> 
> Thanks, I'll take care of all of those.
> 
> > Other than that, I believe this document is ready to go.
> > 
> > RjS
> > 
> > On Tue, 2003-05-13 at 19:39, Rohan Mahy wrote:
> > > Hello Everyone,
> > > 
> > > I would like to begin Working Group Last Call on
> > > 
> > > 
> > 
> http://www.ietf.org/internet-drafts/draft-ietf-sip-authid-body-01.txt
> > > 
> > > WGLC will end on Friday, June 13, 2003.
> > > 
> > > thanks,
> > > -rohan
> > > 
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> > 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From mailnull@www1.ietf.org  Mon Jun 16 01:15:05 2003
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From: "Chen Zaifeng,BISC R&D SA(BJ)" <zaifeng.chen@BISC.SIEMENS.COM.CN>
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Subject: [Sip] Complex status info transfer in SIP presence service
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Dear All:
I have questions about SIP based presence service:

1.  If a Registar is used to be a source of original presence info, how to transfer complex or multiple status
     info (not only Online and Offline), e.g. Busy, Idle, No Disturb, from Presentity to Registar? Can these
     status info be contained in an extension parameter of Contact header of Register message?
2. How about the solution that no Registar is used and Presentity push presence info directly to Presence 
    Server by sending of NOTIFY message?

Any of your ideas or answers will be highly appreciated! Thanks!

Sincerely yours, 
Chen Zaifeng
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From: "Chen Zaifeng,BISC R&D SA(BJ)" <zaifeng.chen@BISC.SIEMENS.COM.CN>
To: "'sip@ietf.org'" <sip@ietf.org>
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Dear All:
I found two statements in rfc3261 which seem to conflict each other:
1. In page 23, in the definition of "Proxy, Proxy Server", it says that "A proxy interprets, and, if necessary, rewrites 
    specific parts of a request message before forwarding it."
2. In page 100, under the title of "1. Copy request", it says that "The proxy MUST NOT add to, modify, or remove the 
    message body."

On the other hand, there are scenarios that need a proxy to modify a SIP message body, e.g. for SIP<-->PSTN/ISDN
calls, it is dangerous to delivery ISUP MIME content encapsulated in SIP message body to SIP user client, or receive
such ISUP content from SIP user client (in such cases the ISUP content might be mostly for malicious usage) and
forward it to somewhere like SIP-PSTN gateways, then a proxy might have to screen/remove ISUP MIME content
encapsulated in SIP message body.

Any of your ideas or clarifications to my puzzles are highly appreciated. Thanks!

Sincerely yours, 
Chen Zaifeng
--------------------------------------------------------------------------
Beijing International Switching System Corporation Ltd.
TD/DEW
No. 14 Jiu Xian Qiao Road
Beijing 100016, P. R. China
Tel:  +86-10-8457 1921
Fax: +86-10-6436 7732
E-mail: <mailto:zaifeng.chen@bisc.siemens.com.cn>



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



From mailnull@www1.ietf.org  Mon Jun 16 10:07:40 2003
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Chen Zaifeng,BISC R&D SA(BJ)'" <zaifeng.chen@bisc.siemens.com.cn>,
        "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] Never can a proxy modify SIP message body?
Date: Mon, 16 Jun 2003 04:14:15 -0500
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Some notes below.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Chen Zaifeng,BISC R&D SA(BJ)
> [mailto:zaifeng.chen@bisc.siemens.com.cn]
> Sent: Sunday, June 15, 2003 6:19 PM
> To: 'sip@ietf.org'
> Cc: 'Jonathan Rosenberg'
> Subject: [Sip] Never can a proxy modify SIP message body?
> 
> 
> Dear All:
> I found two statements in rfc3261 which seem to conflict each other:
> 1. In page 23, in the definition of "Proxy, Proxy Server", it 
> says that "A proxy interprets, and, if necessary, rewrites 
>     specific parts of a request message before forwarding it."
> 2. In page 100, under the title of "1. Copy request", it says 
> that "The proxy MUST NOT add to, modify, or remove the 
>     message body."
> 

I don't believe these conflict with one another. The first says that a proxy
can rewrite "specific parts" of a request message. Those "specific parts" do
not include the message body. The rules for proxy processing make it quite
clear, I think, which parts proxy servers operate on.

> On the other hand, there are scenarios that need a proxy to 
> modify a SIP message body, e.g. for SIP<-->PSTN/ISDN
> calls, it is dangerous to delivery ISUP MIME content 
> encapsulated in SIP message body to SIP user client, or receive
> such ISUP content from SIP user client (in such cases the 
> ISUP content might be mostly for malicious usage) and
> forward it to somewhere like SIP-PSTN gateways, then a proxy 
> might have to screen/remove ISUP MIME content
> encapsulated in SIP message body.
> 

At the risk of rehashing old issues, I agree that there are risks associated
with the dissemination of RFC3204 ISUP MIME bodies. However, there are
alternatives to mitigating these risks by violating the operational
guidelines for proxy servers. For example, S/MIME, which is
mandatory-to-implement for user agents that support SIP-T, can be used both
to encrypt ISUP MIME bodies (so these bodies cannot be inspected by any old
UAS) and to sign ISUP MIME bodies (so that a recipient can verify that the
ISUP was not created by any old UAC).

I believe it is more secure, overall, to manage these constraints with
cryptography than it is to deploy "screening" proxy servers that are not
RFC3261-compliant.

> Any of your ideas or clarifications to my puzzles are highly 
> appreciated. Thanks!
> 
> Sincerely yours, 
> Chen Zaifeng
> --------------------------------------------------------------
> ------------
> Beijing International Switching System Corporation Ltd.
> TD/DEW
> No. 14 Jiu Xian Qiao Road
> Beijing 100016, P. R. China
> Tel:  +86-10-8457 1921
> Fax: +86-10-6436 7732
> E-mail: <mailto:zaifeng.chen@bisc.siemens.com.cn>
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Mon Jun 16 15:37:52 2003
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From: "Jain, Rajnish" <rajnishjain@xl.com>
To: "Chen Zaifeng,BISC R&D SA(BJ)" <zaifeng.chen@BISC.SIEMENS.COM.CN>,
        <sip@ietf.org>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
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ISUP encapsulation w/in SIP is relevant to PSTN backhaul over 
an intermediary SIP network (not the other way round) such as 
the following:

         PSTN <-----> SIP Network <-----> PSTN 
    
The PSTN/SIP gateways at the two ends here are UAs. The SIP 
network in the middle is perhaps comprised of a chain of proxy 
servers. Your requirements may lead you to a B2BUA approach
which are entitled to do anything to the SIP messages at their
discretion.

- Rajnish

-----Original Message-----
From: Chen Zaifeng,BISC R&D SA(BJ)
[mailto:zaifeng.chen@BISC.SIEMENS.COM.CN]
Sent: Sunday, June 15, 2003 9:19 PM
To: 'sip@ietf.org'
Cc: 'Jonathan Rosenberg'
Subject: [Sip] Never can a proxy modify SIP message body?


Dear All:
I found two statements in rfc3261 which seem to conflict each other:
1. In page 23, in the definition of "Proxy, Proxy Server", it says that "A proxy interprets, and, if necessary, rewrites 
    specific parts of a request message before forwarding it."
2. In page 100, under the title of "1. Copy request", it says that "The proxy MUST NOT add to, modify, or remove the 
    message body."

On the other hand, there are scenarios that need a proxy to modify a SIP message body, e.g. for SIP<-->PSTN/ISDN
calls, it is dangerous to delivery ISUP MIME content encapsulated in SIP message body to SIP user client, or receive
such ISUP content from SIP user client (in such cases the ISUP content might be mostly for malicious usage) and
forward it to somewhere like SIP-PSTN gateways, then a proxy might have to screen/remove ISUP MIME content
encapsulated in SIP message body.

Any of your ideas or clarifications to my puzzles are highly appreciated. Thanks!

Sincerely yours, 
Chen Zaifeng
--------------------------------------------------------------------------
Beijing International Switching System Corporation Ltd.
TD/DEW
No. 14 Jiu Xian Qiao Road
Beijing 100016, P. R. China
Tel:  +86-10-8457 1921
Fax: +86-10-6436 7732
E-mail: <mailto:zaifeng.chen@bisc.siemens.com.cn>



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



From mailnull@www1.ietf.org  Mon Jun 16 21:51:19 2003
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Date: Mon, 16 Jun 2003 10:22:10 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sip] Complex status info transfer in SIP presence service
In-reply-to: <23BCA2174A77D111A52D00A0C967A6E2042016D2@HP5>
To: "'Chen Zaifeng,BISC R&D SA(BJ)'" <zaifeng.chen@BISC.SIEMENS.COM.CN>,
        sip@ietf.org
Message-id: <000001c3341b$0f0ac030$248823a6@hsinnreich2>
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Please see the I-D <draft-schulzrinne-simple-rpids-01.txt> at the
ietf.org archive.

Henry

Henry Sinnreich
MCI
400 International Parkway
Richardson, Texas 75081
USA
 

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Chen Zaifeng,BISC R&D SA(BJ)
> Sent: Sunday, June 15, 2003 8:41 PM
> To: 'sip@ietf.org'
> Subject: [Sip] Complex status info transfer in SIP presence service
> 
> 
> Dear All:
> I have questions about SIP based presence service:
> 
> 1.  If a Registar is used to be a source of original presence 
> info, how to transfer complex or multiple status
>      info (not only Online and Offline), e.g. Busy, Idle, No 
> Disturb, from Presentity to Registar? Can these
>      status info be contained in an extension parameter of 
> Contact header of Register message? 2. How about the solution 
> that no Registar is used and Presentity push presence info 
> directly to Presence 
>     Server by sending of NOTIFY message?
> 
> Any of your ideas or answers will be highly appreciated! Thanks!
> 
> Sincerely yours, 
> Chen Zaifeng
> --------------------------------------------------------------
> ------------
> Beijing International Switching System Corporation Ltd.
> TD/DEW
> No. 14 Jiu Xian Qiao Road
> Beijing 100016, P. R. China
> Tel:  +86-10-8457 1921
> Fax: +86-10-6436 7732
> E-mail: <mailto:zaifeng.chen@bisc.siemens.com.cn>
> 
> 
> 
> _______________________________________________
> 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] Never can a proxy modify SIP message body?
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There are two possibilities here: the SIP UA can legitimately send encapsulated ISUP 
to the network, but that ISUP has to be checked for valid and legitimate content, or 
the SIP UA has no right to send encapsulated ISUP.  In the first case, you need a UA 
to do the checking.  This can be a B2BUA or a SIP-to-ISUP gateway, with the former 
being the more likely solution.

If the SIP UA has no right to send encapsulated ISUP, a proxy can reject the message 
on the basis of the Content-Type header field.

Chen Zaifeng,BISC R&D SA(BJ) wrote:

> Dear All:
> I found two statements in rfc3261 which seem to conflict each other:
> 1. In page 23, in the definition of "Proxy, Proxy Server", it says that "A proxy interprets, and, if necessary, rewrites 
>     specific parts of a request message before forwarding it."
> 2. In page 100, under the title of "1. Copy request", it says that "The proxy MUST NOT add to, modify, or remove the 
>     message body."
> 
> On the other hand, there are scenarios that need a proxy to modify a SIP message body, e.g. for SIP<-->PSTN/ISDN
> calls, it is dangerous to delivery ISUP MIME content encapsulated in SIP message body to SIP user client, or receive
> such ISUP content from SIP user client (in such cases the ISUP content might be mostly for malicious usage) and
> forward it to somewhere like SIP-PSTN gateways, then a proxy might have to screen/remove ISUP MIME content
> encapsulated in SIP message body.
> 
> Any of your ideas or clarifications to my puzzles are highly appreciated. Thanks!
> 
> Sincerely yours, 
> Chen Zaifeng
> --------------------------------------------------------------------------
> Beijing International Switching System Corporation Ltd.
> TD/DEW
> No. 14 Jiu Xian Qiao Road
> Beijing 100016, P. R. China
> Tel:  +86-10-8457 1921
> Fax: +86-10-6436 7732
> E-mail: <mailto:zaifeng.chen@bisc.siemens.com.cn>
> 
> 
> 
> _______________________________________________
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> 

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Subject: [Sip] Bug? Matching rules of Server Transaction
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sipHi all,

  When I fulfill "Call pickup" service , I meet a problem about Server Transaction matching rules. As RFC3261 says in section 17.2.3:
<quote>
   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.
</quote>

   So, if the notifier notifies subscriber the resources state in two NOTIFY messages, as below:

             notifier                      subscriber
                |        ............          |
                |-------NOTIFY (F1)----------->|
                |<------200 OK (F2)------------|
                |                              |  the interval is less Timer J, J does not fire yet.
                |                              |
                |-------NOTIFY (F3)----------->|
                |<------200 OK (F4)------------|
                |                              |

  the message is :
<NOTIFY_F1>

 NOTIFY sip:bob@biloxi.example.com SIP/2.0
   Via: SIP/2.0/UDP Bob.biloxi.example.com:5060;branch=z9hG4bK74bf
   Max-Forwards: 70
   From: Bob <sip:bob@biloxi.example.com>;tag=31451098
   To: Carol <sip:carol@biloxi.example.com>;tag=8675309
   Call-ID: rt4353gs2egg@pc.biloxi.example.com
   CSeq: 1 NOTIFY
   Contact: <sip:carol@Bob.biloxi.example.com>
   Event: dialog
   Subscription-State: active;expires=3600
   Content-Type: application/dialog-info+xml
   Content-Length: ...

</NOTIFY_F1>

<NOTIFY_F2>

   NOTIFY sip:bob@biloxi.example.com SIP/2.0
   Via: SIP/2.0/UDP Bob.biloxi.example.com:5060;branch=z9hG4bK74bf
   Max-Forwards: 70
   From: Bob <sip:bob@biloxi.example.com>;tag=31451098
   To: Carol <sip:carol@biloxi.example.com>;tag=8675309
   Call-ID: rt4353gs2egg@pc.biloxi.example.com
   CSeq: 2 NOTIFY
   Contact: <sip:carol@Bob.biloxi.example.com>
   Event: dialog
   Subscription-State: active;expires=3600
   Content-Type: application/dialog-info+xml
   Content-Length: ...

</NOTIFY_F2>

  Notice that the two NOTIFY have the SAME branch & the magic cookie "z9hG4bK", using the matching rules in 17.2.3, all the three rules MATCH! So I can judge F3 belongs to the same transaction which created by F1, because timer J does not fire, and F2 is sent to response F1 , as per 17.2.2 of RFC3261, when F3 comes, I will think F3 is the retransmision of F2!
But they are not the same indeed: different CSeq and report different resource's state!

  Maybe I missing somthing, but I'm really puzzled .

  Thanks a lot to clarify!



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Subject: [Sip] Referredby revision
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I've just submitted
http://www.nostrum.com/~rjsparks/draft-ietf-sip-referredby-02.txt
to the repository.

I believe this document is ready for last call.

RjS


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From exim@www1.ietf.org  Wed Jun 18 10:33:02 2003
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Subject: RE: [Sip] RE: Use of AIB in referredby
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Mary Barnes <mbarnes@nortelnetworks.com>
Cc: "'Peterson, Jon'" <jon.peterson@neustar.biz>, sip@ietf.org
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It turns out that we already have an identifier that distinguishes
the AIB used for the refer token - it is called out explicitly by
the cid parameter to the Referred-By header field itself. 
A consumer won't be scanning all the AIBs in a message wondering
which one is the referred-by token.

So, I left it as 'aib' in referred-by-02.

RjS

On Fri, 2003-06-13 at 10:58, Mary Barnes wrote:
> My general preference is to be more explicit and towards that end I would
> think a different Content-Disposition disposition-type could be useful.
> That said, if AIB were as general as Robert suggested initially, then it
> could possibly be reused even for the Inserted headers (like History-Info),
> where I'm currently modeling the solution after AIB, but was planning on a
> new Content-Disposition disposition-type.  However, even I would agree that
> the Inserted Headers are more different than the original intent of AIB than
> the Referred-by use and again I do prefer to be more explicit.
> 
> Mary.
> 
> -----Original Message-----
> From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
> Sent: Friday, June 13, 2003 2:17 AM
> To: 'Robert Sparks'; sip@ietf.org
> Subject: [Sip] RE: Use of AIB in referredby
> 
> 
> 
> I look at the AIB as the application of sipfrags to the problem described in
> RFC3261 23.4 - the tunneling of integrity and authentication properties
> within the MIME bodies of SIP messages. In the AIB, these properties are
> intended to provide identity, hence the name 'authenticated identity body'.
> I think that the usage in referredby is sufficiently close to that purpose,
> providing authenticated identity within a body, that I don't think we have a
> problem with reuse of the term.
> 
> The real question, I think, is whether or not there is any value in
> differentiating an AIB resulting from a REFER from any other AIB (such as a
> 'normal' one representing the sender of an INVITE) that might be in a
> request. Surely these need to be differentiated somehow - the signature on
> the AIB itself would be one indicator, as would the headers in the body, but
> Content-Disposition is another point at which the two could be
> distinguished. 
> 
> I don't have a strong intuition about whether or not the use of a different
> Content-Disposition would be valuable.
> 
> Jon Peterson
> NeuStar, Inc.
> 
> > -----Original Message-----
> > From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
> > Sent: Monday, June 09, 2003 1:00 PM
> > To: sip@ietf.org
> > Cc: Jon Peterson
> > Subject: Use of AIB in referredby
> > 
> > 
> > Several weeks ago, Pekka made a suggestion that I would like to
> > follow up on now (sorry for the very long delay Pekka). His 
> > suggestion was to use a different Content-Disposition
> > disposition-type for referredby-tokens.
> > 
> > This draws attention to what the "aib" value means. 
> > 
> > Is it merely a synonym for "signed message/sipfrag"? If so, the
> > authid-body draft defines the value _and_ specifies one particular
> > use of a body with that disposition type (providing integrity and
> > authentication of sender for a single message). Other uses are valid,
> > so we should reuse it and not add noise to the IANA registry.
> > 
> > If, instead, "aib" means "signed message/sipfrag with these
> > particular security implications", then our referredby token
> > is _not_ an aib. It's something _like_ an aib, but with 
> > slightly different requirements, and artifacts in the protocol
> > (like the disposition-type) should probably reflect that. If this
> > is the case, referredby could register something like "RBIB"
> > (ReferredBy Identity Body), or, treading much more dangerous ground,
> > "TPIB" (Third Party Identity Body).
> > 
> > So, which of the above meanings for AIB is auth-id body establishing?
> > 
> > RjS
> > 
> > 
> _______________________________________________
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Hi all, 

Sorry to use this list for SLP.

SIP server can be located by using SLP (service location protocol), is there
any mailing list for SLP related topic?

I tried srvloc-request@srvloc.org, but failed. Is this list still operating
or using another one instead?

Thanks

Xin Chen
Fujitsu Laboratories of Europe  


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From exim@www1.ietf.org  Thu Jun 19 00:38:56 2003
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From: "A. N. Gautham" <gautham@mistralsoftware.com>
To: y01317@njupt.edu.cn
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Subject: Re: [Sip] Bug? Matching rules of Server Transaction
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Hi,
	If an entity inserts a branch parameter that begins with the magic cookie, then it is 3261 compliant. In 3261, the branch parameter has to be unique. As given in Section 8.1.1.7 of 3261, " The branch parameter value MUST be unique across space and time for
   all requests sent by the UA.  The exceptions to this rule are CANCEL and ACK for non-2xx responses." Thus, the two NOTIFY requests (F1) and (F3) MUST have different branch parameters since they belong to different transactions. Since this has not been followed by the 'notifier', you are finding that F3 gets treated as a retransmission of F1.
	Hope this helps

-Gautham.
	


On Tue, 17 Jun 2003 21:10:3 +0800
FengZhang <y01317@njupt.edu.cn> wrote:

> sipHi all,
> 
>   When I fulfill "Call pickup" service , I meet a problem about Server Transaction matching rules. As RFC3261 says in section 17.2.3:
> <quote>
>    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.
> </quote>
> 
>    So, if the notifier notifies subscriber the resources state in two NOTIFY messages, as below:
> 
>              notifier                      subscriber
>                 |        ............          |
>                 |-------NOTIFY (F1)----------->|
>                 |<------200 OK (F2)------------|
>                 |                              |  the interval is less Timer J, J does not fire yet.
>                 |                              |
>                 |-------NOTIFY (F3)----------->|
>                 |<------200 OK (F4)------------|
>                 |                              |
> 
>   the message is :
> <NOTIFY_F1>
> 
>  NOTIFY sip:bob@biloxi.example.com SIP/2.0
>    Via: SIP/2.0/UDP Bob.biloxi.example.com:5060;branch=z9hG4bK74bf
>    Max-Forwards: 70
>    From: Bob <sip:bob@biloxi.example.com>;tag=31451098
>    To: Carol <sip:carol@biloxi.example.com>;tag=8675309
>    Call-ID: rt4353gs2egg@pc.biloxi.example.com
>    CSeq: 1 NOTIFY
>    Contact: <sip:carol@Bob.biloxi.example.com>
>    Event: dialog
>    Subscription-State: active;expires=3600
>    Content-Type: application/dialog-info+xml
>    Content-Length: ...
> 
> </NOTIFY_F1>
> 
> <NOTIFY_F2>
> 
>    NOTIFY sip:bob@biloxi.example.com SIP/2.0
>    Via: SIP/2.0/UDP Bob.biloxi.example.com:5060;branch=z9hG4bK74bf
>    Max-Forwards: 70
>    From: Bob <sip:bob@biloxi.example.com>;tag=31451098
>    To: Carol <sip:carol@biloxi.example.com>;tag=8675309
>    Call-ID: rt4353gs2egg@pc.biloxi.example.com
>    CSeq: 2 NOTIFY
>    Contact: <sip:carol@Bob.biloxi.example.com>
>    Event: dialog
>    Subscription-State: active;expires=3600
>    Content-Type: application/dialog-info+xml
>    Content-Length: ...
> 
> </NOTIFY_F2>
> 
>   Notice that the two NOTIFY have the SAME branch & the magic cookie "z9hG4bK", using the matching rules in 17.2.3, all the three rules MATCH! So I can judge F3 belongs to the same transaction which created by F1, because timer J does not fire, and F2 is sent to response F1 , as per 17.2.2 of RFC3261, when F3 comes, I will think F3 is the retransmision of F2!
> But they are not the same indeed: different CSeq and report different resource's state!
> 
>   Maybe I missing somthing, but I'm really puzzled .
> 
>   Thanks a lot to clarify!
> 
> 
> 
> _______________________________________________
> 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 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 SIP Referred-By Mechanism
	Author(s)	: R. Sparks
	Filename	: draft-ietf-sip-referredby-02.txt
	Pages		: 26
	Date		: 2003-6-18
	
The SIP REFER method [2] provides a mechanism where one party (the
referrer) gives a second party (the referree) an arbitrary URI to
reference.  If that URI is a SIP URI, the referree will send a SIP
request, often an INVITE, to that URI (the refer target).  This
document extends the REFER method allowing the referrer to provide
information about the reference to the refer target using the
referree as an intermediary.  This information includes the identity
of the referrer and the URI to which the referrer referred.  The
mechanism utilizes S/MIME to help protect this information from a
malicious intermediary.  This protection is optional, but a recipient
may refuse to accept a request unless it is present.

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From exim@www1.ietf.org  Thu Jun 19 12:22:40 2003
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To: jdrosen@dynamicsoft.com
Cc: rohan@cisco.com, dwillis@dynamicsoft.com, sip@ietf.org,
        jon.peterson@neustar.biz, hardie@qualcomm.com, erik.nordmark@sun.com
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Date: Thu, 19 Jun 2003 09:21:05 -0700
From: Allison Mankin <mankin@psg.com>
Message-Id: <E19T29q-000CDd-EG@psg.com>
Subject: [Sip] IESG Discuss Comments on Symmetric Response Routing
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The IESG reviewed draft-ietf-sip-symmetric-response-00.txt and there were
two sets of Discuss comments.  They do not seem too hard to address.  In
addition, the FQDNs in the draft need to be changed to example.com.

Below are the comments.  They are also in the ID tracker.

Allison

Comments:

Ted Hardie:
First, let me say that a large number of the issues I had
with the document turned out to be very well documented
in the IAB considerations section of the draft. A few
other issues:

In Section 3:

      When the client sends the request, if the request is sent using UDP,
        the client MUST be prepared to receive the response on the same
        socket the request was sent on. Specifically, it MUST be prepared to
        receive the response on the same IP address and port present in the
        source IP address and source port of the request. For backwards
        compatibility, the client MUST still be prepared to receive a
        response on the port indicated in the sent-by field of the topmost
        Via header field value, as specified in Section 18.1.1 of SIP [1].


It seems pretty clear from the "on the same socket" that the following
sentence means the IP address and port present in the source IP
and port of the request _when it is sent_. I think it would be clearer,
though, to use phrasing like "it MUST be prepared to receive the
response on the same IP address and port it used to populate
the source IP address and source port of the request.".

Also in Section 3:

        To keep the binding fresh, the client SHOULD retransmit its INVITE every 20
        seconds or so. These retransmissions will need to take place even
        after receiving a provisional response.

based on the idea the one minute seems to be a common UDP binding lifetime.
Section 9.3 notes, however, that there is no way to determine the UDP binding
lifetime. Is there anyway to introduce a growing/shrinking algorithm
to this for cases where the binding lifetime is much longer (to avoid the
aggressiveness of 20 seconds for an arbitrary period of time) or to handle
the cases where the binding lifetime is shorter than 20+transmission time to
the NAT (since this has to handle the NAT being at some arbitrary place in
the topology).

In the initial section of 9:

        The client can then perform an additional registration,
        using this address in a Contact header. This would allow a client to
        receive incoming requests, such as INVITE, on the socket through
        which the registration was sent.

As is noted below that point, there are cases in which the port binding
is only valid for the server to which the original registration was sent.
Will the Contact header "leak" that so that a direction connection between a
different sip agent (user or proxy) might attempt to use it? If so, a forward
pointer to the limitation might be useful here.

I would personally consider the UNSAF document a normative reference,
but this is not a big deal.

Erik Nordmark:
 The security considerations section doesn't set a very good example. 
 It could easily be read as "we didn't look hard for security issues".

 Thus I'd like it to provide the reasoning that lead to thinking
 that there are no added security issues. Presumably a paragraph or two would
 suffice.





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



From exim@www1.ietf.org  Thu Jun 19 14:39:34 2003
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Cc: rohan@cisco.com, Jon Peterson <jon.peterson@neustar.biz>,
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Hello Everyone,

I would like to begin a Working Group last call on:

http://www.ietf.org/internet-drafts/draft-ietf-sip-referredby-02.txt

WGLC will end on Friday July 18th.

thanks,
-rohan


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From exim@www1.ietf.org  Thu Jun 19 15:25:37 2003
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From: The IESG <iesg-secretary@ietf.org>
Reply-to: iesg@ietf.org
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Date: Thu, 19 Jun 2003 15:20:22 -0400
Subject: [Sip] Last Call: A Mechanism for Content Indirection in Session
 Initiation Protocol (SIP) Messages to a Proposed Standard
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---------------
The IESG has received a request from Session Initiation Protocol WG to 
consider the following Internet-Draft(s) as a Proposed Standard. 
o Session Initiation Protocol (SIP): A Mechanism for Content Indirection 
  in Session Initiation Protocol (SIP) Messages 
    <draft-ietf-sip-content-indirect-mech-03.txt>
    Proposed Standard
                                                                                       

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2003-07-03.
                                                                                       
Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-content-indirect-mech-03.txt

_______________________________________________
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 exim@www1.ietf.org  Thu Jun 19 15:48:35 2003
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Date: Thu, 19 Jun 2003 15:45:08 -0400
Subject: [Sip] Last Call: The Stream Control Transmission Protocol as a Transport
 for for the Session Initiation Protocol to a Proposed Standard
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---------------
The IESG has received a request from Session Initiation Protocol WG to 
consider the following Internet-Draft(s) as a Proposed Standard. 
o Session Initiation Protocol (SIP): The Stream Control Transmission 
  Protocol as a Transport for for the Session Initiation Protocol 
    <draft-ietf-sip-sctp-03.txt>
    Proposed Standard
                                                                                       

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2003-07-03.
                                                                                       
Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-sctp-03.txt

_______________________________________________
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 exim@www1.ietf.org  Fri Jun 20 07:35:41 2003
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Subject: [Sip] I-D ACTION:draft-camarillo-sip-parameter-registry-01.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: The Internet Assigned Number Authority Header Field
                          Parameter Registry for the Session Initiation Protocol
	Author(s)	: G. Camarillo
	Filename	: draft-camarillo-sip-parameter-registry-01.txt
	Pages		: 6
	Date		: 2003-6-19
	
This document creates an IANA registry for SIP header field
parameters. It also lists the already existing parameters to be used
as initial values for that registry.

.

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

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



From exim@www1.ietf.org  Fri Jun 20 08:22:31 2003
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Date: Fri, 20 Jun 2003 07:34:58 -0400
Subject: [Sip] I-D ACTION:draft-camarillo-sip-uri-parameter-reg-00.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: The Internet Assigned Number Authority Universal 
                          Resource Identifier Parameter Registry for the Session
                          Initiation Protocol
	Author(s)	: G. Camarillo
	Filename	: draft-camarillo-sip-uri-parameter-reg-00.txt
	Pages		: 5
	Date		: 2003-6-19
	
This document creates an IANA registry for SIP URI and SIPS URI
parameters. It also lists the already existing parameters to be used
as initial values for that registry.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-camarillo-sip-uri-parameter-reg-00.txt

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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From exim@www1.ietf.org  Fri Jun 20 12:37:37 2003
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From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'Robert Sparks'" <rsparks@dynamicsoft.com>, sip@ietf.org
Cc: Jon Peterson <jon.peterson@neustar.biz>,
        Dean Willis
	 <dean.willis@softarmor.com>,
        "'Rohan Mahy'" <rohan@cisco.com>
Subject: RE: [Sip] WGLC for referred-by
Date: Fri, 20 Jun 2003 11:36:12 -0500
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There were a couple points that I made on the list around the -01 version
that don't seem to have been addressed
http://www1.ietf.org/mail-archive/working-groups/sip/current/msg07955.html
and I couldn't find that they had been responded to on the list in the
archives: 

Editorial Nits:
-------------------
- References: authid-body and cc-transfer need upversioning.

Other items:
--------------------------------------
- Section 2.3: Last paragraph: should this be a MUST (reject an otherwise
well-formed request with an invalid token)?   Reasonably, one should, but
perhaps there are users that would still like to be able to decide
themselves, thus I would suggest this be stated similar to the previous
paragraph as:

"The refer target SHOULD reject an otherwise well-formed request with an
invalid Referred-By token (see Section 4) with a 429 error response. If the
agent chooses to proceed with the request and provides any information from
the Referred-By header to its user as part of processing the request, it
MUST notify the user that the information was determined to be invalid." 

My reasoning is that it just seems that if an optional parm is mucked up (in
general or from a security perspective), then following the adage of being
generous in what you accept ...that whether to accept the request, provided
that it is warned that it's bad,  should still be up to the user.  This also
seems consistent with the MAYs in the previous 2 paragraphs. I do understand
the reasoning that you'd want to let the Referrer know and that by allowing
the request, you're actually bypassing the security put in place to keep the
bad guys from mucking with the messages, but again since it's optional, it
just seems that you can't make it stronger than a SHOULD. 

- Section 4.1:  2nd paragraph suggesting that "A target SHOULD verify the
request...".  I don't think this is useful since retargeting makes this
check not meaningful.  UNLESS, of course, you're using History-Info.  With
History-Info, you could verify that the Refer-To matches one of the
Targeted-to-URIs (and of course, this implies that this information has all
been sent securely, not mucked with by the proxies, etc.). 

Regards,
Mary.  

-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]
Sent: Thursday, June 19, 2003 1:39 PM
To: sip@ietf.org
Cc: rohan@cisco.com; Jon Peterson; Dean Willis; 'Robert Sparks'
Subject: [Sip] WGLC for referred-by


Hello Everyone,

I would like to begin a Working Group last call on:

http://www.ietf.org/internet-drafts/draft-ietf-sip-referredby-02.txt

WGLC will end on Friday July 18th.

thanks,
-rohan


_______________________________________________
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] WGLC for referred-by
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Mary Barnes <mbarnes@nortelnetworks.com>
Cc: sip@ietf.org, Jon Peterson <jon.peterson@neustar.biz>,
        Dean Willis <dean.willis@softarmor.com>,
        "'Rohan Mahy'" <rohan@cisco.com>
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On Fri, 2003-06-20 at 11:36, Mary Barnes wrote:
> There were a couple points that I made on the list around the -01 version
> that don't seem to have been addressed
> http://www1.ietf.org/mail-archive/working-groups/sip/current/msg07955.html
> and I couldn't find that they had been responded to on the list in the
> archives: 
> 
> Editorial Nits:
> -------------------
> - References: authid-body and cc-transfer need upversioning.

Hmm - I'm generating this using the xmlrefs stuff from xml2rfc and
I _did_ update my copy of that content before building this doc. 
Something is broken. I'll fix it when I cut the version that pulls
out the "changes since" section.

I failed to respond to the following two points on list. This was not
intentional, and I apologize.

> Other items:
> --------------------------------------
> - Section 2.3: Last paragraph: should this be a MUST (reject an otherwise
> well-formed request with an invalid token)?   Reasonably, one should, but
> perhaps there are users that would still like to be able to decide
> themselves, thus I would suggest this be stated similar to the previous
> paragraph as:

> "The refer target SHOULD reject an otherwise well-formed request with an
> invalid Referred-By token (see Section 4) with a 429 error response. If the
> agent chooses to proceed with the request and provides any information from
> the Referred-By header to its user as part of processing the request, it
> MUST notify the user that the information was determined to be invalid." 
> 
> My reasoning is that it just seems that if an optional parm is mucked up (in
> general or from a security perspective), then following the adage of being
> generous in what you accept ...that whether to accept the request, provided
> that it is warned that it's bad,  should still be up to the user.  This also
> seems consistent with the MAYs in the previous 2 paragraphs. I do understand
> the reasoning that you'd want to let the Referrer know and that by allowing
> the request, you're actually bypassing the security put in place to keep the
> bad guys from mucking with the messages, but again since it's optional, it
> just seems that you can't make it stronger than a SHOULD. 

This situation for this clause is different from the preceding MAYs.

Here, you _know_ you've got something invalid, where the two earlier
paragraphs deal with the case that you have data that you have no
indication of its integrity one way or another.

If the token is invalid, something's gone wrong and it will be
sufficiently hard to tell whether that's due to attack or implementation
error, that the action needs to be to return an error.

Diving into this just a little more deeply:

Well, you've got two classes of possible "invalidity".

1) The signature is not valid. Here, you have now way to know what
   got mucked with. No way to know if the failure is related to an
   optional parameter or something important. Clearly, this condition 
   warrants only rejection.

2) The signature is valid, but the contents of the token don't
   match the request. You or I might be able to inspect the message
   and make an educated decision about the safety of accepting
   the request. Ordinary people won't have that skill. (Ordinary
   devices currently aren't making it easy for me to get the
   information I would need to make that educated decision anyhow).
   So, again, rejection is warranted.

> 
> - Section 4.1:  2nd paragraph suggesting that "A target SHOULD verify the
> request...".  I don't think this is useful since retargeting makes this
> check not meaningful.  UNLESS, of course, you're using History-Info.  With
> History-Info, you could verify that the Refer-To matches one of the
> Targeted-to-URIs (and of course, this implies that this information has all
> been sent securely, not mucked with by the proxies, etc.). 

There is a lot of information present than just the Request-URI that
can be validated. The method, requested end-to-end headers, etc.

> 
> Regards,
> Mary.  
> 
> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: Thursday, June 19, 2003 1:39 PM
> To: sip@ietf.org
> Cc: rohan@cisco.com; Jon Peterson; Dean Willis; 'Robert Sparks'
> Subject: [Sip] WGLC for referred-by
> 
> 
> Hello Everyone,
> 
> I would like to begin a Working Group last call on:
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sip-referredby-02.txt
> 
> WGLC will end on Friday July 18th.
> 
> thanks,
> -rohan
> 
> 
> _______________________________________________
> 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 exim@www1.ietf.org  Fri Jun 20 16:00:58 2003
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Message-ID: <870397D7C140C84DB081B88396458DAF578A0A@zrc2c000.us.nortel.com>
From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'Robert Sparks'" <rsparks@dynamicsoft.com>
Cc: sip@ietf.org, Jon Peterson <jon.peterson@neustar.biz>,
        Dean Willis
	 <dean.willis@softarmor.com>,
        "'Rohan Mahy'" <rohan@cisco.com>
Subject: RE: [Sip] WGLC for referred-by
Date: Fri, 20 Jun 2003 14:58:45 -0500
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Robert,

Thanks for your response; I just have a couple of final comments embedded
below [MB].

Mary.

-----Original Message-----
From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
Sent: Friday, June 20, 2003 2:18 PM
To: Barnes, Mary [NGC:B602:EXCH]
Cc: sip@ietf.org; Jon Peterson; Dean Willis; 'Rohan Mahy'
Subject: RE: [Sip] WGLC for referred-by




On Fri, 2003-06-20 at 11:36, Mary Barnes wrote:
> There were a couple points that I made on the list around the -01 version
> that don't seem to have been addressed
> http://www1.ietf.org/mail-archive/working-groups/sip/current/msg07955.html
> and I couldn't find that they had been responded to on the list in the
> archives: 
> 
> Editorial Nits:
> -------------------
> - References: authid-body and cc-transfer need upversioning.

Hmm - I'm generating this using the xmlrefs stuff from xml2rfc and
I _did_ update my copy of that content before building this doc. 
Something is broken. I'll fix it when I cut the version that pulls
out the "changes since" section.

I failed to respond to the following two points on list. This was not
intentional, and I apologize.

> Other items:
> --------------------------------------
> - Section 2.3: Last paragraph: should this be a MUST (reject an otherwise
> well-formed request with an invalid token)?   Reasonably, one should, but
> perhaps there are users that would still like to be able to decide
> themselves, thus I would suggest this be stated similar to the previous
> paragraph as:

> "The refer target SHOULD reject an otherwise well-formed request with an
> invalid Referred-By token (see Section 4) with a 429 error response. If
the
> agent chooses to proceed with the request and provides any information
from
> the Referred-By header to its user as part of processing the request, it
> MUST notify the user that the information was determined to be invalid." 
> 
> My reasoning is that it just seems that if an optional parm is mucked up
(in
> general or from a security perspective), then following the adage of being
> generous in what you accept ...that whether to accept the request,
provided
> that it is warned that it's bad,  should still be up to the user.  This
also
> seems consistent with the MAYs in the previous 2 paragraphs. I do
understand
> the reasoning that you'd want to let the Referrer know and that by
allowing
> the request, you're actually bypassing the security put in place to keep
the
> bad guys from mucking with the messages, but again since it's optional, it
> just seems that you can't make it stronger than a SHOULD. 

This situation for this clause is different from the preceding MAYs.

Here, you _know_ you've got something invalid, where the two earlier
paragraphs deal with the case that you have data that you have no
indication of its integrity one way or another.

If the token is invalid, something's gone wrong and it will be
sufficiently hard to tell whether that's due to attack or implementation
error, that the action needs to be to return an error.

Diving into this just a little more deeply:

Well, you've got two classes of possible "invalidity".

1) The signature is not valid. Here, you have now way to know what
   got mucked with. No way to know if the failure is related to an
   optional parameter or something important. Clearly, this condition 
   warrants only rejection.

2) The signature is valid, but the contents of the token don't
   match the request. You or I might be able to inspect the message
   and make an educated decision about the safety of accepting
   the request. Ordinary people won't have that skill. (Ordinary
   devices currently aren't making it easy for me to get the
   information I would need to make that educated decision anyhow).
   So, again, rejection is warranted.

[MB]: I do understand your point, but I'm still wondering whether it
shouldn't be a SHOULD or perhaps RECOMMENDED rather than a MUST (to allow
for devices that might be able to provide a different level of user
interface for that scenario).  I do realize that the user interface should
not require a high level of technical knowledge, but I still think that a
specific implementation shouldn't be limited by the protocol (eg there may
be an "Advanced User" mode whereby this can be appropriately handled and a
user wants to know when they're being "attacked").  

> 
> - Section 4.1:  2nd paragraph suggesting that "A target SHOULD verify the
> request...".  I don't think this is useful since retargeting makes this
> check not meaningful.  UNLESS, of course, you're using History-Info.  With
> History-Info, you could verify that the Refer-To matches one of the
> Targeted-to-URIs (and of course, this implies that this information has
all
> been sent securely, not mucked with by the proxies, etc.). 

There is a lot of information present than just the Request-URI that
can be validated. The method, requested end-to-end headers, etc.

[MB]: I agree. I think my point initially arose because I read that Note as
directly coupled with the first, so it might be useful to add a 2nd sentence
between the current two and RECOMMEND some of the comparisons that should be
done to verify the identity, since you cannot rely on the Request-URI.

> 
> Regards,
> Mary.  
> 
> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: Thursday, June 19, 2003 1:39 PM
> To: sip@ietf.org
> Cc: rohan@cisco.com; Jon Peterson; Dean Willis; 'Robert Sparks'
> Subject: [Sip] WGLC for referred-by
> 
> 
> Hello Everyone,
> 
> I would like to begin a Working Group last call on:
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sip-referredby-02.txt
> 
> WGLC will end on Friday July 18th.
> 
> thanks,
> -rohan
> 
> 
> _______________________________________________
> 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 exim@www1.ietf.org  Mon Jun 23 01:37:28 2003
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From: Klaus Umschaden <Klaus.Umschaden@tuwien.ac.at>
Date: Wed, 18 Jun 2003 13:27:15 +0200
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I think that this draft may be of interest for this mailing group.

Best regards,
=09Klaus

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

Subject: I-D ACTION:draft-umschaden-smime-midcom-sip-proxy-00.txt

A New Internet-Draft is available from the on-line Internet-Drafts
directories.


=09Title=09=09: End-to-end Security for Firewall/NAT Traversal within
                          the Session Initiation Protocol (SIP)
=09Author(s)=09: K. Umschaden et al.
=09Filename=09: draft-umschaden-smime-midcom-sip-proxy-00.txt
=09Pages=09=09: 37
=09Date=09=09: 2003-5-6

This document describes an extension for the Session Initiation
Protocol (SIP), which enables end-to-end security of the Session
Description Protocol (SDP) together with firewall/Network Address
Translation (NAT) traversal. This solution relies on Secure
Multipurpose Internet Mail Extension (S/MIME) and the middlebox
communications (MIDCOM) protocol. The user authorises a proxy server
to encrypt the session description on behalf of the user. The proxy
determines the capabilities of the receiving party and encrypts the
SDP for a SIP proxy server in the receiving domain. Using MIDCOM,
each proxy can dynamically control its firewall to open pinholes or
request NAT bindings for the media flows. As long as each end-user
may contact its trustworthy SIP proxy via a secure connection and
authorise this proxy to encrypt the signalling data, the session
information is secured end-to-end.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-umschaden-smime-midcom-sip-prox=
y-00
=2Etxt

To remove yourself from the IETF Announcement list, send a message to
ietf-announce-request with the word unsubscribe in the body of the messag=
e.

Internet-Drafts are also available by anonymous FTP. Login with the usern=
ame
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
=09"get draft-umschaden-smime-midcom-sip-proxy-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.=
txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
=09mailserv@ietf.org.
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=09"FILE /internet-drafts/draft-umschaden-smime-midcom-sip-proxy-00.txt".

NOTE:=09The mail server at ietf.org can return the document in
=09MIME-encoded form by using the "mpack" utility.  To use this
=09feature, insert the command "ENCODING mime" before the "FILE"
=09command.  To decode the response(s), you will need "munpack" or
=09a MIME-compliant mail reader.  Different MIME-compliant mail readers
=09exhibit different behavior, especially when dealing with
=09"multipart" MIME messages (i.e. documents which have been split
=09up into multiple messages), so check your local documentation on
=09how to manipulate these messages.


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<ftp://ftp.ietf.org/internet-drafts/draft-umschaden-smime-midcom-sip-prox=
y-00
=2Etxt>

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


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From exim@www1.ietf.org  Mon Jun 23 03:02:47 2003
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To: sip@ietf.org
Date: Mon, 23 Jun 2003 06:57:39 +0000
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Subject: [Sip] Implementing Kerberos in SIP
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Hi,

The draft draft-wagle-sip-kerbpki-00 has already expired.
Does any one know of updates to this or new drafts about implementation of 
Kerberos in SIP.

James

_________________________________________________________________
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From exim@www1.ietf.org  Mon Jun 23 05:53:14 2003
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Subject: [Sip] new connection reuse draft available
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Hello,

I just submitted an updated version of my connection reuse draft to SIP  
as an individual submission. In the interim, it is available at the  
links below.

I tried to clarify the proposed mechanism which was in the requirements  
draft, and for the next revision I will write real Client  
Behavior/Server Behavior sections.  My hope is that this mechanism  
draft can become the basis for a WG item in SIP.

thanks,
-rohan
(as an individual WG member/contributor)


http://www.softarmor.com/sipwg/drafts/draft-mahy-sip-connect-reuse- 
00.html
http://www.softarmor.com/sipwg/drafts/draft-mahy-sip-connect-reuse- 
00.txt

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This list is for NEW development of the core SIP Protocol
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From exim@www1.ietf.org  Tue Jun 24 00:33:41 2003
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Date: Tue, 24 Jun 2003 09:54:09 +0530
From: Ranjit Avasarala <ranjitk@samsung.com>
Subject: RE: [Sip] SDP doubts
In-reply-to: <0C674B14EAEBD61196D900B0D03DB49F04FDE1@blrw502w.blr.infineon.com>
To: "'Shetty Bharathraj (IFIN DC COM)'" <Bharathraj.Shetty@infineon.com>,
        sip@ietf.org
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As per SDP grammer, \r\n ie CRLF is mandatory at the end of each SDp
field. Also connection field is mandatory. So any standard SDP parser /
builder should contain be able to interpret and buid them.
So in ur case provide the required values for c= and provide CRLF at the
end of each SDP line to test ur SDP implementation

Ranjit

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Shetty
Bharathraj (IFIN DC COM)
Sent: Monday, June 23, 2003 5:57 PM
To: 'sip@ietf.org'
Subject: [Sip] SDP doubts


HI

I have some questions on implementation of SDP decoder.

 We are doing interoperable testing our SIP stack with RADVISION prolab
manager, 
the SDP test scripts available with this tools are not consistent with
the syntax of RFC 2327. Some of them are as follows.

*	In Some of SDP test scripts, CRLF or  LF is missing at the end
of
SDP script(Last field).
*	The connection field(c=) and time(t=) are not mandatory in
scripts
available in prolab manager.
*	The order of the fields, email field and phone fields are not as
per
the RFC2327(e-mail field should come before the phone field).

How do we go ahead with the testing of SDP with these kind of test
cases?.

Regards
Bharath


Bharathraj Shetty A.N.
Software Engineer,
Infineon Technologies India Pvt.Ltd. 
10th Floor, Discoverer Block, 
International Tech Park, 
Whitefield Road, 
Bangalore - 560 066. India. 
Tel     :+91-80-8410017/18   (Extn.2039) 
Fax    :+91-80-8410012 
email : bharathraj.shetty@infineon.com  
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Use sip-implementors@cs.columbia.edu for questions on current sip Use
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From exim@www1.ietf.org  Tue Jun 24 07:01:48 2003
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To: IETF-Announce: ;
Cc: sip@ietf.org
From: Internet-Drafts@ietf.org
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Date: Tue, 24 Jun 2003 07:00:47 -0400
Subject: [Sip] I-D ACTION:draft-ietf-sip-history-info-00.txt
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--NextPart

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

	Title		: An Extension to the Session Initiation Protocol for 
                          Request History Information
	Author(s)	: M. Barnes
	Filename	: draft-ietf-sip-history-info-00.txt
	Pages		: 23
	Date		: 2003-6-23
	
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 HistInfo should be returned in responses to a request which has 
captured the history information.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-00.txt

To remove yourself from the IETF Announcement list, send a message to 
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Internet-Drafts are also available by anonymous FTP. Login with the username
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type "cd internet-drafts" and then
	"get draft-ietf-sip-history-info-00.txt".

A list of Internet-Drafts directories can be found in
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
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Subject: [Sip] Comments of Resource Priority header
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To all,

Comments on draft-ietf-sip-resource-priority-00:

The abstract says "defines a new SIP header field ... called 
Resource-Priority", but the draft then defines two headers. If the second header is included 
(see comments below), references should be added in the abstract and throughout 
the Introduction.

In the statement in the abstract that this header "does not influence the 
forwarding behavior of IP routers", it needs to be made clear that, while there 
is no "direct" influence, it is certainly possible for such influence to exist, 
based on procedures and processes not described in this document. For 
example, the descriptions for "Assured Service" have always presumed that this 
Resource-Priority header in the session (call) setup signaling could be used to 
cause preferential treatment of media packet forwarding, including multiple drop 
precedences in an EF queue. I believe the sentence should read: "While it does 
not directly influence the forwarding behavior of IP routers, procedures for 
using this header to cause such influence may be defined in other documents."

3. The definition of the Resource-Priority header should not allow multiple 
resource values to be carried. I am unaware of any requirement for this. The 
use described always needs exactly one value within one namespace. Anything 
beyond this will bring on questions like "Which priority has priority?"

Note: REQ-13 says that "Some UAs will need to support multiple different 
priority schemes". It does not say that the indication (header) needs to support a 
"list of names".

3. Instead of defining a separate Accept header, it seems that there could be 
a single Resource-priority Header which is then included in specific messages 
(responses) to mean "accept". (I'm trying to find an existing example of a 
header that is used like this.)

The example of the originator requesting "q735.1, gets.enabled" and the 
response indicating which dsn values are supported is invalid. The dsn would not 
function this way. If it understands the value received based on interworking 
arrangements (e.g., gets.enabled), it may convert it to its equivalent dsn 
value, according to the interworking rules. Or if it is set up to understand and 
support multiple namespaces (e.g., gets and dsn), then it would respond to 
either and internally would know how to prioritize values from the two namespaces. 
Otherwise it simply processes the call setup attempt as if it didn't have a 
Resource-Priority header. (see 4.2 below).

When one defines the specific procedures for specific namespaces (not in this 
document), the only possible response to a request should be to indicate 
"accepted" priority values within the same namespace as the request. Otherwise, it 
seems to contradict the point of defining namespaces.

4.2 "Unknown namespace" error: The sentence with "it MAY serve ... if and 
only if" seems strange. The "if and only if" appears to be indicating a "MUST" 
situation. The rest of the sentence is confusing, since it is saying that it 
depends on "if the request would ... experience treatment no different than a 
non-labeled request", but the whole point of the sentence is that it's treated 
"as if it had no priority indication", that is, is non-labeled. Also, it seems 
that there should be different treatment when the namespace is recognized but 
the priority value isn't. I believe the whole sentence should be replaced with: 
"If an element receives a request with a namespace that it does not 
recognize, it SHOULD serve the request as if it had no Resource-Priority header. If an 
element receives a request with a namespace it recognizes but a priority value 
that it does not recognize, it SHOULD serve the request as if it had the 
default value specified for that namespace." (Section 12 says what is in this last 
sentence.)

The 2nd paragraph (beginning "If an unknown resource ...") continue to be a 
problem, since it seems to mandate an impossible operation: the element must 
make a decision based on the operation that should have occurred for a namespace 
that it does not recognize. It must be assumed that the phrase "or would get 
a different treatment" intends to say "than if the namespace were recognized". 
This second paragraph should be deleted.

The 3rd paragraph gives the incorrect meanings for the response code 417. 
Section 11 defines it as "Unknown Resource-Priority". (Also wrong in 4.4 and 7.2.)
 
4.4 The text here limits the UAC to sending one Resource-Priority Header 
field. However, Section 3 contained the statement that "There may be multiple 
resource values or, equivalently, multiple Resource-Priority header fields." I 
believe Section 3 is incorrect.

The description in the 2nd paragraph would not be the allowed operation in 
any case that I am aware of. For any use of this Resource-Priority header, the 
user specifies the appropriate priority value for the call being attempted. 
Therefore, no device (UAC) is allowed to reattempt a failed call at a different 
(presumably higher) level. If the call is rejected due to the attempted 
unauthorized use of a level or if the call is blocked when attempted at a particular 
level, the only option is to notify the user of this fact. If the user decides 
to reattempt the call with a higher level, the UAC can not prevent this. But 
no UAC can be allowed to automatically use a higher level.

In any case, the rules for use of this Resource-Priority header should be 
left to documents describing specific uses (namespaces).

B.1 Reference to "critic-ecp" should be added before "flash-override" in the 
list. The reference to "critic-ecp" should be deleted from the 2nd paragraph 
since it was added to the 1st. A relative order needs to be explicitly 
specified as requires by Section 12. This should specify that "Routine" is the 
default. The following is suggested for the 1st paragraph:

This document defines the namespace "dsn". The namespace "dsn" (Defense 
Switched Network) contains the following priority values in the relative order 
listed: "critic-ecp", "flash-override", "flash", immediate", "priority", "and 
"routine", with "critic-ecp as the highest and "routine" as the lowest. The 
default is "routine".

B.3 and B.4
There is a confusion here since GETS and TDR are really the same thing. These 
can not be split into separate namespaces.
Since the IANA considerations states that "the registration must indicate the 
default", it is not possible to have a namespace with only one value. It is 
necessary to define a second value which can be designated as the default even 
if that value is never used in signaling.

B.4 The current defined value is incorrect. "Authorized_emergency" is not the 
same as 911 or 112 calls placed by the general public. "Authorized emergency" 
belongs with GETS. In the previous drafts, authorized emergency and 911 were 
intended as two distinct values within one namespace that covered GETS-type 
calls, 911-type calls, and normal calls (i.e., the regular "public network").

I believe the correct approach is to define a single namespace for the 
"public network" environment with the following values (at a minimum):
- Authorized emergency (GETS, etc.)
- Public emergency (911, 112, etc.)
- Normal
Of course, "normal" is the default. Further, I know that there are some who 
believe that multiple "Authorized emergency" levels are needed, similar to what 
is defined for the Wireless Priority System mentioned in Section 3.

If this issue of the definition of multiple values to support authorized 
emergency (GETS or TDR), public emergency calling (e.g., 911), and various other 
priority schemes that might reside in the same "public" network can't be 
quickly resolved, these should be removed from this draft, to be added later by IANA 
registration.

B.4 I'm not sure why the second paragraph under B.4 for the Defense Red 
Switched Network was added. (This should have been a new section number.) If kept, 
it also should specify that "Routine" is the default. We're looking into 
whether this new "FOO" precedence level is needed and whether it should simply be 
added to the existing dsn namespace.

Reference 11 needs to be updated to read RFC 3487.

Mike Pierce
Artel


--part1_1e4.bd0cbeb.2c29ddb1_boundary
Content-Type: text/html; charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2>To all,
<BR>
<BR>Comments on draft-ietf-sip-resource-priority-00:
<BR>
<BR>The abstract says "defines a new SIP header field ... called Resource-Pr=
iority", but the draft then defines two headers. If the second header is inc=
luded (see comments below), references should be added in the abstract and t=
hroughout the Introduction.
<BR>
<BR>In the statement in the abstract that this header "does not influence th=
e forwarding behavior of IP routers", it needs to be made clear that, while=20=
there is no "direct" influence, it is certainly possible for such influence=20=
to exist, based on procedures and processes not described in this document.=20=
For example, the descriptions for "Assured Service" have always presumed tha=
t this Resource-Priority header in the session (call) setup signaling could=20=
be used to cause preferential treatment of media packet forwarding, includin=
g multiple drop precedences in an EF queue. I believe the sentence should re=
ad: "While it does not directly influence the forwarding behavior of IP rout=
ers, procedures for using this header to cause such influence may be defined=
 in other documents."
<BR>
<BR>3. The definition of the Resource-Priority header should not allow multi=
ple resource values to be carried. I am unaware of any requirement for this.=
 The use described always needs exactly one value within one namespace. Anyt=
hing beyond this will bring on questions like "Which priority has priority?"
<BR>
<BR>Note: REQ-13 says that "Some UAs will need to support multiple different=
 priority schemes". It does not say that the indication (header) needs to su=
pport a "list of names".
<BR>
<BR>3. Instead of defining a separate Accept header, it seems that there cou=
ld be a single Resource-priority Header which is then included in specific m=
essages (responses) to mean "accept". (I'm trying to find an existing exampl=
e of a header that is used like this.)
<BR>
<BR>The example of the originator requesting "q735.1, gets.enabled" and the=20=
response indicating which dsn values are supported is invalid. The dsn would=
 not function this way. If it understands the value received based on interw=
orking arrangements (e.g., gets.enabled), it may convert it to its equivalen=
t dsn value, according to the interworking rules. Or if it is set up to unde=
rstand and support multiple namespaces (e.g., gets and dsn), then it would r=
espond to either and internally would know how to prioritize values from the=
 two namespaces. Otherwise it simply processes the call setup attempt as if=20=
it didn't have a Resource-Priority header. (see 4.2 below).
<BR>
<BR>When one defines the specific procedures for specific namespaces (not in=
 this document), the only possible response to a request should be to indica=
te "accepted" priority values within the same namespace as the request. Othe=
rwise, it seems to contradict the point of defining namespaces.
<BR>
<BR>4.2 "Unknown namespace" error: The sentence with "it MAY serve ... if an=
d only if" seems strange. The "if and only if" appears to be indicating a "M=
UST" situation. The rest of the sentence is confusing, since it is saying th=
at it depends on "if the request would ... experience treatment no different=
 than a non-labeled request", but the whole point of the sentence is that it=
's treated "as if it had no priority indication", that is, is non-labeled. A=
lso, it seems that there should be different treatment when the namespace is=
 recognized but the priority value isn't. I believe the whole sentence shoul=
d be replaced with: "If an element receives a request with a namespace that=20=
it does not recognize, it SHOULD serve the request as if it had no Resource-=
Priority header. If an element receives a request with a namespace it recogn=
izes but a priority value that it does not recognize, it SHOULD serve the re=
quest as if it had the default value specified for that namespace." (Section=
 12 says what is in this last sentence.)
<BR>
<BR>The 2nd paragraph (beginning "If an unknown resource ...") continue to b=
e a problem, since it seems to mandate an impossible operation: the element=20=
must make a decision based on the operation that should have occurred for a=20=
namespace that it does not recognize. It must be assumed that the phrase "or=
 would get a different treatment" intends to say "than if the namespace were=
 recognized". This second paragraph should be deleted.
<BR>
<BR>The 3rd paragraph gives the incorrect meanings for the response code 417=
. Section 11 defines it as "Unknown Resource-Priority". (Also wrong in 4.4 a=
nd 7.2.)
<BR>=20
<BR>4.4 The text here limits the UAC to sending one Resource-Priority Header=
 field. However, Section 3 contained the statement that "There may be multip=
le resource values or, equivalently, multiple Resource-Priority header field=
s." I believe Section 3 is incorrect.
<BR>
<BR>The description in the 2nd paragraph would not be the allowed operation=20=
in any case that I am aware of. For any use of this Resource-Priority header=
, the user specifies the appropriate priority value for the call being attem=
pted. Therefore, no device (UAC) is allowed to reattempt a failed call at a=20=
different (presumably higher) level. If the call is rejected due to the atte=
mpted unauthorized use of a level or if the call is blocked when attempted a=
t a particular level, the only option is to notify the user of this fact. If=
 the user decides to reattempt the call with a higher level, the UAC can not=
 prevent this. But no UAC can be allowed to automatically use a higher level=
.
<BR>
<BR>In any case, the rules for use of this Resource-Priority header should b=
e left to documents describing specific uses (namespaces).
<BR>
<BR>B.1 Reference to "critic-ecp" should be added before "flash-override" in=
 the list. The reference to "critic-ecp" should be deleted from the 2nd para=
graph since it was added to the 1st. A relative order needs to be explicitly=
 specified as requires by Section 12. This should specify that "Routine" is=20=
the default. The following is suggested for the 1st paragraph:
<BR>
<BR>This document defines the namespace "dsn". The namespace "dsn" (Defense=20=
Switched Network) contains the following priority values in the relative ord=
er listed: "critic-ecp", "flash-override", "flash", immediate", "priority",=20=
"and "routine", with "critic-ecp as the highest and "routine" as the lowest.=
 The default is "routine".
<BR>
<BR>B.3 and B.4
<BR>There is a confusion here since GETS and TDR are really the same thing.=20=
These can not be split into separate namespaces.
<BR>Since the IANA considerations states that "the registration must indicat=
e the default", it is not possible to have a namespace with only one value.=20=
It is necessary to define a second value which can be designated as the defa=
ult even if that value is never used in signaling.
<BR>
<BR>B.4 The current defined value is incorrect. "Authorized_emergency" is no=
t the same as 911 or 112 calls placed by the general public. "Authorized eme=
rgency" belongs with GETS. In the previous drafts, authorized emergency and=20=
911 were intended as two distinct values within one namespace that covered G=
ETS-type calls, 911-type calls, and normal calls (i.e., the regular "public=20=
network").
<BR>
<BR>I believe the correct approach is to define a single namespace for the "=
public network" environment with the following values (at a minimum):
<BR>- Authorized emergency (GETS, etc.)
<BR>- Public emergency (911, 112, etc.)
<BR>- Normal
<BR>Of course, "normal" is the default. Further, I know that there are some=20=
who believe that multiple "Authorized emergency" levels are needed, similar=20=
to what is defined for the Wireless Priority System mentioned in Section 3.
<BR>
<BR>If this issue of the definition of multiple values to support authorized=
 emergency (GETS or TDR), public emergency calling (e.g., 911), and various=20=
other priority schemes that might reside in the same "public" network can't=20=
be quickly resolved, these should be removed from this draft, to be added la=
ter by IANA registration.
<BR>
<BR>B.4 I'm not sure why the second paragraph under B.4 for the Defense Red=20=
Switched Network was added. (This should have been a new section number.) If=
 kept, it also should specify that "Routine" is the default. We're looking i=
nto whether this new "FOO" precedence level is needed and whether it should=20=
simply be added to the existing dsn namespace.
<BR>
<BR>Reference 11 needs to be updated to read RFC 3487.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR></FONT></HTML>

--part1_1e4.bd0cbeb.2c29ddb1_boundary--

_______________________________________________
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 exim@www1.ietf.org  Tue Jun 24 14:45:36 2003
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From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: sip@ietf.org
Date: Tue, 24 Jun 2003 13:43:13 -0500
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Subject: [Sip] RE: I-D ACTION:draft-ietf-sip-history-info-00.txt
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Hi all,

The most significant change to this version of the document (from
draft-barnes-sipping-history-info-02) was per the discussion at IETF-56 to
make the INDEX a MUST.  I see the only major open issue to be the security
solution, for which I do have a separate draft that I've submitted to the
SIPPING WG.  Since, it wasn't submitted until Sunday afternoon, it will
probably be a couple days before it appears.  For folks that would like to
review it now, it is cached on my server at:
http://home.attbi.com/~mbarnes42/IETF/draft-barnes-sipping-sec-inserted-info
-00.txt

This draft definitely needs more work, however, I think fundamental
agreement on the problem being solved and the approach detailed up to and
including section 3 is first necessary prior to completing section 4 and
beyond. 
Until the WG has agreed whether this document belongs in the SIP WG, I would
suggest questions/comments on the security draft be sent to the SIPPING WG
mailing list. 

Regards,
Mary H. Barnes
mbarnes@nortelnetworks.com

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Tuesday, June 24, 2003 6:01 AM
Cc: sip@ietf.org
Subject: I-D ACTION:draft-ietf-sip-history-info-00.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-00.txt
	Pages		: 23
	Date		: 2003-6-23
	
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 HistInfo should be returned in responses to a request which has 
captured the history information.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-00.txt

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Date: Tue, 24 Jun 2003 14:16:07 -0500
From: Alan Johnston <alan.johnston@mci.com>
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Subject: [Sip] Centralized Conferencing (xcon) BOF Announcement
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FYI.

Centralized Conferencing BOF (xcon)

Tuesday, July 15 at 1700-1800
==============================

CHAIRS: Alan Johnston (alan.johnston@mci.com)
         Adam Roach (adam@dynamicsoft.com)

AGENDA:

- Intro and Agenda Bashing (5 min)
- Problem Statement and Overview of Scope (10 min)
- Charter Discussion (15 min)
- Open Discussion (30 min)


Description of Proposed Working Group


The focus of this working group is to develop a standardized suite of 
protocols for tightly-coupled multimedia conferences, where strong security 
and authorization requirements are integral to the solution. 
Tightly-coupled conferences have a central point of control and 
authorization so they can enforce specific media and membership 
relationships, and provide an accurate roster of participants. The media 
mixing or combining function of a tightly-coupled conference need not be 
performed centrally, however. Unlike previous attempts at standardizing 
conferencing-related activities, the scope of this effort is intentionally 
very narrow, and is intended to enable interoperability in a commercial 
environment which already has a number of implementations.


Privacy, security, and authorization mechanisms are integral to the 
solution generated by the working group. This includes allowing 
participants to be completely invisible or to be visible but participate 
anonymously with respect to some or all of the other participants. 
Authorization rules allow for participants and non-participants to have 
roles (ex: speaker, moderator, owner), and to be otherwise authorized to 
perform membership and media manipulation for or on behalf of other 
participants. In order to preserve these properties, the protocols used 
will require implementation of channel security and authentication services.


Initially this combination of protocols will be specified with respect to 
session setup with SIP, but most of the specific components would be 
applicable to conferences setup using other protocols. [None of the 
protocols defined by this group will be SIP or require SIP extensions.] The 
group will use the high-level requirements and framework already described 
by documents published by the SIPPING WG.


The deliverables for the group will be:
- A mechanism for membership and authorization control
- A mechanism to manipulate and describe media "layout" or "topology" for 
multiple media types (audio, video, text)
- A mechanism for notification of conference related events/changes (for 
example a roster)
- A basic floor control protocol
- Peer-to-peer cascading of conferences (one conference is a participant in 
another and vice versa)


The following items are specifically out-of-scope:
- Voting
- Multicast media (due to security concerns)
- Fully distributed conferences
- Loosely-coupled conferences (no central point of control)
- Far-end device control
- Protocol used between the conference controller and the mixer(s)
- Capabilities negotiation of the mixer(s)
- Master-slave cascaded conferences


The working group will coordinate closely with the SIPPING and MMUSIC 
working groups. In addition the working group will cooperate with other 
groups as needed, including SIP, AVT, and the W3C SMIL working groups.
In addition, the working group will consider a number of existing drafts (a 
non-exhaustive list is included below) as input to the working group.


Related documents in other working groups:
- draft-ietf-sipping-conferencing-requirements-00.txt
- draft-ietf-sipping-conferencing-framework-00.txt
- draft-ietf-sipping-cc-conferencing-00.txt
- draft-ietf-sipping-cc-framework-02.txt
- draft-ietf-sipping-conference-package-00.txt

Partial list of input documents:
(All documents available at http://ee.wustl.edu/~alan/xcon/ until they 
appear in the IETF archives.)

- draft-even-xcon-conference-scenarios-00.txt
- draft-koskelainen-xcon-cpcp-reqs-00.txt
- draft-even-xcon-media-policy-requirements-00.txt
- draft-koskelainen-xcon-floor-control-req-00.txt
- draft-koskelainen-xcon-xcap-cpcp-usage-00.txt
- draft-levin-xcon-cpcp-00.txt
- draft-mahy-xcon-media-policy-control-00.txt

Comments on the above drafts are welcome on the xcon mailing list:

Mailing-List:   xcon@softarmor.com
                 http://www.softarmor.com/mailman/listinfo/xcon









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From exim@www1.ietf.org  Wed Jun 25 11:35:56 2003
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Media Objects Markup Language (MOML)
	Author(s)	: T. Melanchuk, G. Sharratt
	Filename	: draft-melanchuk-sipping-moml-00.txt
	Pages		: 42
	Date		: 2003-6-24
	
The Media Objects Markup Language (MOML) is used to define media 
processing objects which execute on media servers. It defines a set 
of primitive media objects (called primitives) and provides tools to 
group primitives together and specify how they interact with each 
other. Clients use MOML to create precisely tailored media processing 
objects which may be used as parts of application interactions with 
users or conferences or to transform media flowing internal to a 
media server. IVR is an example of an application interaction with a 
user.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-melanchuk-sipping-moml-00.txt

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From exim@www1.ietf.org  Wed Jun 25 11:35:56 2003
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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		: Communications Resource Priority for the Session 
                          Initiation Protocol (SIP)
	Author(s)	: H. Schulzrinne, J. Polk
	Filename	: draft-ietf-sip-resource-priority-00.txt
	Pages		: 24
	Date		: 2003-6-24
	
This document defines a new SIP header field for communications
resource priority, called 'Resource-Priority'. This header field can
influence the behavior of SIP UAs, such as GSTN gateways, and SIP
proxies. It does not influence the forwarding behavior of IP routers.

A URL for this Internet-Draft is:
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From exim@www1.ietf.org  Wed Jun 25 11:35:57 2003
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Subject: [Sip] [Serusers] Problems with ser
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Hi all

i have some problems with ser.

I've configurated ser with mysql. When I register a user with serctl it add 
to ser database but this is the only information that i can see on ser 
database. How can i register all information about ser activities on 
database? Where can i found some log of sip activities?

I can't use my server sip with MSN. When I configure user for communication 
services MSN can't found server. How can i configure server for use with MSN?


thanks,
andrea 


_______________________________________________
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From exim@www1.ietf.org  Wed Jun 25 12:22:05 2003
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From: giAndrea <andrea@giandrea.com>
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Subject: [Sip] [Serusers] Problems with ser
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Hi all

i have some problems with ser.

I've configurated ser with mysql. When I register a user with serctl it add 
to ser database but this is the only information that i can see on ser 
database. How can i register all information about ser activities on 
database? Where can i found some log of sip activities?

I can't use my server sip with MSN. When I configure user for communication 
services MSN can't found server. How can i configure server for use with MSN?


thanks,
andrea 


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A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Media Sessions Markup Language (MSML)
	Author(s)	: T. Melanchuk, G. Sharratt
	Filename	: draft-melanchuk-sipping-msml-00.txt
	Pages		: 31
	Date		: 2003-6-24
	
The Media Sessions Markup Language (MSML) is used to control and 
invoke many different types of services on IP media servers. Clients 
can use it define how media sessions interact on a media server and 
to apply services to individual or groups of users. MSML can be used, 
for example, to control media server advanced conferencing features, 
create sidebars, and insert media processing objects into media 
streams. As well, clients can use it with other languages such as the 
Media Objects Markup Language (MOML) or VoiceXML to interact with 
individual users or with groups of conference participants.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-melanchuk-sipping-msml-00.txt

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In a message dated 6/25/2003 7:13:23 AM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


> > 
> > 3. The definition of the Resource-Priority header should not allow 
> > multiple resource values to be carried. I am unaware of any requirement 
> > for this. The use described always needs exactly one value within one 
> > namespace. Anything beyond this will bring on questions like "Which 
> > priority has priority?"
> > 
> > Note: REQ-13 says that "Some UAs will need to support multiple different 
> > priority schemes". It does not say that the indication (header) needs to 
> > support a "list of names".
> 
> The only reason that I can think of is to support the case where I'm 
> operating in a multi-namespace environment and where I want to say 
> "please use namespace X if you can, or Y if that suits you". I don't 
> know whether this is likely to occur in practice. The priority 
> resolution is easy - the entity receiving the header ignores the stuff 
> it doesn't know and picks the highest priority among those it does. 
> However, I agree that this is probably more complication than warranted. 
> The Accept-R-P header can achieve roughly the same thing, with less 
> trouble, albeit more round-trips.
> 



This seems to be a very significant conceptual difference of ideas on what 
the namespace is used for and how. I believe the "namespace" exists only to 
allow an easy-to-manage way to define different sets of priority levels for 
different cases. Definition of a priority level "high" for one case is independent 
from "high" for the other. Namespace does not exist for use by the user.

In practice, in actual use, the orginator does not select (and then maybe 
change, as you wrote) a namespace. The device is configured to use the right one, 
while the person selects the proper priority level. If there are cases in 
which one of several namespaces could be used, it is not the human who selects 
the "namespace", except indirectly by what is dialed. The human in unaware of 
this concept. For example, if a military phone is setup and authorized to handle 
MLPP priorities, an emergency worker with GETS access may use the phone, 
perform the magic procedures to get such GETS authorization, which may result in a 
Resource-Priority header being sent with namespace=gets, priority=enabled (to 
use the values currently in the draft). It does not result in sending two 
different priority values with some other entity being given the chance to decide 
which is better. In fact, the treatments may be different, with one using 
preemption and the other queuing for resources.

No, it should never happen that the user would ask "please use namespace X if 
you can, or Y if that suits you". The request is always "use namespace X with 
priority level Z and do the best you can", (with the actual user being 
unaware of the namespace). If the call is blocked, the user may do something else 
(which might not have anything to do with a namespace).

I'm mostly afraid that any attempts to describe use of multiple namespaces 
will cause significant difficulty in moving this draft forward, since there are 
lots of unexplained interactions.

Mike Pierce
Artel



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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2>In a message dated 6/25/2=
003 7:13:23 AM Eastern Standard Time, hgs@cs.columbia.edu writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">&gt;=20
<BR>&gt; 3. The definition of the Resource-Priority header should not allow=20
<BR>&gt; multiple resource values to be carried. I am unaware of any require=
ment=20
<BR>&gt; for this. The use described always needs exactly one value within o=
ne=20
<BR>&gt; namespace. Anything beyond this will bring on questions like "Which=
=20
<BR>&gt; priority has priority?"
<BR>&gt;=20
<BR>&gt; Note: REQ-13 says that "Some UAs will need to support multiple diff=
erent=20
<BR>&gt; priority schemes". It does not say that the indication (header) nee=
ds to=20
<BR>&gt; support a "list of names".
<BR>
<BR>The only reason that I can think of is to support the case where I'm=20
<BR>operating in a multi-namespace environment and where I want to say=20
<BR>"please use namespace X if you can, or Y if that suits you". I don't=20
<BR>know whether this is likely to occur in practice. The priority=20
<BR>resolution is easy - the entity receiving the header ignores the stuff=20
<BR>it doesn't know and picks the highest priority among those it does.=20
<BR>However, I agree that this is probably more complication than warranted.=
=20
<BR>The Accept-R-P header can achieve roughly the same thing, with less=20
<BR>trouble, albeit more round-trips.
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D3 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">
<BR>
<BR>
<BR>This seems to be a very significant conceptual difference of ideas on wh=
at the namespace is used for and how. I believe the "namespace" exists only=20=
to allow an easy-to-manage way to define different sets of priority levels f=
or different cases. Definition of a priority level "high" for one case is in=
dependent from "high" for the other. Namespace does not exist for use by the=
 user.
<BR>
<BR>In practice, in actual use, the orginator does not select (and then mayb=
e change, as you wrote) a namespace. The device is configured to use the rig=
ht one, while the person selects the proper priority level. If there are cas=
es in which one of several namespaces could be used, it is not the human who=
 selects the "namespace", except indirectly by what is dialed. The human in=20=
unaware of this concept. For example, if a military phone is setup and autho=
rized to handle MLPP priorities, an emergency worker with GETS access may us=
e the phone, perform the magic procedures to get such GETS authorization, wh=
ich may result in a Resource-Priority header being sent with namespace=3Dget=
s, priority=3Denabled (to use the values currently in the draft). It does no=
t result in sending two different priority values with some other entity bei=
ng given the chance to decide which is better. In fact, the treatments may b=
e different, with one using preemption and the other queuing for resources.
<BR>
<BR>No, it should never happen that the user would ask "please use namespace=
 X if you can, or Y if that suits you". The request is always "use namespace=
 X with priority level Z and do the best you can", (with the actual user bei=
ng unaware of the namespace). If the call is blocked, the user may do someth=
ing else (which might not have anything to do with a namespace).
<BR>
<BR>I'm mostly afraid that any attempts to describe use of multiple namespac=
es will cause significant difficulty in moving this draft forward, since the=
re are lots of unexplained interactions.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR>
<BR></FONT></HTML>

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In a message dated 6/24/2003 11:54:45 PM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


> > 4.2 "Unknown namespace" error: The sentence with "it MAY serve ... if 
> > and only if" seems strange. The "if and only if" appears to be 
> > indicating a "MUST" situation. The rest of the sentence is confusing, 
> > since it is saying that it depends on "if the request would ... 
> > experience treatment no different than a non-labeled request", but the 
> > whole point of the sentence is that it's treated "as if it had no 
> > priority indication", that is, is non-labeled. Also, it seems that there 
> 
> I rephrased it, but basically the intent was that if, at the time of 
> arrival of the request, labeled and non-labeled requests are treated the 
> same, there doesn't seem to be a point to say "sorry, don't know this 
> (but it wouldn't make any difference in any event)". In other words, 
> under normal, no-overload operations conditions, where all requests are 
> served, there is no need to reject requests that have the wrong label.
> 
> I don't think treating unknown namespaces the same as no R-P is right if 
> the treatment differs. Rather than being routed into some inferior 
> corner of the network if I guessed the wrong namespace, I should be told 
> that I should be using a different namespace.
> 




I'm still confused. By definition, labeled and non-labeled requests may be 
treated differently someplace in the network at any time. In any case, an entity 
could make such a distinction between labels that it understands and requests 
without labels. But it is impossible to make decisions based on a label it 
does not understand, since it would have no way of knowing whether the request 
might be treated differently at this moment if it understood the label. What 
if, at the specific moment, one label would result in different treatment while 
a second would not? Then what would be the decision for a 3rd, unknown label?

I would hate to think that deciding how to process the incoming unknown label 
would depend on whether there is currently overload, especially since that 
question itself (is there overload) may depend on what the label is.

This again relates to my comments about the concept and use of "namespace". 
We are not concerned with someone "guessing" the wrong namespace, since the 
user doesn't select it directly.

In a message dated 6/24/2003 11:54:45 PM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


> > 
> > The 2nd paragraph (beginning "If an unknown resource ...") continue to 
> > be a problem, since it seems to mandate an impossible operation: the 
> > element must make a decision based on the operation that should have 
> > occurred for a namespace that it does not recognize. It must be assumed 
> > that the phrase "or would get a different treatment" intends to say 
> > "than if the namespace were recognized". This second paragraph should be 
> > deleted.
> 
> I disagree. It simply says that if the treatment is different I ask for 
> X, but you don't know X, and if the current treatment for some 
> priorities is better than for default, I should be given a chance to 
> apply for this better treatment rather than being silently sent into the 
> long queue.
> 
> 


I don't understand the point. It seems too complicated and unecessary. As an 
alternative to what I suggested (treat the request as if it had no R-P 
header), it would be possible instead to always reject with "Unknown 
Resource-Priority", except that code does not indicate that it was the namespace that was 
unknown. Always rejecting for an unknown namespace would be okay, since this can 
only happen due to misconfiguration in the originating device.


Mike Pierce
Artel


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2>In a message dated 6/24/2=
003 11:54:45 PM Eastern Standard Time, hgs@cs.columbia.edu writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">&gt; 4.2 "Unknown namespace=
" error: The sentence with "it MAY serve ... if=20
<BR>&gt; and only if" seems strange. The "if and only if" appears to be=20
<BR>&gt; indicating a "MUST" situation. The rest of the sentence is confusin=
g,=20
<BR>&gt; since it is saying that it depends on "if the request would ...=20
<BR>&gt; experience treatment no different than a non-labeled request", but=20=
the=20
<BR>&gt; whole point of the sentence is that it's treated "as if it had no=20
<BR>&gt; priority indication", that is, is non-labeled. Also, it seems that=20=
there=20
<BR>
<BR>I rephrased it, but basically the intent was that if, at the time of=20
<BR>arrival of the request, labeled and non-labeled requests are treated the=
=20
<BR>same, there doesn't seem to be a point to say "sorry, don't know this=20
<BR>(but it wouldn't make any difference in any event)". In other words,=20
<BR>under normal, no-overload operations conditions, where all requests are=20
<BR>served, there is no need to reject requests that have the wrong label.
<BR>
<BR>I don't think treating unknown namespaces the same as no R-P is right if=
=20
<BR>the treatment differs. Rather than being routed into some inferior=20
<BR>corner of the network if I guessed the wrong namespace, I should be told=
=20
<BR>that I should be using a different namespace.
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D3 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">
<BR>
<BR>
<BR>
<BR>I'm still confused. By definition, labeled and non-labeled requests may=20=
be treated differently someplace in the network at any time. In any case, an=
 entity could make such a distinction between labels that it understands and=
 requests without labels. But it is impossible to make decisions based on a=20=
label it does not understand, since it would have no way of knowing whether=20=
the request might be treated differently at this moment if it understood the=
 label. What if, at the specific moment, one label would result in different=
 treatment while a second would not? Then what would be the decision for a 3=
rd, unknown label?
<BR>
<BR>I would hate to think that deciding how to process the incoming unknown=20=
label would depend on whether there is currently overload, especially since=20=
that question itself (is there overload) may depend on what the label is.
<BR>
<BR>This again relates to my comments about the concept and use of "namespac=
e". We are not concerned with someone "guessing" the wrong namespace, since=20=
the user doesn't select it directly.
<BR>
<BR>In a message dated 6/24/2003 11:54:45 PM Eastern Standard Time, hgs@cs.c=
olumbia.edu writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">&gt;=20
<BR>&gt; The 2nd paragraph (beginning "If an unknown resource ...") continue=
 to=20
<BR>&gt; be a problem, since it seems to mandate an impossible operation: th=
e=20
<BR>&gt; element must make a decision based on the operation that should hav=
e=20
<BR>&gt; occurred for a namespace that it does not recognize. It must be ass=
umed=20
<BR>&gt; that the phrase "or would get a different treatment" intends to say=
=20
<BR>&gt; "than if the namespace were recognized". This second paragraph shou=
ld be=20
<BR>&gt; deleted.
<BR>
<BR>I disagree. It simply says that if the treatment is different I ask for=20
<BR>X, but you don't know X, and if the current treatment for some=20
<BR>priorities is better than for default, I should be given a chance to=20
<BR>apply for this better treatment rather than being silently sent into the=
=20
<BR>long queue.
<BR>
<BR></BLOCKQUOTE>
<BR>
<BR>
<BR>I don't understand the point. It seems too complicated and unecessary. A=
s an alternative to what I suggested (treat the request as if it had no R-P=20=
header), it would be possible instead to always reject with "Unknown Resourc=
e-Priority", except that code does not indicate that it was the namespace th=
at was unknown. Always rejecting for an unknown namespace would be okay, sin=
ce this can only happen due to misconfiguration in the originating device.
<BR>
<BR>
<BR>Mike Pierce
<BR>Artel
<BR></FONT></HTML>

--part1_54.14345e76.2c2b3cad_boundary--

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Date: Thu, 26 Jun 2003 10:16:35 +0300
From: Hassan CHOUMAR <Hassan.Choumar@int-evry.fr>
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Hello, 
I installed server SER in my work, and I configured the SERWEB to use
the graphic interface but I have a problem and that during launch of
web page, I do not know which is the URL( www....) address to be put
to see the graphic interface of SERWEB.
Thank you in advance if anybody knows the answer
Hassan

_______________________________________________
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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From exim@www1.ietf.org  Thu Jun 26 04:25:18 2003
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Date: Thu, 26 Jun 2003 10:16:22 +0300
From: Hassan CHOUMAR <Hassan.Choumar@int-evry.fr>
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Subject: [Sip] Problems with  URL of serweb
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Hello,=20
I installed server SER in my work, and I configured the SERWEB to use
the graphic interface but I have a problem and that during launch of
web page, I do not know which is the URL( www....) address to be put
to see the graphic interface of SERWEB.
Thank you in advance if anybody knows the answer
Hassan
-------------------
>
>Hi all
>
>i have some problems with ser.
>
>I've configurated ser with mysql. When I register a user with serctl
it add=20
>to ser database but this is the only information that i can see on
ser=20
>database. How can i register all information about ser activities on=20
>database? Where can i found some log of sip activities?
>
>I can't use my server sip with MSN. When I configure user for
communication=20
>services MSN can't found server. How can i configure server for use
with MSN?
>
>
>thanks,
>andrea=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
>

CHOUMAR Hassan                                                   =20
cit=E9 universitaire de la pacaterie (chambre459)
91400 Orsay                                    =20
Por :33/0675909977
Email: hmchoumar@hotmail.com

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



From exim@www1.ietf.org  Thu Jun 26 07:28:41 2003
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Date: Thu, 26 Jun 2003 13:25:57 +0300
From: Hassan CHOUMAR <Hassan.Choumar@int-evry.fr>
X-Originating-IP: [157.159.15.145]
Subject: Re: [Sip] [Serusers] Problems with ser
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Hello,

When you add the users through (serctl), it is necessary to make the
following points:
- Activation of the data base mysql by commands:
> / etc. / init.d / mysqld start

- Enter the following directory:
cd / usr / sbin/
- Type the command: =20
usr/sbin> Serctl add < use > < password > < mail >

Later return on the basis of data of SER by:

> Mysql
Mysql > show databases;
Mysql > uses(wears out) ser;
Mysql > show tables;
Mysql > select * from subscribers;

Hassan=20


-------------------
>
>Hi all
>
>i have some problems with ser.
>
>I've configurated ser with mysql. When I register a user with serctl
it add=20
>to ser database but this is the only information that i can see on
ser=20
>database. How can i register all information about ser activities on=20
>database? Where can i found some log of sip activities?
>
>I can't use my server sip with MSN. When I configure user for
communication=20
>services MSN can't found server. How can i configure server for use
with MSN?
>
>
>thanks,
>andrea=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
>

CHOUMAR Hassan                                                   =20
cit=E9 universitaire de la pacaterie (chambre459)
91400 Orsay                                    =20
Por :33/0675909977
Email: hmchoumar@hotmail.com

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



From exim@www1.ietf.org  Thu Jun 26 08:04:36 2003
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To: IETF-Announce: ;
CC: sip@ietf.org, sipping@ietf.org
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Thu, 26 Jun 2003 08:03:20 -0400
Subject: [Sip] I-D ACTION:draft-mahy-sip-connect-reuse-00.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Connection Reuse in the Session Initiation Protocol 
                          (SIP)
	Author(s)	: R. Mahy
	Filename	: draft-mahy-sip-connect-reuse-00.txt
	Pages		: 12
	Date		: 2003-6-25
	
When SIP entities use a connection oriented protocol to send a
request, they typically originate their connections from an ephemeral
port. The SIP protocol includes mechanisms which insure that
responses to a request, and new requests sent in the original
direction reuse an existing connection.  However, new requests sent
in the opposite direction are unlikely to reuse the existing
connection.  This frequently causes a pair of SIP entities to use one
connection for requests sent in each direction, and can result in
potential scaling and performance problems.  This document proposes
requirements and a mechanism which address this deficiency.

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

	Title		: Indicating User Agent Capabilities in the Session 
                          Initiation Protocol (SIP)
	Author(s)	: J. Rosenberg et al.
	Filename	: draft-ietf-sip-callee-caps-00.txt
	Pages		: 40
	Date		: 2003-6-26
	
This specification defines mechanisms by which a Session Initiation
Protocol (SIP) user agent can convey its capabilities and
characteristics to other user agents. These capabilities are conveyed
as parameters of the Contact header field.

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From exim@www1.ietf.org  Thu Jun 26 14:26:26 2003
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From: "Shetty Bharathraj (IFIN DC COM)" <Bharathraj.Shetty@infineon.com>
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HI

I have some questions on implementation of SDP decoder.

 We are doing interoperable testing our SIP stack with RADVISION prolab
manager, 
the SDP test scripts available with this tools are not consistent with the
syntax of RFC 2327.
Some of them are as follows.

*	In Some of SDP test scripts, CRLF or  LF is missing at the end of
SDP script(Last field).
*	The connection field(c=) and time(t=) are not mandatory in scripts
available in prolab manager.
*	The order of the fields, email field and phone fields are not as per
the RFC2327(e-mail field should come before the phone field).

How do we go ahead with the testing of SDP with these kind of test cases?.

Regards
Bharath


Bharathraj Shetty A.N.
Software Engineer,
Infineon Technologies India Pvt.Ltd. 
10th Floor, Discoverer Block, 
International Tech Park, 
Whitefield Road, 
Bangalore - 560 066. India. 
Tel     :+91-80-8410017/18   (Extn.2039) 
Fax    :+91-80-8410012 
email : bharathraj.shetty@infineon.com  
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From exim@www1.ietf.org  Thu Jun 26 14:49:32 2003
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We have come across what may be a bug in RFC 3261.  The following bug is
closely related, but it is not clear to me it is the same.

http://bugs.sipit.net/sipwg/show_bug.cgi?id=3D706

The issue involves the Invite client transaction state machine.  As the
above bug describes, there is no timer to exit the Proceeding state.
This is normally fine, as it is an application decision when give up.

However, when the application does give up and sends a Cancel, we still
rely on the UAS to return the forced 487 to properly exit the Proceeding
state.  If the 487 (or any final response) is never returned for
whatever reason, we remain stuck in Proceeding.

I am not clear if there should be a timer specified in the spec to allow
exit from this state in this particular scenario, or if this should also
be considered an application layer problem and covered under the
existing bug.

John Hearty
Level3

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From exim@www1.ietf.org  Thu Jun 26 14:51:53 2003
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Mpierce1@aol.com wrote:


> 4.2 "Unknown namespace" error: The sentence with "it MAY serve ... if 
> and only if" seems strange. The "if and only if" appears to be 
> indicating a "MUST" situation. The rest of the sentence is confusing, 
> since it is saying that it depends on "if the request would ... 
> experience treatment no different than a non-labeled request", but the 
> whole point of the sentence is that it's treated "as if it had no 
> priority indication", that is, is non-labeled. Also, it seems that there 

I rephrased it, but basically the intent was that if, at the time of 
arrival of the request, labeled and non-labeled requests are treated the 
same, there doesn't seem to be a point to say "sorry, don't know this 
(but it wouldn't make any difference in any event)". In other words, 
under normal, no-overload operations conditions, where all requests are 
served, there is no need to reject requests that have the wrong label.

I don't think treating unknown namespaces the same as no R-P is right if 
the treatment differs. Rather than being routed into some inferior 
corner of the network if I guessed the wrong namespace, I should be told 
that I should be using a different namespace.

> 
> The 2nd paragraph (beginning "If an unknown resource ...") continue to 
> be a problem, since it seems to mandate an impossible operation: the 
> element must make a decision based on the operation that should have 
> occurred for a namespace that it does not recognize. It must be assumed 
> that the phrase "or would get a different treatment" intends to say 
> "than if the namespace were recognized". This second paragraph should be 
> deleted.

I disagree. It simply says that if the treatment is different I ask for 
X, but you don't know X, and if the current treatment for some 
priorities is better than for default, I should be given a chance to 
apply for this better treatment rather than being silently sent into the 
long queue.


> The description in the 2nd paragraph would not be the allowed operation 
> in any case that I am aware of. For any use of this Resource-Priority 
> header, the user specifies the appropriate priority value for the call 
> being attempted. Therefore, no device (UAC) is allowed to reattempt a 
> failed call at a different (presumably higher) level. If the call is 
> rejected due to the attempted unauthorized use of a level or if the call 
> is blocked when attempted at a particular level, the only option is to 
> notify the user of this fact. If the user decides to reattempt the call 
> with a higher level, the UAC can not prevent this. But no UAC can be 
> allowed to automatically use a higher level.

While I can see that 'upgrading' priority is unlikely to be useful 
(although maybe human nature...), changing namespaces will be common.

> 
> In any case, the rules for use of this Resource-Priority header should 
> be left to documents describing specific uses (namespaces).

This is just a MAY and simply describes UAC behavior.

> 
> B.1 Reference to "critic-ecp" should be added before "flash-override" in 
> the list. The reference to "critic-ecp" should be deleted from the 2nd 
> paragraph since it was added to the 1st. A relative order needs to be 
> explicitly specified as requires by Section 12. This should specify that 
> "Routine" is the default. The following is suggested for the 1st paragraph:
> 
> This document defines the namespace "dsn". The namespace "dsn" (Defense 
> Switched Network) contains the following priority values in the relative 
> order listed: "critic-ecp", "flash-override", "flash", immediate", 
> "priority", "and "routine", with "critic-ecp as the highest and 
> "routine" as the lowest. The default is "routine".

Maybe you and James can get together and figure out whether critic-ecp 
exists or not. He told me to remove it; I'm not familiar enough with the 
DSN to make the call.

> 
> B.3 and B.4
> There is a confusion here since GETS and TDR are really the same thing. 
> These can not be split into separate namespaces.

Again, please discuss with my co-author. He insisted on two.

> Since the IANA considerations states that "the registration must 
> indicate the default", it is not possible to have a namespace with only 
> one value. It is necessary to define a second value which can be 
> designated as the default even if that value is never used in signaling.

Noted.

> 
> B.4 The current defined value is incorrect. "Authorized_emergency" is 
> not the same as 911 or 112 calls placed by the general public. 
> "Authorized emergency" belongs with GETS. In the previous drafts, 
> authorized emergency and 911 were intended as two distinct values within 
> one namespace that covered GETS-type calls, 911-type calls, and normal 
> calls (i.e., the regular "public network").
> 
> I believe the correct approach is to define a single namespace for the 
> "public network" environment with the following values (at a minimum):
> - Authorized emergency (GETS, etc.)
> - Public emergency (911, 112, etc.)
> - Normal
> Of course, "normal" is the default. Further, I know that there are some 
> who believe that multiple "Authorized emergency" levels are needed, 
> similar to what is defined for the Wireless Priority System mentioned in 
> Section 3.
> 
> If this issue of the definition of multiple values to support authorized 
> emergency (GETS or TDR), public emergency calling (e.g., 911), and 
> various other priority schemes that might reside in the same "public" 
> network can't be quickly resolved, these should be removed from this 
> draft, to be added later by IANA registration.

Since this doesn't seem like a topic that is likely to be resolved soon, 
I would agree that punting on all of them seems the right thing. I don't 
want to further delay this while this is being hashed out by third parties.

> 
> B.4 I'm not sure why the second paragraph under B.4 for the Defense Red 
> Switched Network was added. (This should have been a new section 
> number.) If kept, it also should specify that "Routine" is the default. 

Section heading was omitted by accident; added.



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From exim@www1.ietf.org  Thu Jun 26 15:25:36 2003
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From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Simple question about NOTIFY
Date: Thu, 26 Jun 2003 15:27:25 -0400
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> As I am reading through the two RFCs 3261, 
> and 3265, I am just wondering if a 
> SIP NOTIFY can be forked? 

Yes a NOTIFY can be forked.

> Also if it can be forked, can someone please 
> explain how will that work, since NOTIFY 
> would be sent in a dialog, and according to 
> the rules set in section 12 of RFC 3261, 
> the Request-URI would need to be set to the 
> remote target (which is the contact address 
> of the subscriber), and a Proxy is just 
> supposed to forward to that target.

Requests can fork within a dialog during
fault-tolerant/redundancy situations where
a device's added Contact or Record-Route entry
resolves to multiple address.  ICMP errors
or retry exhaustion can trigger the target
advancing.

The NOTIFY can also fork outside of dialog
when the "subscription" was not established
through the typical SIP mechanisms.  RFC 3265 
section 3.2 discusses subscriptions created 
by not using a SUBSCRIBE.


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From exim@www1.ietf.org  Thu Jun 26 15:59:11 2003
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From: Robert Sparks <rsparks@dynamicsoft.com>
To: sip-implementors@cs.columbia.edu, sip@ietf.org, simple@ietf.org
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Date: 26 Jun 2003 14:53:33 -0500
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Subject: [Sip] SIPIT 13 Registration
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Registration for SIPIT 13 is open.

The event will be hosted by Mitel in the brand-new
Brookstreet Hotel and Conference Center in Ottawa
August 18-24th.

Information about the event and facilities as well
as online registration can be found at:
http://www.mitel.com/sipit/index.cfm

Registration through the website will close on July 18th.

See you there,
RjS



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From exim@www1.ietf.org  Thu Jun 26 16:52:35 2003
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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Simple question about NOTIFY
Date: Thu, 26 Jun 2003 16:56:28 -0400
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> If a NOTIFY can fork, then it should be 
> possible to fork a re-INVITE also. 
> Should not it? 

Yes most requests can fork within a dialog.

> Assuming the same reasoning as you provided 
> for NOTIFY in a dialog. In which case the 
> following text in RFC 3261 is not really true. 
>
> ****
> Section 14.1, Pg 87 in RFC 3261.
> 
> Unlike an INVITE, which can fork, a re-INVITE 
> will never fork, and
> therefore, only ever generate a single final 
> response. The reason a
> re-INVITE will never fork is that the 
> Request-URI identifies the
> target as the UA instance it established the 
> dialog with, rather than
> identifying an address-of-record for the user.
> ****

I agree; the text does not appear to be true.

> Regards
> - Ajay
> 
> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com] 
> Sent: Thursday, June 26, 2003 2:27 PM
> To: sip@ietf.org
> Subject: RE: [Sip] Simple question about NOTIFY
> 
> > As I am reading through the two RFCs 3261, 
> > and 3265, I am just wondering if a 
> > SIP NOTIFY can be forked? 
> 
> Yes a NOTIFY can be forked.
> 
> > Also if it can be forked, can someone please 
> > explain how will that work, since NOTIFY 
> > would be sent in a dialog, and according to 
> > the rules set in section 12 of RFC 3261, 
> > the Request-URI would need to be set to the 
> > remote target (which is the contact address 
> > of the subscriber), and a Proxy is just 
> > supposed to forward to that target.
> 
> Requests can fork within a dialog during
> fault-tolerant/redundancy situations where
> a device's added Contact or Record-Route entry
> resolves to multiple address.  ICMP errors
> or retry exhaustion can trigger the target
> advancing.
> 
> The NOTIFY can also fork outside of dialog
> when the "subscription" was not established
> through the typical SIP mechanisms.  RFC 3265 
> section 3.2 discusses subscriptions created 
> by not using a SUBSCRIBE.
> 
> 
> _______________________________________________
> 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 exim@www1.ietf.org  Fri Jun 27 18:31:27 2003
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Subject: RE: [Sip] Stuck in Proceeding after Cancel
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Thread-Topic: [Sip] Stuck in Proceeding after Cancel
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From: "Hearty, John" <John.Hearty@Level3.com>
To: <sip@ietf.org>, "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
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Jonathan,

 I found the following text that covers this scenario, so it may not be
considered a bug.  I don't particularly like the idea of an unnamed
timer though, and the below text could be a bit more explicit about when
to start the unnamed timer (upon sending Cancel).  I would appreciate
this scenario being added to the existing bugzilla bug so it is also
included in some future clarification of the Invite client transaction
section.

   Note that both the transaction corresponding to the original request
   and the CANCEL transaction will complete independently.  However, a
   UAC canceling a request cannot rely on receiving a 487 (Request
   Terminated) response for the original request, as an RFC 2543-
   compliant UAS will not generate such a response.  If there is no
   final response for the original request in 64*T1 seconds (T1 is




Rosenberg, et. al.          Standards Track                    [Page 54]
=20

RFC 3261            SIP: Session Initiation Protocol           June 2002


   defined in Section 17.1.1.1), the client SHOULD then consider the
   original transaction cancelled and SHOULD destroy the client
   transaction handling the original request.


John Hearty
Level3

> -----Original Message-----
> From: Hearty, John
> Sent: Thursday, June 26, 2003 12:47 PM
> To: sip@ietf.org
> Subject: [Sip] Stuck in Proceeding after Cancel
>=20
> We have come across what may be a bug in RFC 3261.  The following bug
is
> closely related, but it is not clear to me it is the same.
>=20
> http://bugs.sipit.net/sipwg/show_bug.cgi?id=3D706
>=20
> The issue involves the Invite client transaction state machine.  As
the
> above bug describes, there is no timer to exit the Proceeding state.
> This is normally fine, as it is an application decision when give up.
>=20
> However, when the application does give up and sends a Cancel, we
still
> rely on the UAS to return the forced 487 to properly exit the
Proceeding
> state.  If the 487 (or any final response) is never returned for
> whatever reason, we remain stuck in Proceeding.
>=20
> I am not clear if there should be a timer specified in the spec to
allow
> exit from this state in this particular scenario, or if this should
also
> be considered an application layer problem and covered under the
> existing bug.
>=20
> John Hearty
> Level3
>=20
> _______________________________________________
> 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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Mpierce1@aol.com wrote:


> one case is independent from "high" for the other. Namespace does not 
> exist for use by the user.

Why not?

> 
> In practice, in actual use, the orginator does not select (and then 
> maybe change, as you wrote) a namespace. The device is configured to use 
> the right one, while the person selects the proper priority level. If 

While this may be true in current devices, I see no reason that this has 
to be true in the future. If I have a generic SIP device, why shouldn't 
this be usable in, say, a Q.735 network and a GETS-enabled network or, 
most likely, a network where I can reach both?


> there are cases in which one of several namespaces could be used, it is 
> not the human who selects the "namespace", except indirectly by what is 
> dialed. The human in unaware of this concept. For example, if a military 
> phone is setup and authorized to handle MLPP priorities, an emergency 
> worker with GETS access may use the phone, perform the magic procedures 
> to get such GETS authorization, which may result in a Resource-Priority 
> header being sent with namespace=gets, priority=enabled (to use the 
> values currently in the draft). It does not result in sending two 
> different priority values with some other entity being given the chance 
> to decide which is better. In fact, the treatments may be different, 
> with one using preemption and the other queuing for resources.

I'm not arguing that there should be two simultaneously, in the same 
request. That was removed a few email iterations ago...


> 
> No, it should never happen that the user would ask "please use namespace 
> X if you can, or Y if that suits you". The request is always "use 
> namespace X with priority level Z and do the best you can", (with the 
> actual user being unaware of the namespace). If the call is blocked, the 
> user may do something else (which might not have anything to do with a 
> namespace).


> 
> I'm mostly afraid that any attempts to describe use of multiple 
> namespaces will cause significant difficulty in moving this draft 
> forward, since there are lots of unexplained interactions.

Namespaces are only used sequentially, not simultaneously. If you reach 
a device that understands namespace X, but you supply Y, it helps if the 
device tells you that you should be using X. After all, you might well 
have authorization for both or just forgotten to reconfigure the phone 
from one mode to another. I fail to see the problem.

> 
> Mike Pierce
> Artel
> 


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From exim@www1.ietf.org  Fri Jun 27 18:50:34 2003
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From: Idnani Ajaykumar-AIDNANI1 <Ajaykumar.Idnani@motorola.com>
To: "'brett@broadsoft.com'" <brett@broadsoft.com>, sip@ietf.org
Subject: RE: [Sip] Simple question about NOTIFY
Date: Thu, 26 Jun 2003 14:59:47 -0500
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If a NOTIFY can fork, then it should be possible to fork a re-INVITE also. Should not it? Assuming the same reasoning as you provided for NOTIFY in a dialog. In which case the following text in RFC 3261 is not really true. 

****
Section 14.1, Pg 87 in RFC 3261.

Unlike an INVITE, which can fork, a re-INVITE will never fork, and
therefore, only ever generate a single final response. The reason a
re-INVITE will never fork is that the Request-URI identifies the
target as the UA instance it established the dialog with, rather than
identifying an address-of-record for the user.
****

Regards
- Ajay

-----Original Message-----
From: Brett Tate [mailto:brett@broadsoft.com] 
Sent: Thursday, June 26, 2003 2:27 PM
To: sip@ietf.org
Subject: RE: [Sip] Simple question about NOTIFY

> As I am reading through the two RFCs 3261, 
> and 3265, I am just wondering if a 
> SIP NOTIFY can be forked? 

Yes a NOTIFY can be forked.

> Also if it can be forked, can someone please 
> explain how will that work, since NOTIFY 
> would be sent in a dialog, and according to 
> the rules set in section 12 of RFC 3261, 
> the Request-URI would need to be set to the 
> remote target (which is the contact address 
> of the subscriber), and a Proxy is just 
> supposed to forward to that target.

Requests can fork within a dialog during
fault-tolerant/redundancy situations where
a device's added Contact or Record-Route entry
resolves to multiple address.  ICMP errors
or retry exhaustion can trigger the target
advancing.

The NOTIFY can also fork outside of dialog
when the "subscription" was not established
through the typical SIP mechanisms.  RFC 3265 
section 3.2 discusses subscriptions created 
by not using a SUBSCRIBE.


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From exim@www1.ietf.org  Fri Jun 27 19:32:04 2003
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Subject: Re: [Sip] Problems with  URL of serweb
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To: Hassan CHOUMAR <Hassan.Choumar@int-evry.fr>
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Hi,

Please direct your questions directly to iptel.org.  This mailing list=20=

is for discussion of development of the SIP protocol, not usage of a=20
specific implementation.

thank you.
-rohan mahy
co-chair SIP WG




On Thursday, June 26, 2003, at 12:16 AM, Hassan CHOUMAR wrote:

> Hello,
> I installed server SER in my work, and I configured the SERWEB to use
> the graphic interface but I have a problem and that during launch of
> web page, I do not know which is the URL( www....) address to be put
> to see the graphic interface of SERWEB.
> Thank you in advance if anybody knows the answer
> Hassan
> -------------------
>>
>> Hi all
>>
>> i have some problems with ser.
>>
>> I've configurated ser with mysql. When I register a user with serctl
> it add
>> to ser database but this is the only information that i can see on
> ser
>> database. How can i register all information about ser activities on
>> database? Where can i found some log of sip activities?
>>
>> I can't use my server sip with MSN. When I configure user for
> communication
>> services MSN can't found server. How can i configure server for use
> with MSN?
>>
>>
>> thanks,
>> andrea
>>
>>
>> _______________________________________________
>> 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
>>
>
> CHOUMAR Hassan
> cit=E9 universitaire de la pacaterie (chambre459)
> 91400 Orsay
> Por :33/0675909977
> Email: hmchoumar@hotmail.com
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From exim@www1.ietf.org  Fri Jun 27 19:48:36 2003
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Hi,

Can any body tell me answer to the following question.

When a user who has ALIAS, sending INVITE can include his alias number?. 

Thanks in advance,
-Anil

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From exim@www1.ietf.org  Fri Jun 27 21:52:51 2003
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Date: Fri, 27 Jun 2003 21:45:06 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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Subject: [Sip] Re: Resource Priority header - unknown namespace
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Mpierce1@aol.com wrote:


> I'm still confused. By definition, labeled and non-labeled requests may 
> be treated differently someplace in the network at any time. In any 

If there's no overload and no PSTN labeling (e.g., simple gateway 
priority), during normal operation, all requests will effectively be 
treated the same.

I've updated the draft with a more structured presentation of the error 
cases. I'm not sure all three error modes (lenient, semi-strict, strict) 
are necessary, but I think they span the gamut of what's plausible. We 
can either
- enumerate all three (as in this revision)
- pick one
- or remain silent on the issue and leave it up to the implementation to 
do what makes sense.

Henning




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From exim@www1.ietf.org  Mon Jun 30 11:16:55 2003
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Subject: RE: [Sip] Stuck in Proceeding after Cancel
Date: Mon, 30 Jun 2003 09:13:55 -0600
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Thread-Topic: [Sip] Stuck in Proceeding after Cancel
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From: "Hearty, John" <John.Hearty@Level3.com>
To: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>,
        <sip@ietf.org>, "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
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From the duration, it sounds like timer B, though I didn't want to
presume anything.  However, the text is not clear when said timer gets
set.  Also, if it actually is timer B which is set when the original
request is sent, it could be stopped or ignored (exact action regarding
the timer is not specified) once a provisional response is received.
The below text appears to suggest said timer continues to run in this
specific scenario.

This just seems like another of the countless clarifications that the
evolving specification may need.  If I had to make the clarification, it
would be to reset timer B for the Invite Client Transaction upon sending
a Cancel for that Invite.  That would fix it, and provide an equal time
for a 487 response that a provisional response or other non-2xx final
response had.  I think 487 is a special case compared to other non-2xx
final responses.

John Hearty
Level3

> -----Original Message-----
> From: Christer Holmberg (JO/LMF)
[mailto:christer.holmberg@ericsson.com]
> Sent: Sunday, June 29, 2003 11:55 PM
> To: Hearty, John; sip@ietf.org; Jonathan Rosenberg
> Subject: RE: [Sip] Stuck in Proceeding after Cancel
>=20
>=20
> Hi,
>=20
> Isn't the timer you are talking about timer B? I agree that it would
be
> more clear to write "timer B" instead of "64*T1 seconds".
>=20
> I also think it is important to remember that timer B may expire AFTER
> CANCEL has been sent, but BEFORE 487 has been received, and since the
> transaction will be terminated there is no guarantee an ACK will be
sent
> when 487 eventually arrives. But, I guess this is similar to any case
> where the final response is sent too late by the UAS...?
>=20
> Regards,
>=20
> Christer Holmberg
> Ericsson Finland
>=20
>=20
>=20
> >  I found the following text that covers this scenario, so it
> > may not be
> > considered a bug.  I don't particularly like the idea of an unnamed
> > timer though, and the below text could be a bit more explicit
> > about when
> > to start the unnamed timer (upon sending Cancel).  I would
appreciate
> > this scenario being added to the existing bugzilla bug so it is also
> > included in some future clarification of the Invite client
transaction
> > section.
> >
> >    Note that both the transaction corresponding to the
> > original request
> >    and the CANCEL transaction will complete independently.  However,
a
> >    UAC canceling a request cannot rely on receiving a 487 (Request
> >    Terminated) response for the original request, as an RFC 2543-
> >    compliant UAS will not generate such a response.  If there is no
> >    final response for the original request in 64*T1 seconds (T1 is
> >
> >
> >
> >
> > Rosenberg, et. al.          Standards Track
> >  [Page 54]
> >
> >
> > RFC 3261            SIP: Session Initiation Protocol
> >  June 2002
> >
> >
> >    defined in Section 17.1.1.1), the client SHOULD then consider the
> >    original transaction cancelled and SHOULD destroy the client
> >    transaction handling the original request.
> >
> >
> > John Hearty
> > Level3
> >
> > > -----Original Message-----
> > > From: Hearty, John
> > > Sent: Thursday, June 26, 2003 12:47 PM
> > > To: sip@ietf.org
> > > Subject: [Sip] Stuck in Proceeding after Cancel
> > >
> > > We have come across what may be a bug in RFC 3261.  The
> > following bug
> > is
> > > closely related, but it is not clear to me it is the same.
> > >
> > > http://bugs.sipit.net/sipwg/show_bug.cgi?id=3D706
> > >
> > > The issue involves the Invite client transaction state machine.
As
> > the
> > > above bug describes, there is no timer to exit the Proceeding
state.
> > > This is normally fine, as it is an application decision
> > when give up.
> > >
> > > However, when the application does give up and sends a Cancel, we
> > still
> > > rely on the UAS to return the forced 487 to properly exit the
> > Proceeding
> > > state.  If the 487 (or any final response) is never returned for
> > > whatever reason, we remain stuck in Proceeding.
> > >
> > > I am not clear if there should be a timer specified in the spec to
> > allow
> > > exit from this state in this particular scenario, or if this
should
> > also
> > > be considered an application layer problem and covered under the
> > > existing bug.
> > >
> > > John Hearty
> > > Level3
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of
sip
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >

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



From exim@www1.ietf.org  Mon Jun 30 12:22:40 2003
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Subject: [Sip] Re: IESG Discuss Comments on Symmetric Response Routing
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My apologies for the delay. I managed to get in a revision to the 
draft before the deadline which addresses these comments. Until it 
appears in the archives, you can find it at:

http://www.jdrosen.net/papers/draft-ietf-sip-symmetric-response-01.txt


I also converted to xml while I was at it, so the general formatting 
should be better. Responses below.

Allison Mankin wrote:

> The IESG reviewed draft-ietf-sip-symmetric-response-00.txt and there were
> two sets of Discuss comments.  They do not seem too hard to address.  In
> addition, the FQDNs in the draft need to be changed to example.com.

OK, fixed. I also changed public addresses to be within 192.0.2/24. 
The private addresses remain with 10/8.

> 
> Below are the comments.  They are also in the ID tracker.
> 
> Allison
> 
> Comments:
> 
> Ted Hardie:
> First, let me say that a large number of the issues I had
> with the document turned out to be very well documented
> in the IAB considerations section of the draft. A few
> other issues:
> 
> In Section 3:
> 
>       When the client sends the request, if the request is sent using UDP,
>         the client MUST be prepared to receive the response on the same
>         socket the request was sent on. Specifically, it MUST be prepared to
>         receive the response on the same IP address and port present in the
>         source IP address and source port of the request. For backwards
>         compatibility, the client MUST still be prepared to receive a
>         response on the port indicated in the sent-by field of the topmost
>         Via header field value, as specified in Section 18.1.1 of SIP [1].
> 
> 
> It seems pretty clear from the "on the same socket" that the following
> sentence means the IP address and port present in the source IP
> and port of the request _when it is sent_. I think it would be clearer,
> though, to use phrasing like "it MUST be prepared to receive the
> response on the same IP address and port it used to populate
> the source IP address and source port of the request.".

OK, I changed the wording to the text you suggest. I made this change 
everywhere the term "socket" was used, in order to be explicit about 
what is meant.

> 
> Also in Section 3:
> 
>         To keep the binding fresh, the client SHOULD retransmit its INVITE every 20
>         seconds or so. These retransmissions will need to take place even
>         after receiving a provisional response.
> 
> based on the idea the one minute seems to be a common UDP binding lifetime.
> Section 9.3 notes, however, that there is no way to determine the UDP binding
> lifetime. Is there anyway to introduce a growing/shrinking algorithm
> to this for cases where the binding lifetime is much longer (to avoid the
> aggressiveness of 20 seconds for an arbitrary period of time) or to handle
> the cases where the binding lifetime is shorter than 20+transmission time to
> the NAT (since this has to handle the NAT being at some arbitrary place in
> the topology).

Yes. STUN describes a binary search algorithm for trying to detect the 
lifetime. That is not without peril as well, and the issues there are 
documented in STUN. I added the following text:

<t>
A UA MAY execute the binding lifetime discovery algorithm in Section
10.2 of <xref target="RFC 3489">RFC 3489</xref> to determine the
actual binding lifetime in the NAT. If it is longer than 1
minute, the client SHOULD increase the interval for request
retransmissions up to half of the discovered lifetime. If it is
shorter than one minute, it SHOULD decrease the interval for request
retransmissions to half of the discovered lifetime. Note that
discovery of binding lifetimes can be unreliable. See Section 14.3 of
<xref target="RFC3489">RFC 3489</xref>.
</t>


> 
> In the initial section of 9:
> 
>         The client can then perform an additional registration,
>         using this address in a Contact header. This would allow a client to
>         receive incoming requests, such as INVITE, on the socket through
>         which the registration was sent.
> 
> As is noted below that point, there are cases in which the port binding
> is only valid for the server to which the original registration was sent.
> Will the Contact header "leak" that so that a direction connection between a
> different sip agent (user or proxy) might attempt to use it? If so, a forward
> pointer to the limitation might be useful here.

This is possible with the Path extension to SIP (RFC 3327). That spec 
allows for servers between the client and the registrar to indicate 
that they need to be on the path of requests from the registrar to the 
client. Its a good idea to include a reference to that and a brief 
discussion.

Here is what I added:

<t>
In many cases, the server to whom the registration is sent won't be
the registrar itself, but rather, a proxy which then sends the request
to the registrar. In such a case, any incoming requests for the client
must traverse the proxy to whom the registration was directly
sent. The <xref target="RFC3327">Path header extension to SIP</xref>
allows the proxy to indicate that it must be on the path of such
requests.
</t>

and this introduced some additional brittleness that I documented:

<t>If the registrar and the server to whom the client sent its
REGISTER request are not the same, the approach will only work if the
server uses the <xref target="RFC3327">Path header field</xref>. There
is not an easy and reliable way for the server to determine that the
Path header should be used for a registration. Using Path when the
address in the topmost Via header field is a private address will
usually work, but may result in usage of Path when it is not actually
needed.
</t>

> 
> I would personally consider the UNSAF document a normative reference,
> but this is not a big deal.

RFC 3424 is informational. I dont think you can have a normative 
reference to an informational document. As such, I have kept the UNSAF 
reference as informational.

> 
> Erik Nordmark:
>  The security considerations section doesn't set a very good example. 
>  It could easily be read as "we didn't look hard for security issues".
> 
>  Thus I'd like it to provide the reasoning that lead to thinking
>  that there are no added security issues. Presumably a paragraph or two would
>  suffice.

OK. Here is the text I put in there:

<t>When a server uses this specification, responses that it sends will
now include the source port where the request came from. In some
instances, the source address and port of a request are sensitive
information. If they are sensitive, requests SHOULD be protected by
using SIP over TLS <xref target="RFC3261"/>. In such a case, this
specification does not provide any response routing functions (as
these only work with TCP); it merely provides the client with
information about the source port as seen by the server.
</t>

<t>
It is possible that an attacker might try to disrupt service to a
client by acting as a man-in-the-middle, modifying the rport parameter
in a Via header in a request sent by a client. Removal of this
parameter will prevent clients from behind NATs from receiving
service. Addition of the parameter will generally have no
impact. Of course, if an attacker is capable of launching a
man-in-the-middle attack, there are many other ways of denying
service, such as merely discarding the request. Therefore, this attack
does not seem significant.
</t>

Thanks,
Jonathan R.

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


_______________________________________________
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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 exim@www1.ietf.org  Mon Jun 30 18:25:38 2003
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: <rohan@cisco.com>, <jon.peterson@neustar.biz>, <mankin@psg.com>
Date: Mon, 30 Jun 2003 17:23:16 -0500
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Subject: [Sip] Jon Peterson exiting Chair role in SIP WG
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You've probably noticed that Jon Peterson was recently selected to serve =
on
the IESG as an Area Director for the Transport Area. Of course, the SIP
Working Group is in the Transport Area.

As it is slightly unusual for someone to be their own AD on an ongoing
basis, Jon is retiring from his role as a co-chair of the SIP working =
group,
effective immediately. Rohan and I will continue as co-chairs. I believe
Allison Mankin will be continuing as the primary AD for SIP, but we can
expect that Jon will be very able to assist in the IESG review and
supervision of our work.

Jon has been a big help to the SIP community during his time as a =
co-chair
of our working group, and I think we all owe him a big "Thank You!" for =
his
efforts. Of course, we'll probably owe him one even more for the hard =
work
he'll be putting in on the IESG. As I understand it, that's not exactly =
an
easy job.

So, I hope my ASCII art skills hold up . . .


TTTTTTTT  H    H      A      N   N  K   K  SSSSS =20
   T      H    H     A A     NN  N  K  K   S     =20
   T      HHHHHH    AAAAA    N N N  KKK    SSSSS =20
   T      H    H   A     A   N  NN  K  K       S =20
   T      H    H  A       A  N   N  K   K  SSSSS =20

                   J   OOO   N   N
                   J  O   O  NN  N
                   J  O   O  N N N
                J  J  O   O  N  NN
                JJJJ   OOO   N   N


--
Dean



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Date: Mon, 30 Jun 2003 19:33:00 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sip] Jon Peterson exiting Chair role in SIP WG
In-reply-to: <000301c33f56$346ad5e0$ee036e3f@txdwillis>
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Thanks Jon!  

And please contribute to the IETF getting its act finally together and
making SIMPLE _THE_ presence and IM standard. It is much overdue.

Henry

Henry Sinnreich
MCI
400 International Parkway
Richardson, Texas 75081
USA

Have you disconnected the PBX?
 

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Monday, June 30, 2003 5:23 PM
> To: sip@ietf.org
> Cc: rohan@cisco.com; jon.peterson@neustar.biz; mankin@psg.com
> Subject: [Sip] Jon Peterson exiting Chair role in SIP WG
> 
> 
> 
> You've probably noticed that Jon Peterson was recently 
> selected to serve on the IESG as an Area Director for the 
> Transport Area. Of course, the SIP Working Group is in the 
> Transport Area.
> 
> As it is slightly unusual for someone to be their own AD on 
> an ongoing basis, Jon is retiring from his role as a co-chair 
> of the SIP working group, effective immediately. Rohan and I 
> will continue as co-chairs. I believe Allison Mankin will be 
> continuing as the primary AD for SIP, but we can expect that 
> Jon will be very able to assist in the IESG review and 
> supervision of our work.
> 
> Jon has been a big help to the SIP community during his time 
> as a co-chair of our working group, and I think we all owe 
> him a big "Thank You!" for his efforts. Of course, we'll 
> probably owe him one even more for the hard work he'll be 
> putting in on the IESG. As I understand it, that's not 
> exactly an easy job.
> 
> So, I hope my ASCII art skills hold up . . .
> 
> 
> TTTTTTTT  H    H      A      N   N  K   K  SSSSS  
>    T      H    H     A A     NN  N  K  K   S      
>    T      HHHHHH    AAAAA    N N N  KKK    SSSSS  
>    T      H    H   A     A   N  NN  K  K       S  
>    T      H    H  A       A  N   N  K   K  SSSSS  
> 
>                    J   OOO   N   N
>                    J  O   O  NN  N
>                    J  O   O  N N N
>                 J  J  O   O  N  NN
>                 JJJJ   OOO   N   N
> 
> 
> --
> Dean
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 


_______________________________________________
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



