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To: wenkai <wkai@huawei.com>
CC: Christer Holmberg <christer.holmberg@lmf.ericsson.se>, sip@ietf.org
Subject: Re: [Sip] can i change my own dialog info  in mid-dialog?
References: <00b001c30a2b$6a1db870$af064d0a@w07087a> <3EA78A6F.1C67E563@lmf.ericsson.se> <000c01c30a34$90589f10$af064d0a@w07087a> <3EA9F280.8010800@dynamicsoft.com> <000701c30d71$f4ac3d60$af064d0a@w07087a>
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wenkai wrote:
> I would like make my original question more clear by dividing it into two
> parts.
> 
> The first part:
> "UA can change his local uri and remote uri by himself in mid-dialog.", i
> think it is correct in RFC3261.

To be clear; rfc3261 forbids this because of compatibility with rfc2543.

> 
> The second part:
> "UA can change his request-uri by himself in mid-dialog."  Maybe we can not
> do it.
> RFC3261 said,  in mid-dialog, "The UAC uses the remote target and route set
> to build the Request-URI".  UA can not change "remote target" and "route
> set" by himself in mid-dialog.  So, UA can not change his request-uri by
> himself in mid-dialog. The UA can change his request-uri by opposite UA's
> instruction.
> 
> I ask this question because request-uri can be used as service
> identification. If UA can change the request-uri by itself, so UA can use
> request-uri to carry different service request during mid-dialog. Maybe we
> can use parameters to achieve this target.

I have seen this proposed in the past as one of sevearl possible 
approaches for using SIP as an application control protocol. INFO is the 
other favorite. I still believe that this is generally outside the scope 
of SIP. If you want to control an application to which you have 
initiated a session, you should open a "control session" which is e2e. 
This control session, like the audio session, is signaling with the SDP, 
and based on some non-SIP protocol.

I actually really like MSRP (draft-campbell-simple-im-sessions-01) as a 
means for conveying e2e control messages. I think many folks still like 
INFO, but perhaps this is mostly because of the lack of a viable 
alternative.

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
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Subject: Re: [Sip] Complete list of transport protocols upon which SIP can
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TCP, UDP and SCTP. It can also use TLS over TCP.

-Jonathan R.

Valerie Kenneally wrote:
> Hi All,
> 
> 	Just wondering if anyone knows where I might find a complete list of the
> transport protocols upon which SIP can run?
> 
> Cheers,
> Val
> 
> 
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> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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From: =?Windows-1252?Q?Manuel_Moreno_Mart=EDn?= <mmoreno@cipres.upm.es>
To: "Sip" <sip@ietf.org>
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Subject: [Sip] over SIP timers
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Please:

Is correct the denomination "retransmit timer" for the timers T2,A,D,E,G,I,J
and K ???

Thank you

Manuel

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From mailnull@www1.ietf.org  Thu May  1 06:24:07 2003
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Subject: [Sip] over SIP timers
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Please:

With more precission ...

Are the timers T2, A, D, E, G, I, J and K "retransmission timers"???

Are the timers B, C, F and H "timeout timers"??

Which denomination to use for the timers T1 and T4??

Thank you 

Manuel
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To: "'Roland Gloeckler'" <gloeckle@ikr.uni-stuttgart.de>
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Subject: RE: [Sip] Motivation for RFC 3325
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I believe the two main drivers in its development were the IP Cable people and 3GPP. 

It is currently specified for use to provide calling user and connected user identity in 3GPP IP multimedia core network subsystem specifications, which use SIP (3GPP TS 24.229).

It is also specified for use in SS7 to ISUP/BICC interworking specifications in ITU-T.

The key constraint on its use is that it requires a trust domain to exist, and that is why it is not generally applicable in all SIP environments.

Keith

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

> -----Original Message-----
> From: Roland Gloeckler [mailto:gloeckle@ikr.uni-stuttgart.de]
> Sent: 30 April 2003 11:16
> To: mwatson@nortelnetworks.com; Jon.Peterson@NeuStar.biz;
> fluffy@cisco.com
> Cc: sip@ietf.org
> Subject: [Sip] Motivation for RFC 3325
> 
> 
> Dear Sirs,
> 
> I would like to know which Scenarios you had in mind for the 
> usage of RFC 3325. 
> 
> In chapter 1 you state the following:
> "An example deployment would be a closed network which 
> emulates a traditional circuit switched telephone network."
> 
> Is this the Scenario that motivated you or are the others?
> 
> Regards
> 
> Roland Glöckler
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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: "Shane McCormack" <smccormack@tssg.org>
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Date: Thu, 1 May 2003 10:36:00 +0100
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Subject: [Sip] SIP INVITE request mapping to call object
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Hi all,

I am currently working on my thesis which involves me creating a simplified
generic java API layer above SIP and parlay for easier service development.
This involves the creation of call objects similar to Java Call Control, and
Parlay.

When mapping an incoming INVITE (or incoming requests such as a NOTIFY for
presence for other similar objects), some mapping has to be performed. My
question is would you associate a call object with a request callid or
dialog? or both. So that when some routing operation is performed by the
client application using my API the correct incoming INVITE is referenced in
the underlying SIP stack when the call object is passed back to be routed or
proxied.

I am actually using SIP servlets for this implementation so SIP stack is not
correct term to use.

Regards

Shane McCormack

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

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



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Subject: [Sip] Question about resource management
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Dear all,

I have a question regarding resource management and in particular related to
at which point in the session establishment may resource reservation be
initiated. I would appreciate any comments on this.

I have noticed that the examples given in RFC 3312 seem to indicate that
resource reservation may be initiated as soon as an answer has been sent (by
the answerer) or received (by the offerer). At some point the RFC says:

"If the resource reservation mechanism used requires participation of
   both user agents, the answerer SHOULD start resource reservation
   after having sent the answer and the offerer SHOULD start resource
   reservation as soon as the answer is received.  If participation of
   the peer user agent is not needed (e.g., segmented status type), the
   offerer MAY start resource reservation before sending the offer and
   the answerer MAY start it before sending the answer."

My question is: if the answer (lets assume that it is carried in a 183
response) is carrying a list of codecs taken from the initial offer (my
understanding is that this is possible under the answer/offer model),
shouldn't both user agents wait for the updated (final) answer to be sent
and received (in the PRACK or the 200 OK to the PRACK) before initiating
resource reservation? 

Following on that, I have also noticed that 3GPP call flows as specified in
TS 24.228 show resource reservation happening after the 200 OK for the PRACK
is sent and received. Couldn't the reservation start just after the PRACK
message (carrying the final answer) is sent/received?

So, when can we assume that resource reservation can start? Can we assume
that it can start as soon as a final offer is sent/received?


Thank you very much in advance.
Regards
Maria 



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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: krisztian.kiss@nokia.com
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Subject: RE: [Sip] Question about resource management
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Hi Maria, 

a quick clarification for the 3GPP call flows:
according to the very latest changes to the last version of the call flows document (3GPP TS 24.228 5.4.0): 
- the offerer can start the resource reservation procedure right after receiving the answer in 183, if the answer is satisfactory, i.e. it contains single codecs per media flows. In case there is a need to include a second offer in PRACK in order to reduce to the "single codec per media" case, the resource reservation can be started right after sending the PRACK.
- the answerer can start the resource reservation procedure right after having sent the answer in 183, if the answer contains single codec per media flows. In case the first answer has multiple codecs per media, the answerer should wait with resource reservation until generating the second answer.

Regards,
Krisztian


> -----Original Message-----
> From: ext maria.a.cuevas@bt.com [mailto:maria.a.cuevas@bt.com]
> Sent: Friday, May 02, 2003 4:35 PM
> To: sip@ietf.org
> Subject: [Sip] Question about resource management
> 
> 
> Dear all,
> 
> I have a question regarding resource management and in 
> particular related to
> at which point in the session establishment may resource 
> reservation be
> initiated. I would appreciate any comments on this.
> 
> I have noticed that the examples given in RFC 3312 seem to 
> indicate that
> resource reservation may be initiated as soon as an answer 
> has been sent (by
> the answerer) or received (by the offerer). At some point the 
> RFC says:
> 
> "If the resource reservation mechanism used requires participation of
>    both user agents, the answerer SHOULD start resource reservation
>    after having sent the answer and the offerer SHOULD start resource
>    reservation as soon as the answer is received.  If participation of
>    the peer user agent is not needed (e.g., segmented status 
> type), the
>    offerer MAY start resource reservation before sending the offer and
>    the answerer MAY start it before sending the answer."
> 
> My question is: if the answer (lets assume that it is carried in a 183
> response) is carrying a list of codecs taken from the initial 
> offer (my
> understanding is that this is possible under the answer/offer model),
> shouldn't both user agents wait for the updated (final) 
> answer to be sent
> and received (in the PRACK or the 200 OK to the PRACK) before 
> initiating
> resource reservation? 
> 
> Following on that, I have also noticed that 3GPP call flows 
> as specified in
> TS 24.228 show resource reservation happening after the 200 
> OK for the PRACK
> is sent and received. Couldn't the reservation start just 
> after the PRACK
> message (carrying the final answer) is sent/received?
> 
> So, when can we assume that resource reservation can start? 
> Can we assume
> that it can start as soon as a final offer is sent/received?
> 
> 
> Thank you very much in advance.
> Regards
> Maria 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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  Fri May  2 12:10:37 2003
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To: krisztian.kiss@nokia.com, sip@ietf.org
Subject: RE: [Sip] Question about resource management
Date: Fri, 2 May 2003 16:45:31 +0100 
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Krisztian, thanks a lot for your answer.

I have a further question on this same subject though:
- Could the updated offer in the PRACK contain more than one codec again?
And, if the answer in the 200 OK (PRACK) still contains a list of codecs,
when will resource reservation happend then?

Is any of the user agents (offerer or answerer) obliged to reduce the list
to a single codec at some point in the offer/answer model? 

Thank you very much
Regards
Maria


-----Original Message-----
From: krisztian.kiss@nokia.com [mailto:krisztian.kiss@nokia.com]
Sent: 02 May 2003 16:33
To: Cuevas,MA,Maria,XGH4 R; sip@ietf.org
Subject: RE: [Sip] Question about resource management


Hi Maria, 

a quick clarification for the 3GPP call flows:
according to the very latest changes to the last version of the call flows
document (3GPP TS 24.228 5.4.0): 
- the offerer can start the resource reservation procedure right after
receiving the answer in 183, if the answer is satisfactory, i.e. it contains
single codecs per media flows. In case there is a need to include a second
offer in PRACK in order to reduce to the "single codec per media" case, the
resource reservation can be started right after sending the PRACK.
- the answerer can start the resource reservation procedure right after
having sent the answer in 183, if the answer contains single codec per media
flows. In case the first answer has multiple codecs per media, the answerer
should wait with resource reservation until generating the second answer.

Regards,
Krisztian


> -----Original Message-----
> From: ext maria.a.cuevas@bt.com [mailto:maria.a.cuevas@bt.com]
> Sent: Friday, May 02, 2003 4:35 PM
> To: sip@ietf.org
> Subject: [Sip] Question about resource management
> 
> 
> Dear all,
> 
> I have a question regarding resource management and in 
> particular related to
> at which point in the session establishment may resource 
> reservation be
> initiated. I would appreciate any comments on this.
> 
> I have noticed that the examples given in RFC 3312 seem to 
> indicate that
> resource reservation may be initiated as soon as an answer 
> has been sent (by
> the answerer) or received (by the offerer). At some point the 
> RFC says:
> 
> "If the resource reservation mechanism used requires participation of
>    both user agents, the answerer SHOULD start resource reservation
>    after having sent the answer and the offerer SHOULD start resource
>    reservation as soon as the answer is received.  If participation of
>    the peer user agent is not needed (e.g., segmented status 
> type), the
>    offerer MAY start resource reservation before sending the offer and
>    the answerer MAY start it before sending the answer."
> 
> My question is: if the answer (lets assume that it is carried in a 183
> response) is carrying a list of codecs taken from the initial 
> offer (my
> understanding is that this is possible under the answer/offer model),
> shouldn't both user agents wait for the updated (final) 
> answer to be sent
> and received (in the PRACK or the 200 OK to the PRACK) before 
> initiating
> resource reservation? 
> 
> Following on that, I have also noticed that 3GPP call flows 
> as specified in
> TS 24.228 show resource reservation happening after the 200 
> OK for the PRACK
> is sent and received. Couldn't the reservation start just 
> after the PRACK
> message (carrying the final answer) is sent/received?
> 
> So, when can we assume that resource reservation can start? 
> Can we assume
> that it can start as soon as a final offer is sent/received?
> 
> 
> Thank you very much in advance.
> Regards
> Maria 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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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Hi,

> >>>>1) Since ACK for 200 response indicates a new transaction and follows
> >>>>rules of sec. 4.2 to find out the ip address of next hop element to send
> >>>>ACK to, how does the client transaction detect failure for ACK and move
> >>>>to next record in the list if used with UDP and the next hop does not generate transport errors.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>[CHH] It is true that the remote server will not be able to send eg a 503 response for an ACK, but if the failure occurs on the transport level you will be notified >it the same way as you would for any other request (ICMP errors etc). These are sent independently on if the remote server generates SIP responses or not.
> >>>
> >>>
> >>>
> >>[SS] But the client transport layer will not get ICMP errors if the
> >>first record from the list has crashed or has it's interface shutdown.
> >>
> >>
> >
> >[CHH] Well, if the ACK doesn't reach the server you will get retransmits of the 200 OK (unless they are supposed to be sent from the same interface which has crashed, of course)..
> >
> [SS] Right, but how does the client's transaction figure out that 200
> retransmissions are because of Ack going to the wrong interface from the
> list  vs. Ack being lost in the network/network delay and whether it
> should cycle to the next record or not.

[CHH] I guess you will have to use logic based on local policy to decide when to "switch" from one address to another, if the 200 responses don't stop coming, even if you send ACK...

> >>>2) For PRACK, if the client transaction layer times out waiting for 200
> >>>OK to PRACK before moving to the next record, the server transaction may
> >>>run out of 18x retries waiting for PRACK and disconnect the call, if the
> >>>next record happens to be the one that sent 18x.
> >>>
> >>>
> >>>
> >>>[CHH] In theory, yes. Normally, however, I would say that PRACKs are sent to the same IP address as the INVITE, so if you don't get a reply (even if you get >replies to the INVITE) something is probably wrong on the remote side anway...
> >>>
> >>>
> >>>
> >>[SS] I do not think that's true. If the proxy in-between did  not
> >>record-route, PRACK will be sent to Contact of 18x and not where the
> >>Invite was sent.
> >>
> >>
> >
> >[CHH] True, but what I meant is that the remote UAS (ie the one generating and sending the 18x messages, and that will eventually receive the PRACK) is the same.
> >
> >UAC             PROXY             UAS
> >
> >--- INVITE --->   --->INVITE--->
> ><--18x----------  <---18x----------
> >----PRACK----------------------->
> >
> [SS] Not neccesarily, if the UAS has dual interface (for redundancy
> etc.) and uses fqdn in contact of 18x with one interface having higher
> priority than the other one and the high priority interface down for
> some reason. UAC, after dns resolution of contact's fqdn, will use it
> for sending Prack and will keep  retransmitting till it gets 200 OK and
> will cycle to the next record when it runs out of retries. But the UAS
> is also retransmitting 18x waiting for Prack since the interface it's
> getting Prack on is down and when it runs out of retries will disconnect
> the call.

[CHH] Well, if one interface, which is to be used to send messages to, is not working, I think disconnecting the call is the only thing you can do...

These kind of scenarios have been discussed on the list before. It is possible to come up with an infinite number of them, and the spec can't cover them all, so it's pretty much an implementation issue. In general I would say: if something in the call setup fails you terminate the call setup.

Regards,

Christer Holmberg
Ericsson Finland


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From: Roland Gloeckler <gloeckle@ikr.uni-stuttgart.de>
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To: "Drage, Keith (Keith)" <drage@lucent.com>
Subject: Re: [Sip] Motivation for RFC 3325
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Dear Mr. Drage,

thanks for your answer.
Do you know of other documents coping with ISUP interworking and Network Asserted Identities?

Regards

Roland Glöckler
>
> It is also specified for use in SS7 to ISUP/BICC interworking
> specifications in ITU-T.
>
> The key constraint on its use is that it requires a trust domain to exist,
> and that is why it is not generally applicable in all SIP environments.
>
> Keith
>
> Keith Drage
> Lucent Technologies
> Tel: +44 1793 776249
> Email: drage@lucent.com
>
> > -----Original Message-----
> > From: Roland Gloeckler [mailto:gloeckle@ikr.uni-stuttgart.de]
> > Sent: 30 April 2003 11:16
> > To: mwatson@nortelnetworks.com; Jon.Peterson@NeuStar.biz;
> > fluffy@cisco.com
> > Cc: sip@ietf.org
> > Subject: [Sip] Motivation for RFC 3325
> >
> >
> > Dear Sirs,
> >
> > I would like to know which Scenarios you had in mind for the
> > usage of RFC 3325.
> >
> > In chapter 1 you state the following:
> > "An example deployment would be a closed network which
> > emulates a traditional circuit switched telephone network."
> >
> > Is this the Scenario that motivated you or are the others?
> >
> > Regards
> >
> > Roland Glöckler
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP 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] Question about resource management
Date: Mon, 5 May 2003 21:59:19 +0300
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Hi Maria,

> Krisztian, thanks a lot for your answer.
> 
> I have a further question on this same subject though:
> - Could the updated offer in the PRACK contain more than one 
> codec again?

If either the offerer or the answerer is only able to support the "single codec per media" case, then they should reduce in the first answer (answerer) or in the second offer (offerer).

> And, if the answer in the 200 OK (PRACK) still contains a 
> list of codecs,
> when will resource reservation happend then?

If both parties are able to support multiple codecs per media, then there is no need for the second offer-answer, resource reservation can be performed according to the result of the first offer-answer.
 
> Is any of the user agents (offerer or answerer) obliged to 
> reduce the list
> to a single codec at some point in the offer/answer model? 

Only, if one of them cannot support the multiple codecs per media. But, in any case, two offer-answer exchanges make the job.

Regards,
Krisztian

> 
> Thank you very much
> Regards
> Maria
> 
> 
> -----Original Message-----
> From: krisztian.kiss@nokia.com [mailto:krisztian.kiss@nokia.com]
> Sent: 02 May 2003 16:33
> To: Cuevas,MA,Maria,XGH4 R; sip@ietf.org
> Subject: RE: [Sip] Question about resource management
> 
> 
> Hi Maria, 
> 
> a quick clarification for the 3GPP call flows:
> according to the very latest changes to the last version of 
> the call flows
> document (3GPP TS 24.228 5.4.0): 
> - the offerer can start the resource reservation procedure right after
> receiving the answer in 183, if the answer is satisfactory, 
> i.e. it contains
> single codecs per media flows. In case there is a need to 
> include a second
> offer in PRACK in order to reduce to the "single codec per 
> media" case, the
> resource reservation can be started right after sending the PRACK.
> - the answerer can start the resource reservation procedure 
> right after
> having sent the answer in 183, if the answer contains single 
> codec per media
> flows. In case the first answer has multiple codecs per 
> media, the answerer
> should wait with resource reservation until generating the 
> second answer.
> 
> Regards,
> Krisztian
> 
> 
> > -----Original Message-----
> > From: ext maria.a.cuevas@bt.com [mailto:maria.a.cuevas@bt.com]
> > Sent: Friday, May 02, 2003 4:35 PM
> > To: sip@ietf.org
> > Subject: [Sip] Question about resource management
> > 
> > 
> > Dear all,
> > 
> > I have a question regarding resource management and in 
> > particular related to
> > at which point in the session establishment may resource 
> > reservation be
> > initiated. I would appreciate any comments on this.
> > 
> > I have noticed that the examples given in RFC 3312 seem to 
> > indicate that
> > resource reservation may be initiated as soon as an answer 
> > has been sent (by
> > the answerer) or received (by the offerer). At some point the 
> > RFC says:
> > 
> > "If the resource reservation mechanism used requires 
> participation of
> >    both user agents, the answerer SHOULD start resource reservation
> >    after having sent the answer and the offerer SHOULD 
> start resource
> >    reservation as soon as the answer is received.  If 
> participation of
> >    the peer user agent is not needed (e.g., segmented status 
> > type), the
> >    offerer MAY start resource reservation before sending 
> the offer and
> >    the answerer MAY start it before sending the answer."
> > 
> > My question is: if the answer (lets assume that it is 
> carried in a 183
> > response) is carrying a list of codecs taken from the initial 
> > offer (my
> > understanding is that this is possible under the 
> answer/offer model),
> > shouldn't both user agents wait for the updated (final) 
> > answer to be sent
> > and received (in the PRACK or the 200 OK to the PRACK) before 
> > initiating
> > resource reservation? 
> > 
> > Following on that, I have also noticed that 3GPP call flows 
> > as specified in
> > TS 24.228 show resource reservation happening after the 200 
> > OK for the PRACK
> > is sent and received. Couldn't the reservation start just 
> > after the PRACK
> > message (carrying the final answer) is sent/received?
> > 
> > So, when can we assume that resource reservation can start? 
> > Can we assume
> > that it can start as soon as a final offer is sent/received?
> > 
> > 
> > Thank you very much in advance.
> > Regards
> > Maria 
> > 
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP 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: Hassan CHOUMAR <Hassan.Choumar@int-evry.fr>
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Hello all
Anybody can help me to find information to buy the entities of SIP or
to download them free on the Internet.
Thank you in advance 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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] Question about entities SIP
To: Hassan CHOUMAR <Hassan.Choumar@int-evry.fr>
Cc: sip@ietf.org
Date: Tue, 6 May 2003 16:09:43 +0530
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Hi,

One of helpful sites is vovida.org

regards
Rayees





Hassan CHOUMAR <Hassan.Choumar@int-evry.fr>@ietf.org on 05/06/2003 02:51:32
PM

Sent by:    sip-admin@ietf.org


To:    sip@ietf.org
cc:

Subject:    [Sip] Question about entities SIP


Hello all
Anybody can help me to find information to buy the entities of SIP or
to download them free on the Internet.
Thank you in advance
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
 Use sipping@ietf.org for new developments on the application of sip


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



From mailnull@www1.ietf.org  Tue May  6 07:54:56 2003
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From: =?utf-8?B?67CV7Iq57ZuI?= <surym@brain21.com>
To: "Hassan CHOUMAR" <Hassan.Choumar@int-evry.fr>, <sip@ietf.org>
References: <20030506101545.2441FC21BE@molure.int-evry.fr>
Subject: Re: [Sip] Question about entities SIP
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Dear Hassan,
I got some of you want.

#1 SIP free server and simple test ua. (vocal)
http://www.vovida.org/applications/downloads/vocal/

#2 light weight SIP Stack  (osip)
http://www.gnu.org/software/osip/

#3 linux sip phone (using osip)
http://www.linphone.org

I hope these will help you.

Regards,
Surym
shp
----- Original Message ----- 
From: "Hassan CHOUMAR" <Hassan.Choumar@int-evry.fr>
To: <sip@ietf.org>
Sent: Tuesday, May 06, 2003 6:21 PM
Subject: [Sip] Question about entities SIP


> Hello all
> Anybody can help me to find information to buy the entities of SIP or
> to download them free on the Internet.
> Thank you in advance 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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: Tue, 06 May 2003 17:34:47 +0530
From: Ranjit Avasarala <ranjitk@samsung.com>
Subject: RE: [Sip] Question about entities SIP
In-reply-to: <20030506101545.2441FC21BE@molure.int-evry.fr>
To: "'Hassan CHOUMAR'" <Hassan.Choumar@int-evry.fr>, sip@ietf.org
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U can refer http://www.sipcentre.com

Regards
Ranjit


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Hassan
CHOUMAR
Sent: Tuesday, May 06, 2003 2:52 PM
To: sip@ietf.org
Subject: [Sip] Question about entities SIP


Hello all
Anybody can help me to find information to buy the entities of SIP or to
download them free on the Internet. Thank you in advance 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip Use
sipping@ietf.org for new developments on the application of sip

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



From mailnull@www1.ietf.org  Tue May  6 14:40:16 2003
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From: "sal" <loretosa@vodafone.it>
To: "Shane McCormack" <smccormack@tssg.org>
Cc: <sip@ietf.org>
References: <HKEILFGJCBMBHOCGMAJNEEDNDDAA.smccormack@tssg.org>
Subject: Re: [Sip] SIP INVITE request mapping to call object
Date: Tue, 6 May 2003 20:41:17 +0200
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I Shane,
why don't use directly Java Call Control ?

two years ago I helped a guy in his degree thesis,
we developed a SIP client over JCC...
in those days we didn't implement NOTIFY... but I think
should not be difficulto to do...

But your thesis is not on SIP over SCTP ?
maybe you don't remember we had several discussions some
months ago (more than a year ago... if I don't wrong)

Sal


----- Original Message -----
From: "Shane McCormack" <smccormack@tssg.org>
To: <sip@ietf.org>
Sent: Thursday, May 01, 2003 11:36 AM
Subject: [Sip] SIP INVITE request mapping to call object


> Hi all,
>
> I am currently working on my thesis which involves me creating a
simplified
> generic java API layer above SIP and parlay for easier service
development.
> This involves the creation of call objects similar to Java Call Control,
and
> Parlay.
>
> When mapping an incoming INVITE (or incoming requests such as a NOTIFY for
> presence for other similar objects), some mapping has to be performed. My
> question is would you associate a call object with a request callid or
> dialog? or both. So that when some routing operation is performed by the
> client application using my API the correct incoming INVITE is referenced
in
> the underlying SIP stack when the call object is passed back to be routed
or
> proxied.
>
> I am actually using SIP servlets for this implementation so SIP stack is
not
> correct term to use.
>
> Regards
>
> Shane McCormack
>
> Research Assistant,
> TSSG, Confederation House,
> Waterford Business Park, Cork Road,
> Waterford, Co. Waterford.
>
> Phone: +353 51 302927     Fax: +353 51 302901
> Mobile: +353 87 9955505
> http://www.tssg.org <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

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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] Question about entities SIP
To: <ranjitk@samsung.com>
Cc: "'Hassan CHOUMAR'" <Hassan.Choumar@int-evry.fr>, sip@ietf.org,
        sip-admin@ietf.org
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From: "Vishal Phirke" <vishal_phirke@nmss.com>
Date: Tue, 6 May 2003 17:54:58 -0400
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and lot more here.....
http://www.ict.tuwien.ac.at/darilion/bookmarks-sip-rtp-ua.html
./vishal


                                                                                                           
                      Ranjit Avasarala                                                                     
                      <ranjitk@samsung.        To:       "'Hassan CHOUMAR'" <Hassan.Choumar@int-evry.fr>,  
                      com>                      <sip@ietf.org>                                             
                      Sent by:                 cc:                                                         
                      <sip-admin@ietf.o        Subject:  RE: [Sip] Question about entities SIP             
                      rg>                                                                                  
                                                                                                           
                                                                                                           
                      05/06/2003 08:04                                                                     
                      AM                                                                                   
                      Please respond to                                                                    
                      ranjitk                                                                              
                                                                                                           
                                                                                                           




U can refer http://www.sipcentre.com

Regards
Ranjit


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Hassan
CHOUMAR
Sent: Tuesday, May 06, 2003 2:52 PM
To: sip@ietf.org
Subject: [Sip] Question about entities SIP


Hello all
Anybody can help me to find information to buy the entities of SIP or to
download them free on the Internet. Thank you in advance
_______________________________________________
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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
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_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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This list is for NEW development of the core SIP Protocol
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From: "Elwell, John" <john.elwell@siemens.com>
To: "'sip@ietf.org'" <sip@ietf.org>
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Subject: [Sip] UPDATE and P-Asserted-ID
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In RFC3325 section 9.1 it indicates that P-Asserted-ID is not allowed in
UPDATE. Does anyone know the reason for this? A useful application would be
to indicate a changed asserted identity when a change of party occurs in a
legacy network (e.g., due to transfer). If not allowed in UPDATE, a
re-INVITE would need to be used, but this introduces an extra message.

John Elwell (john.elwell@siemens.com)


_______________________________________________
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Subject: RE: [Sip] UPDATE and P-Asserted-ID
Date: Wed, 7 May 2003 09:57:02 -0500
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Thread-Topic: [Sip] UPDATE and P-Asserted-ID
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From: "Henry Chen" <hjlechen@cisco.com>
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I agreed and sometimes Update is better than re-Invite if we need to avoid mid-dialog media session changes.

Thanks,

Henry

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: Wednesday, May 07, 2003 4:20 AM
> To: 'sip@ietf.org'
> Subject: [Sip] UPDATE and P-Asserted-ID
> 
> 
> In RFC3325 section 9.1 it indicates that P-Asserted-ID is not 
> allowed in
> UPDATE. Does anyone know the reason for this? A useful 
> application would be
> to indicate a changed asserted identity when a change of 
> party occurs in a
> legacy network (e.g., due to transfer). If not allowed in UPDATE, a
> re-INVITE would need to be used, but this introduces an extra message.
> 
> John Elwell (john.elwell@siemens.com)
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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  Wed May  7 14:15:49 2003
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "Elwell, John" <john.elwell@siemens.com>,
        "'sip@ietf.org'"
	 <sip@ietf.org>
Subject: RE: [Sip] UPDATE and P-Asserted-ID
Date: Wed, 7 May 2003 19:10:39 +0100 
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I believe the thinking when the document was developed was essentially to allow it in methods that either initiated a dialog, or were sent outside of a dialog. There was therefore no intent to cover reINVITE at that time, although I do not think it was explicitly excluded in the RFC. I am not sure how the BYE got added in respect of this, maybe someone else can clarify. The NOTIFY is there in respect of a NOTIFY that confirms a SUBSCRIBE dialog.

This would allow the exchange of identities between the end users under these circumstances.

From what you are indicating, it looks like you would want to extend the above to cover target refresh requests.

Keith

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

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: 07 May 2003 10:20
> To: 'sip@ietf.org'
> Subject: [Sip] UPDATE and P-Asserted-ID
> 
> 
> In RFC3325 section 9.1 it indicates that P-Asserted-ID is not 
> allowed in
> UPDATE. Does anyone know the reason for this? A useful 
> application would be
> to indicate a changed asserted identity when a change of 
> party occurs in a
> legacy network (e.g., due to transfer). If not allowed in UPDATE, a
> re-INVITE would need to be used, but this introduces an extra message.
> 
> John Elwell (john.elwell@siemens.com)
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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  Wed May  7 16:25:06 2003
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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] UPDATE and P-Asserted-ID
Date: Wed, 7 May 2003 16:22:09 -0400
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> In RFC3325 section 9.1 it indicates that 
> P-Asserted-ID is not allowed in UPDATE.

Thanks for pointing it out; I had never
noticed it.
 
> Does anyone know the reason for this?
> A useful application would be to indicate a 
> changed asserted identity when a change of 
> party occurs in a legacy network (e.g., due 
> to transfer). If not allowed in UPDATE, a
> re-INVITE would need to be used, but this 
> introduces an extra message.

Another useful application is to allow the
display-name to be updated prior to answer.  
Without such ability, a B2BUA
(or pstn gateway device) would need to
delay sending INVITE while waiting to obtain
display-name from the pstn network or a remote 
database.

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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: "Ashish Naik" <ashishn@mahindrabt.com>
To: "Santharam, Arun [CC]" <asanth01@sprintspectrum.com>,
        "Sip@Ietf. Org" <sip@ietf.org>
Subject: RE: [Sip] Session Mobility in SIP
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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

  *********************************************************
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<DIV><SPAN class=3D875074804-08052003><FONT face=3DArial color=3D#0000ff =
size=3D2>ok. Is=20
there any specific document on IETF and 3GPP that can explain how it is =
being=20
addressed&nbsp; ?</FONT></SPAN></DIV>
<DIV>&nbsp;</DIV>
<DIV><SPAN class=3D328215804-25042003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D328215804-25042003><FONT face=3DArial=20
size=3D2>Ragards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D328215804-25042003><FONT face=3DArial =
size=3D2>Ashish=20
Naik</FONT></SPAN></DIV>
<DIV><SPAN class=3D328215804-25042003><FONT face=3DArial size=3D2>CoE =
Mobile=20
Computing</FONT></SPAN></DIV>
<DIV><SPAN class=3D328215804-25042003><FONT face=3DArial=20
size=3D2>-----------------------------------</FONT></SPAN></DIV>
<DIV><SPAN class=3D328215804-25042003><FONT face=3DArial size=3D2>Phone =
4018100 Ext=20
3070</FONT></SPAN></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE>
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> =
sip-admin@ietf.org=20
  [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Santharam, Arun=20
  [CC]<BR><B>Sent:</B> Wednesday, April 09, 2003 9:42 PM<BR><B>To:</B> =
Ashish=20
  Naik; Sip@Ietf. Org<BR><B>Subject:</B> RE: [Sip] Session Mobility in=20
  SIP<BR><BR></FONT></DIV>
  <DIV class=3DSection1>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">Ashish,</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Please find =
comments=20
  inline.</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Thank=20
  you</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">arun</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DTahoma =
size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">-----Original=20
  Message-----<BR><B><SPAN style=3D"FONT-WEIGHT: bold">From:</SPAN></B> =
Ashish=20
  Naik [mailto:ashishn@mahindrabt.com] <BR><B><SPAN=20
  style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> </SPAN></FONT><FONT =
face=3DTahoma=20
  size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Tahoma">Wednesday, April 09,=20
  2003</SPAN></FONT><FONT face=3DTahoma size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"> </SPAN></FONT><FONT =
face=3DTahoma=20
  size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">4:01=20
  AM</SPAN></FONT><FONT face=3DTahoma size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"><BR><B><SPAN=20
  style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Sip@Ietf. Org<BR><B><SPAN=20
  style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> [Sip] Session Mobility =
in=20
  SIP</SPAN></FONT></P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Times =
New Roman"=20
  size=3D3><SPAN style=3D"FONT-SIZE: 12pt"></SPAN></FONT>&nbsp;</P>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">How is SIP going to =
address=20
  session mobility ? Is there any initiative taken up for this ?=20
  </SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Times =
New Roman"=20
  color=3Dnavy size=3D3><SPAN=20
  style=3D"FONT-SIZE: 12pt; COLOR: navy"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">&gt; This =
has been=20
  addressed in 3GPP standards. Please refer to the 3GPP All IP =
architecture=20
  specifications.</SPAN></FONT></P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Times =
New Roman"=20
  size=3D3><SPAN style=3D"FONT-SIZE: 12pt"></SPAN></FONT>&nbsp;</P></DIV>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">SIP is said to be a =
better=20
  alternative than Mobile IP. Can I have some latest development =
document=20
  references ?</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">&gt; =
&nbsp;SIP is not=20
  going to replace Mobile IP. It is going to leverage Mobile IP for =
session=20
  management in 3G wireless networks.</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"></SPAN></FONT>&nbsp;</P></DIV>
  <P style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Regards,<BR>Ashish=20
  Naik<BR>-------------------<BR>CoE Mobile Computing<BR>Phone: 4018100 =
Ext 3070=20
  </SPAN></FONT></P>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Times =
New Roman"=20
  size=3D3><SPAN=20
  style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV></DIV><BR>***************************=
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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Date: Thu, 08 May 2003 09:21:53 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] UPDATE and P-Asserted-ID
To: "'Drage, Keith (Keith)'" <drage@lucent.com>,
        "'sip@ietf.org'" <sip@ietf.org>
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Keith,

Thanks for your response. However, I do not wish to cover target refresh. As
I understand from section 12 of RFC3261, local and remote targets are
derived from Contact headers. The scenario I have in mind is where something
happens behind a gateway. However, the gateway remains the same and
therefore it does not need to generate a new Contact header.

John (john.elwell@siemens.com)

> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> Sent: 07 May 2003 19:11
> To: Elwell, John; 'sip@ietf.org'
> Subject: RE: [Sip] UPDATE and P-Asserted-ID
> 
> 
> I believe the thinking when the document was developed was 
> essentially to allow it in methods that either initiated a 
> dialog, or were sent outside of a dialog. There was therefore 
> no intent to cover reINVITE at that time, although I do not 
> think it was explicitly excluded in the RFC. I am not sure 
> how the BYE got added in respect of this, maybe someone else 
> can clarify. The NOTIFY is there in respect of a NOTIFY that 
> confirms a SUBSCRIBE dialog.
> 
> This would allow the exchange of identities between the end 
> users under these circumstances.
> 
> From what you are indicating, it looks like you would want to 
> extend the above to cover target refresh requests.
> 
> Keith
> 
> Keith Drage
> Lucent Technologies
> Tel: +44 1793 776249
> Email: drage@lucent.com 
> 
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens.com]
> > Sent: 07 May 2003 10:20
> > To: 'sip@ietf.org'
> > Subject: [Sip] UPDATE and P-Asserted-ID
> > 
> > 
> > In RFC3325 section 9.1 it indicates that P-Asserted-ID is not 
> > allowed in
> > UPDATE. Does anyone know the reason for this? A useful 
> > application would be
> > to indicate a changed asserted identity when a change of 
> > party occurs in a
> > legacy network (e.g., due to transfer). If not allowed in UPDATE, a
> > re-INVITE would need to be used, but this introduces an 
> extra message.
> > 
> > John Elwell (john.elwell@siemens.com)
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
_______________________________________________
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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 
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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 

 


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<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff size=2>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.</FONT></SPAN></DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff size=2>The 
SIP protocol application is covered in 3GPP TS 24.229 with example flows in 3GPP 
TS 24.228.</FONT></SPAN></DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff size=2>These 
documents may be found at:</FONT></SPAN></DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff size=2><A 
href="ftp://ftp.3gpp.org/Specs/latest/Rel-5/">ftp://ftp.3gpp.org/Specs/latest/Rel-5/</A></FONT></SPAN></DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff size=2>3GPP 
uses these specifications over GPRS to provide mobility.</FONT></SPAN></DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff size=2>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.</FONT></SPAN></DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff 
size=2>regards </FONT></SPAN></DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff 
size=2>Keith</FONT></SPAN></DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=560283713-08052003><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<P><FONT size=2>Keith Drage<BR>Lucent Technologies<BR>Tel: +44 1793 
776249<BR>Email: drage@lucent.com </FONT></P>
<BLOCKQUOTE 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Ashish Naik 
  [mailto:ashishn@mahindrabt.com]<BR><B>Sent:</B> 08 May 2003 
  06:23<BR><B>To:</B> Santharam, Arun [CC]; Sip@Ietf. Org<BR><B>Subject:</B> RE: 
  [Sip] Session Mobility in SIP<BR><BR></FONT></DIV>
  <DIV><SPAN class=875074804-08052003><FONT face=Arial color=#0000ff size=2>ok. 
  Is there any specific document on IETF and 3GPP that can explain how it is 
  being addressed&nbsp; ?</FONT></SPAN></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><SPAN class=328215804-25042003><FONT face=Arial 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=328215804-25042003><FONT face=Arial 
  size=2>Ragards,</FONT></SPAN></DIV>
  <DIV><SPAN class=328215804-25042003><FONT face=Arial size=2>Ashish 
  Naik</FONT></SPAN></DIV>
  <DIV><SPAN class=328215804-25042003><FONT face=Arial size=2>CoE Mobile 
  Computing</FONT></SPAN></DIV>
  <DIV><SPAN class=328215804-25042003><FONT face=Arial 
  size=2>-----------------------------------</FONT></SPAN></DIV>
  <DIV><SPAN class=328215804-25042003><FONT face=Arial size=2>Phone 4018100 Ext 
  3070</FONT></SPAN></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <BLOCKQUOTE>
    <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
    size=2>-----Original Message-----<BR><B>From:</B> sip-admin@ietf.org 
    [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Santharam, Arun 
    [CC]<BR><B>Sent:</B> Wednesday, April 09, 2003 9:42 PM<BR><B>To:</B> Ashish 
    Naik; Sip@Ietf. Org<BR><B>Subject:</B> RE: [Sip] Session Mobility in 
    SIP<BR><BR></FONT></DIV>
    <DIV class=Section1>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Ashish,</SPAN></FONT></P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Please find 
    comments inline.</SPAN></FONT></P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Thank 
    you</SPAN></FONT></P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">arun</SPAN></FONT></P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
    <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face=Tahoma size=2><SPAN 
    style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">-----Original 
    Message-----<BR><B><SPAN style="FONT-WEIGHT: bold">From:</SPAN></B> Ashish 
    Naik [mailto:ashishn@mahindrabt.com] <BR><B><SPAN 
    style="FONT-WEIGHT: bold">Sent:</SPAN></B> </SPAN></FONT><FONT face=Tahoma 
    size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">Wednesday, April 
    09, 2003</SPAN></FONT><FONT face=Tahoma size=2><SPAN 
    style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"> </SPAN></FONT><FONT 
    face=Tahoma size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">4:01 
    AM</SPAN></FONT><FONT face=Tahoma size=2><SPAN 
    style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"><BR><B><SPAN 
    style="FONT-WEIGHT: bold">To:</SPAN></B> Sip@Ietf. Org<BR><B><SPAN 
    style="FONT-WEIGHT: bold">Subject:</SPAN></B> [Sip] Session Mobility in 
    SIP</SPAN></FONT></P>
    <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" 
    size=3><SPAN style="FONT-SIZE: 12pt"></SPAN></FONT>&nbsp;</P>
    <DIV>
    <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face=Arial size=2><SPAN 
    style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">How is SIP going to address 
    session mobility ? Is there any initiative taken up for this ? 
    </SPAN></FONT></P></DIV>
    <DIV>
    <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" 
    color=navy size=3><SPAN 
    style="FONT-SIZE: 12pt; COLOR: navy"></SPAN></FONT>&nbsp;</P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">&gt; This has been 
    addressed in 3GPP standards. Please refer to the 3GPP All IP architecture 
    specifications.</SPAN></FONT></P>
    <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" 
    size=3><SPAN style="FONT-SIZE: 12pt"></SPAN></FONT>&nbsp;</P></DIV>
    <DIV>
    <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face=Arial size=2><SPAN 
    style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">SIP is said to be a better 
    alternative than Mobile IP. Can I have some latest development document 
    references ?</SPAN></FONT></P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">&gt; &nbsp;SIP is 
    not going to replace Mobile IP. It is going to leverage Mobile IP for 
    session management in 3G wireless networks.</SPAN></FONT></P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
    <P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN 
    style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P></DIV>
    <P style="MARGIN-LEFT: 0.5in"><FONT face=Arial size=2><SPAN 
    style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Regards,<BR>Ashish 
    Naik<BR>-------------------<BR>CoE Mobile Computing<BR>Phone: 4018100 Ext 
    3070 </SPAN></FONT></P>
    <DIV>
    <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" 
    size=3><SPAN 
    style="FONT-SIZE: 12pt"></SPAN></FONT>&nbsp;</P></DIV></DIV><BR>*********************************************************<BR>Disclaimer<BR><BR>This 
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From: =?Windows-1252?Q?Manuel_Moreno_Mart=EDn?= <mmoreno@cipres.upm.es>
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Please

I want to clarify me in relation to the following affirmation that
appears in the page 125 of RFC 3261


"The INVITE transaction is different from those of other methods
   because of its extended duration.  Normally, human input is required
   in order to respond to an INVITE.  The long delays expected for
   sending a response argue for a three-way handshake.  On the other
   hand, requests of other methods are expected to complete rapidly.
   Because of the non-INVITE transaction's reliance on a two-way
   handshake, TUs SHOULD respond immediately to non-INVITE requests."

Then,

"The long delays expected for
   sending a response argue for a three-way handshake "

only for this the "three-way handshake"???

Aren´t there others reasons??

Thank you

Manuel

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From: murali <muraliv@bharatmail.com>
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Date: Wed May 7 08:56:23  2003
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Hi,

In section 7.3.1 of RFC 3261 it says

"The exceptions to this rule are the WWW-Authenticate, Authorization, Proxy-Authenticate, and Proxy-Authorization header fields. Multiple header field rows with these names MAY be present in a message, but since their grammar does not follow the general form listed in Section 7.3,   they MUST NOT be combined into a single header field row."

Could any on tell me the possibility wherein we can receive multiple headers of the above types??

Thanks & Regards,
Murali

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

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Date: Fri, 09 May 2003 14:48:03 +0530
From: "Nataraju A.B." <natarajuab@huawei.com>
Subject: Re: [Sip] question
To: =?Windows-1252?Q?Manuel_Moreno_Mart=EDn?= <mmoreno@cipres.upm.es>,
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The coments inline.

Regards,
-Nataraju A.B.
  ----- Original Message -----=20
  From: Manuel Moreno Mart=EDn=20
  To: Sip=20
  Sent: Friday, May 09, 2003 2:08 PM
  Subject: [Sip] question



  Please

  I want to clarify me in relation to the following affirmation that
  appears in the page 125 of RFC 3261


  "The INVITE transaction is different from those of other methods
     because of its extended duration.  Normally, human input is =
required
     in order to respond to an INVITE.  The long delays expected for
     sending a response argue for a three-way handshake.  On the other
     hand, requests of other methods are expected to complete rapidly.
     Because of the non-INVITE transaction's reliance on a two-way
     handshake, TUs SHOULD respond immediately to non-INVITE requests."

  Then,

  "The long delays expected for
     sending a response argue for a three-way handshake "

  only for this the "three-way handshake"???

  [Nataraju A.B.] The other reasons are : we can always expect the user =
intervension need much more time than automatic response. the other =
tranactions normally does not need the human intervension.=20

  Then need for 3-way handshake could be because the 200 response =
constitutes the 2nd half part the dilaog component(To-tag). where as for =
the mid-dialog transactions 2-way handshake is enough (3-way handshake =
not required, because teh dlg is already established)

  Aren=B4t there others reasons??

  Thank you

  Manuel

  _______________________________________________
  Sip mailing list  https://www1.ietf.org/mailman/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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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dwindows-1252">
<META content=3D"MSHTML 5.50.4134.600" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY style=3D"COLOR: #000080; FONT-FAMILY: Courier New" =
bgColor=3D#ffffff>
<DIV><FONT size=3D2>The coments inline.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Regards,<BR>-Nataraju A.B.</FONT></DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000080 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3Dmmoreno@cipres.upm.es =
href=3D"mailto:mmoreno@cipres.upm.es">Manuel=20
  Moreno Mart=EDn</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A title=3Dsip@ietf.org=20
  href=3D"mailto:sip@ietf.org">Sip</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Friday, May 09, 2003 2:08 =
PM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [Sip] question</DIV>
  <DIV><FONT size=3D2></FONT><FONT size=3D2></FONT><FONT =
size=3D2></FONT><FONT=20
  size=3D2></FONT><FONT size=3D2></FONT><BR></DIV>
  <DIV><BR>Please<BR><BR>I want to clarify me in relation to the =
following=20
  affirmation that<BR>appears in the page 125 of RFC =
3261<BR><BR><BR>"The INVITE=20
  transaction is different from those of other methods<BR>&nbsp;&nbsp; =
because=20
  of its extended duration.&nbsp; Normally, human input is=20
  required<BR>&nbsp;&nbsp; in order to respond to an INVITE.&nbsp; The =
long=20
  delays expected for<BR>&nbsp;&nbsp; sending a response argue for a =
three-way=20
  handshake.&nbsp; On the other<BR>&nbsp;&nbsp; hand, requests of other =
methods=20
  are expected to complete rapidly.<BR>&nbsp;&nbsp; Because of the =
non-INVITE=20
  transaction's reliance on a two-way<BR>&nbsp;&nbsp; handshake, TUs =
SHOULD=20
  respond immediately to non-INVITE requests."<BR><BR>Then,<BR><BR>"The =
long=20
  delays expected for<BR>&nbsp;&nbsp; sending a response argue for a =
three-way=20
  handshake "<BR><BR>only for this the "three-way handshake"???<BR><FONT =

  size=3D2></FONT></DIV>
  <DIV><FONT size=3D2>[Nataraju A.B.] The other reasons are : we can =
always expect=20
  the user intervension need much more time than automatic response. the =
other=20
  tranactions normally does not need the human intervension. =
</FONT></DIV>
  <DIV><FONT size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT size=3D2>Then need for 3-way handshake could be because the =
200=20
  response constitutes the 2nd half part the dilaog component(To-tag). =
where as=20
  for the mid-dialog transactions 2-way handshake is enough (3-way =
handshake not=20
  required, because teh dlg is already established)</FONT></DIV>
  <DIV><FONT size=3D2></FONT><FONT size=3D2></FONT><FONT =
size=3D2></FONT><FONT=20
  size=3D2></FONT><FONT size=3D2></FONT><FONT size=3D2></FONT><FONT=20
  size=3D2></FONT><FONT size=3D2></FONT><FONT size=3D2></FONT><FONT=20
  size=3D2></FONT><FONT size=3D2></FONT><FONT =
size=3D2></FONT><BR>Aren=B4t there others=20
  reasons??<BR><BR>Thank=20
  =
you<BR><BR>Manuel<BR><BR>_______________________________________________<=
BR>Sip=20
  mailing list&nbsp; <A=20
  =
href=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org=
/mailman/listinfo/sip</A><BR>This=20
  list is for NEW development of the core SIP Protocol<BR>Use <A=20
  =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>=20
  for questions on current sip<BR>Use <A=20
  href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> for new =
developments on=20
  the application of sip</DIV></BLOCKQUOTE></BODY></HTML>

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Is there any recent standard work on the use of SIP by media servers/IVR to send out params/collect digits/... to some other SIP entity ?

I know there are many proprietary implementations but I'm not aware about a common agreement that solves this issue. Any INFO/Info-Type or Event package solution?

Any suggestion appreciated.
Thanks in advance.
Regards,

Walter Goix

________________________
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Forked request. The proxy that forked will collect all the challenges into one response and forward upstream that response.

Regards,
Hisham

> -----Original Message-----
> From: ext murali [mailto:muraliv@bharatmail.com]
> Sent: Wednesday, May 07, 2003 11:56 AM
> To: sip@ietf.org
> Subject: [Sip] Multiple header presence
> 
> 
> Hi,
> 
> In section 7.3.1 of RFC 3261 it says
> 
> "The exceptions to this rule are the WWW-Authenticate, 
> Authorization, Proxy-Authenticate, and Proxy-Authorization 
> header fields. Multiple header field rows with these names 
> MAY be present in a message, but since their grammar does not 
> follow the general form listed in Section 7.3,   they MUST 
> NOT be combined into a single header field row."
> 
> Could any on tell me the possibility wherein we can receive 
> multiple headers of the above types??
> 
> Thanks & Regards,
> Murali
> 
> -------
> Murali Voleti,
> #503,Maheswari complex,
> Masab Tank,
> Hyderabad
> Tele:(040) 6502272 ext:211
> 
> _____________________________________________________________
> Get Your Free ScanMail and Email At http://mail.ttkbharatplanet.com
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Date: Fri, 09 May 2003 19:55:05 +0800
From: Prasanna Venkatesh <prasanna@huawei.com>
Subject: RE: [Sip] Multiple header presence
In-reply-to: <200305070604.h4764pV06153@bharatmail.com>
To: murali <muraliv@bharatmail.com>, sip@ietf.org
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There are 3 possibilities
Possibility A:
If the message traverses through 2 proxies, with both of them requiring
authentication this may happen.

Possibility B:
This can also happen when the proxies merge forked responses.
For e.g.
    |--B1
A--P
    |--B2
If both B1 and B2 challenge, then the 401 will carry two WWW-Authenticate
headers.

Possibility C:
However, I am not sure about the third possibility where a single entity
governing two realms, inserts two Authenticate headers or an entity capable
of supporting more than one authentication scheme (for e.g. Basic & Digest )
inserts more than one Authenticate headers.
Can somebody tell if this is allowed as per RFC-2617 or RFC-3261

Cheers,
Prasanna

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of murali
Sent: Friday, May 09, 2003 5:24 PM
To: sip@ietf.org
Subject: [Sip] Multiple header presence


Hi,

In section 7.3.1 of RFC 3261 it says

"The exceptions to this rule are the WWW-Authenticate, Authorization,
Proxy-Authenticate, and Proxy-Authorization header fields. Multiple header
field rows with these names MAY be present in a message, but since their
grammar does not follow the general form listed in Section 7.3,   they MUST
NOT be combined into a single header field row."

Could any on tell me the possibility wherein we can receive multiple headers
of the above types??

Thanks & Regards,
Murali

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

_____________________________________________________________
Get Your Free ScanMail and Email At http://mail.ttkbharatplanet.com
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Date: Fri, 09 May 2003 19:55:04 +0800
From: Prasanna Venkatesh <prasanna@huawei.com>
Subject: RE: [Sip] question
In-reply-to: <008101c3160b$e653aa80$5d02120a@in.huawei.com>
To: "Nataraju A.B." <natarajuab@huawei.com>,
        =?Windows-1252?Q?Manuel_Moreno_Mart=EDn?= <mmoreno@cipres.upm.es>,
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The three way handshake is also required as the 200 for the INVITE me=
ssage
(as of 3261) requires to carry the media information.  If there is no
acknowledgement for the 200, then the callee terminal has to depend o=
n other
mechanisms like media sensing to remove the media binding what it has=
 done.
Cheers,
Prasanna
  -----Original Message-----
  From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Na=
taraju
A.B.
  Sent: Friday, May 09, 2003 5:18 PM
  To: Manuel Moreno Mart=EDn; Sip
  Subject: Re: [Sip] question


  The coments inline.

  Regards,
  -Nataraju A.B.
    ----- Original Message -----
    From: Manuel Moreno Mart=EDn
    To: Sip
    Sent: Friday, May 09, 2003 2:08 PM
    Subject: [Sip] question



    Please

    I want to clarify me in relation to the following affirmation tha=
t
    appears in the page 125 of RFC 3261


    "The INVITE transaction is different from those of other methods
       because of its extended duration.  Normally, human input is re=
quired
       in order to respond to an INVITE.  The long delays expected fo=
r
       sending a response argue for a three-way handshake.  On the ot=
her
       hand, requests of other methods are expected to complete rapid=
ly.
       Because of the non-INVITE transaction's reliance on a two-way
       handshake, TUs SHOULD respond immediately to non-INVITE reques=
ts."

    Then,

    "The long delays expected for
       sending a response argue for a three-way handshake "

    only for this the "three-way handshake"???

    [Nataraju A.B.] The other reasons are : we can always expect the =
user
intervension need much more time than automatic response. the other
tranactions normally does not need the human intervension.

    Then need for 3-way handshake could be because the 200 response
constitutes the 2nd half part the dilaog component(To-tag). where as =
for the
mid-dialog transactions 2-way handshake is enough (3-way handshake no=
t
required, because teh dlg is already established)

    Aren=B4t there others reasons??

    Thank you

    Manuel

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


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dwindows-1252">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY style=3D"COLOR: #000080; FONT-FAMILY: Courier New" =
bgColor=3D#ffffff>
<DIV><SPAN class=3D863374711-09052003><FONT size=3D2>The three way =
handshake is also=20
required as the 200 for the INVITE message (as of 3261) requires to =
carry the=20
media information.&nbsp; If there is no acknowledgement for the 200, =
then the=20
callee terminal has to depend on other mechanisms like media sensing to =
remove=20
the media binding what it has done.</FONT></SPAN></DIV>
<DIV><SPAN class=3D863374711-09052003><FONT =
size=3D2>Cheers,</FONT></SPAN></DIV>
<DIV><SPAN class=3D863374711-09052003><FONT =
size=3D2>Prasanna</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> sip-admin@ietf.org =

  [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Nataraju =
A.B.<BR><B>Sent:</B>=20
  Friday, May 09, 2003 5:18 PM<BR><B>To:</B> Manuel Moreno Mart=EDn;=20
  Sip<BR><B>Subject:</B> Re: [Sip] question<BR><BR></FONT></DIV>
  <DIV><FONT size=3D2>The coments inline.</FONT></DIV>
  <DIV><FONT size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>Regards,<BR>-Nataraju =
A.B.</FONT></DIV>
  <BLOCKQUOTE=20
  style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000080 2px solid; MARGIN-RIGHT: 0px">
    <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
    <DIV=20
    style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
    <A title=3Dmmoreno@cipres.upm.es =
href=3D"mailto:mmoreno@cipres.upm.es">Manuel=20
    Moreno Mart=EDn</A> </DIV>
    <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A title=3Dsip@ietf.org=20
    href=3D"mailto:sip@ietf.org">Sip</A> </DIV>
    <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Friday, May 09, 2003 =
2:08=20
PM</DIV>
    <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [Sip] question</DIV>
    <DIV><FONT size=3D2></FONT><FONT size=3D2></FONT><FONT =
size=3D2></FONT><FONT=20
    size=3D2></FONT><FONT size=3D2></FONT><BR></DIV>
    <DIV><BR>Please<BR><BR>I want to clarify me in relation to the =
following=20
    affirmation that<BR>appears in the page 125 of RFC =
3261<BR><BR><BR>"The=20
    INVITE transaction is different from those of other =
methods<BR>&nbsp;&nbsp;=20
    because of its extended duration.&nbsp; Normally, human input is=20
    required<BR>&nbsp;&nbsp; in order to respond to an INVITE.&nbsp; The =
long=20
    delays expected for<BR>&nbsp;&nbsp; sending a response argue for a =
three-way=20
    handshake.&nbsp; On the other<BR>&nbsp;&nbsp; hand, requests of =
other=20
    methods are expected to complete rapidly.<BR>&nbsp;&nbsp; Because of =
the=20
    non-INVITE transaction's reliance on a two-way<BR>&nbsp;&nbsp; =
handshake,=20
    TUs SHOULD respond immediately to non-INVITE=20
    requests."<BR><BR>Then,<BR><BR>"The long delays expected =
for<BR>&nbsp;&nbsp;=20
    sending a response argue for a three-way handshake "<BR><BR>only for =
this=20
    the "three-way handshake"???<BR><FONT size=3D2></FONT></DIV>
    <DIV><FONT size=3D2>[Nataraju A.B.] The other reasons are : we can =
always=20
    expect the user intervension need much more time than automatic =
response.=20
    the other tranactions normally does not need the human intervension. =

    </FONT></DIV>
    <DIV><FONT size=3D2></FONT>&nbsp;</DIV>
    <DIV><FONT size=3D2>Then need for 3-way handshake could be because =
the 200=20
    response constitutes the 2nd half part the dilaog component(To-tag). =
where=20
    as for the mid-dialog transactions 2-way handshake is enough (3-way=20
    handshake not required, because teh dlg is already =
established)</FONT></DIV>
    <DIV><FONT size=3D2></FONT><FONT size=3D2></FONT><FONT =
size=3D2></FONT><FONT=20
    size=3D2></FONT><FONT size=3D2></FONT><FONT size=3D2></FONT><FONT=20
    size=3D2></FONT><FONT size=3D2></FONT><FONT size=3D2></FONT><FONT=20
    size=3D2></FONT><FONT size=3D2></FONT><FONT =
size=3D2></FONT><BR>Aren=B4t there=20
    others reasons??<BR><BR>Thank=20
    =
you<BR><BR>Manuel<BR><BR>_______________________________________________<=
BR>Sip=20
    mailing list&nbsp; <A=20
    =
href=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org=
/mailman/listinfo/sip</A><BR>This=20
    list is for NEW development of the core SIP Protocol<BR>Use <A=20
    =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>=20
    for questions on current sip<BR>Use <A=20
    href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> for new =
developments on=20
    the application of sip</DIV></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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Date: Fri, 09 May 2003 11:28:32 -0700
Subject: Re: [Sip] Media Server info notification
From: Cullen Jennings <fluffy@cisco.com>
To: Goix Walter <Walter.Goix@tilab.com>, sip-ietf <sip@ietf.org>
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Have a look at the app-interaction stuff and KPML.

On 5/9/03 2:29 AM, "Goix Walter" <Walter.Goix@tilab.com> wrote:

> Is there any recent standard work on the use of SIP by media servers/IVR to
> send out params/collect digits/... to some other SIP entity ?
> 
> I know there are many proprietary implementations but I'm not aware about a
> common agreement that solves this issue. Any INFO/Info-Type or Event package
> solution?
> 
> Any suggestion appreciated.
> Thanks in advance.
> Regards,
> 
> Walter Goix
> 
> ________________________
> Laurent-Walter Goix
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Subject: [Sip] integration of sip and cops
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hi all,
anybody knows if there's some work on the integration
of sip and cops protocols accordingly to the QoS
architectures : IntServ and DiffServ. 
thanks for any help.
regards.

hajer.




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From mailnull@www1.ietf.org  Mon May 12 07:17:19 2003
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Subject: [Sip] Duration to re-register at registrar
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I am wondering one thing. 
How long will it take for a UA to re-register at a registrar?
E.g when changing contact information or caller preferences parameters.

Thanks

/A





Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands.
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Subject: [Sip] SIP B2BUA vs Proxy
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Dear all:
I have the following questions about SIP B2BUA vs SIP proxy:

1. For service support is B2BUA better than SIP proxy?

2. Is B2BUA suitable to be used as routing entity in the following SIP/SIPT network planning schemas?
    -Hiberarchy structure like PSTN, which has hiberarchied proxies/routing entities. Some one believe
     that B2BUA can be used to substitute call stateful proxy at the edge of such a SIP/SIPT network
    -Flat structure, which has peered routing entities but hiberarched routing database. In this case some
     one believe that B2BUA is even more suitable as routing entity

Thanks!

Sincerely yours, 
Chen Zaifeng
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From mailnull@www1.ietf.org  Mon May 12 19:14:21 2003
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Subject: Re: [Sip] UPDATE and P-Asserted-ID
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John,

I believe you still should generate a new Contact header, even if you 
are only changing the display name.

thanks,
-rohan

On Thursday, May 8, 2003, at 01:21 AM, Elwell, John wrote:

> Keith,
>
> Thanks for your response. However, I do not wish to cover target 
> refresh. As
> I understand from section 12 of RFC3261, local and remote targets are
> derived from Contact headers. The scenario I have in mind is where 
> something
> happens behind a gateway. However, the gateway remains the same and
> therefore it does not need to generate a new Contact header.
>
> John (john.elwell@siemens.com)
>
>> -----Original Message-----
>> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
>> Sent: 07 May 2003 19:11
>> To: Elwell, John; 'sip@ietf.org'
>> Subject: RE: [Sip] UPDATE and P-Asserted-ID
>>
>>
>> I believe the thinking when the document was developed was
>> essentially to allow it in methods that either initiated a
>> dialog, or were sent outside of a dialog. There was therefore
>> no intent to cover reINVITE at that time, although I do not
>> think it was explicitly excluded in the RFC. I am not sure
>> how the BYE got added in respect of this, maybe someone else
>> can clarify. The NOTIFY is there in respect of a NOTIFY that
>> confirms a SUBSCRIBE dialog.
>>
>> This would allow the exchange of identities between the end
>> users under these circumstances.
>>
>> From what you are indicating, it looks like you would want to
>> extend the above to cover target refresh requests.
>>
>> Keith
>>
>> Keith Drage
>> Lucent Technologies
>> Tel: +44 1793 776249
>> Email: drage@lucent.com
>>
>>> -----Original Message-----
>>> From: Elwell, John [mailto:john.elwell@siemens.com]
>>> Sent: 07 May 2003 10:20
>>> To: 'sip@ietf.org'
>>> Subject: [Sip] UPDATE and P-Asserted-ID
>>>
>>>
>>> In RFC3325 section 9.1 it indicates that P-Asserted-ID is not
>>> allowed in
>>> UPDATE. Does anyone know the reason for this? A useful
>>> application would be
>>> to indicate a changed asserted identity when a change of
>>> party occurs in a
>>> legacy network (e.g., due to transfer). If not allowed in UPDATE, a
>>> re-INVITE would need to be used, but this introduces an
>> extra message.
>>>
>>> John Elwell (john.elwell@siemens.com)
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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A vWijk wrote:
> I am wondering one thing. 
> How long will it take for a UA to re-register at a registrar?
> E.g when changing contact information or caller preferences parameters.
> 

would network RTT + 10 to 100ms sound appropriate? Or, in the event of 
an HTTP-style authentication on the request, 2*RTT + 20-200ms.

Of course some registrars with more complex data storage models (such as 
LDAP) might take a little longer.

Or perhaps I'm not guessing the question that you're really asking.

--
Dean

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Subject: Re: [Sip] Media Server info notification
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Cc: Goix Walter <Walter.Goix@tilab.com>, sip-ietf <sip@ietf.org>
To: Cullen Jennings <fluffy@cisco.com>
From: Rohan Mahy <rohan@cisco.com>
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specifically:

http://www.softarmor.com/sipping/drafts/draft-burger-sipping-kpml-01.txt
http://www.softarmor.com/sipping/drafts/draft-culpepper-sipping-app- 
interact-reqs-03.txt
http://www.softarmor.com/sipping/drafts/draft-jennings-sip-app-info- 
00.txt
http://www.softarmor.com/sipping/drafts/draft-rosenberg-sipping-app- 
interaction-framework-00.txt

On Friday, May 9, 2003, at 11:28 AM, Cullen Jennings wrote:

>
>
> Have a look at the app-interaction stuff and KPML.
>
> On 5/9/03 2:29 AM, "Goix Walter" <Walter.Goix@tilab.com> wrote:
>
>> Is there any recent standard work on the use of SIP by media  
>> servers/IVR to
>> send out params/collect digits/... to some other SIP entity ?
>>
>> I know there are many proprietary implementations but I'm not aware  
>> about a
>> common agreement that solves this issue. Any INFO/Info-Type or Event  
>> package
>> solution?
>>
>> Any suggestion appreciated.
>> Thanks in advance.
>> Regards,
>>
>> Walter Goix
>>
>> ________________________
>> Laurent-Walter Goix
>> Telecom Italia S.p.A.
>> Telecom Italia Lab - StarSIP
>> Services Innovation - Intelligent Services and Platforms (SI/P)
>> Via Reiss Romoli 274
>> I-10148 Torino (Italy)
>> Tel. +39 011 228 5046
>> Fax +39 011 228 5069
>> E-mail mailto:laurentwalter.goix@telecomitalia.it
>> ________________________
>> Visit us @ http://starsip.telecomitalialab.com
>>
>>
>>
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>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
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>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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Subject: ??: [Sip] SIP B2BUA vs Proxy
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Really appreciate your answer, dear Mr.Arjun.

So according with your description may I draw the conclusion that B2BUA is mainly used for service
control and should not substitute proxy as routing entity in a SIP network?

Thanks!

-----Ô­Ê¼ÓÊ¼þ-----
·¢¼þÈË: Arjun Roychowdhury [mailto:aroychow@hns.com]
·¢ËÍÊ±¼ä: Monday, May 12, 2003 10:24 PM
ÊÕ¼þÈË: Chen Zaifeng,BISC TD DEW5(BJ)
³­ËÍ: sip-implementors@cs.columbia.edu
Ö÷Ìâ: Re: [Sip] SIP B2BUA vs Proxy


Moving to implementors..

Inline, prefixed by ARC>

1. For service support is B2BUA better than SIP proxy?

<ARC>
Depends on what kind of service. There are many that do not require 
a B2B at all and can be done by a pure proxy along with CPL, for example.
(Example of such services include hunt groups, call fwd, etc. etc.)
It also depends on the kind of capabilities you can expect from your 
participating
end points. For example, in the SIP model, services such a transfer can be 
directly
hosted on the endpoints by using REFER mechanism. However, it is possible 
in many
deployments one cannot assume intelligent endpoints nor support for REFER 
and in that case, such
services need to be done centrally.

In a nutshell, what you plan to use as the 'service manager' depends on:
        a) What is the nature of service (routing based pre-call service, 
mid-call service etc)
        b) What is the network you are deploying into (assumptions on 
'intelligence' of nodes)
        c) Scalability reqts of your network 


2. Is B2BUA suitable to be used as routing entity in the following 
SIP/SIPT network planning schemas?
    -Hiberarchy structure like PSTN, which has hiberarchied 
proxies/routing entities. Some one believe that B2BUA can be used to 
substitute call stateful proxy at the edge of such a SIP/SIPT network 
-Flat structure, which has peered routing entities but hiberarched routing 
database. In this case some  one believe that B2BUA is even more suitable 
as routing entity

<ARC> 
Again, if I just go by your requirement of the above, I dont believe a B2B 
is required only for this.
If your question is, 'Even if I need a pure proxy ,can I achieve it by a 
B2B' the answer would
be yes - but you need to decide why - by looking at the overall picture of 
your network which
you would know best. For example, does your server need to kick-in mid 
call and perform some
services which need signalling control to be performed by your server ?
If not, you can do without a B2B. Even if required, can you
assume that your network can support endpoints being REFERed to a b2b for 
mid-call services that 
require it ?
If so, you dont need to be a B2B all the time. And so on...
Also do note, a B2B has implied complexities as well, wrt e-e encryption, 
looping etc. You need to be aware of the same.

And then, do read through previous archives, where you will come across 
interesting 'hybrids'
such as 'transparent proxy' , 'B2B proxy' and so forth.

In short, many implementors have combined the functionality of a proxy and 
a B2B and actually
switch between these modes internally depending on the service required. 
Some do it by being 
a pure B2B and some do it by playing 'dialog id' tricks and switching 
between proxy and B2B (for
those who cannot afford to B2B all calls all the time). 
Needless to say, the implementors need to be aware of the perils of doing 
the same and judge 
for themselves between the devil (above mentioned hacks)  and the deep 
blue sea.. (not believing a world exists today that is not pure SIP ;-)

Id suggest you read the following:

http://www.softarmor.com/sipping/drafts/draft-ietf-sipping-service-examples-04.txt
http://www.softarmor.com/sipping/drafts/draft-ietf-sipping-3pcc-03.txt
http://www.ietf.org/rfc/rfc2824.txt (CPL requirements)
http://www.ietf.org/internet-drafts/draft-ietf-iptel-cpl-06.txt
http://www1.cs.columbia.edu/sip/talks.html (some interesting papers on 
implementing certain services
with SIP)

A lot of services and how to do them are encapsulated in the above. There 
are more
drafts for your perusal too.

So basically, you need to decide.

regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860 



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From: "Donald Papalilo" <d.papalilo@pointercom.it>
To: "hajer ferjani" <fer_hajer@yahoo.fr>, <sip@ietf.org>
Cc: "Luca Veltri" <veltri@coritel.it>,
        "Stefano Salsano" <stefano.salsano@uniroma2.it>
References: <20030512022109.25823.qmail@web12202.mail.yahoo.com>
Subject: Re: [Sip] integration of sip and cops
Date: Mon, 12 May 2003 09:37:52 +0200
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Hi ,

here are some reference works:

http://www.ietf.org/internet-drafts/draft-veltri-sip-qsip-01.txt

http://www.coritel.it/projects/qsip/papers.htm

Regards,
Donald

----- Original Message -----
From: "hajer ferjani" <fer_hajer@yahoo.fr>
To: <sip@ietf.org>
Sent: Monday, May 12, 2003 4:21 AM
Subject: [Sip] integration of sip and cops


> hi all,
> anybody knows if there's some work on the integration
> of sip and cops protocols accordingly to the QoS
> architectures : IntServ and DiffServ.
> thanks for any help.
> regards.
>
> hajer.
>
>
>
>
> ___________________________________________________________
> Do You Yahoo!? -- Une adresse @yahoo.fr gratuite et en français !
> Yahoo! Mail : http://fr.mail.yahoo.com
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> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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Rohan,

The qsig2sip draft is silent on the subject of the contact header. So too is
the ISUP-SIP draft (except on the subject of contact headers in 3xx
response). It is not clear to me that a contact header should necessarily
reflect the identity of the remote party in the legacy network. Therefore
there may be no need to change the contact header when transfer occurs in
the legacy network.

Consider the following case. An INVITE request leaves the SIP network at an
egress gateway to QSIG. The gateway receives a QSIG ALERTING message and
sends back a 180 response containing a contact header. It would be possible
for this contact header to reflect the identity of the party to whom the
QSIG SETUP message was directed or, except where prevented by privacy, the
identity of the party actually being alerted (can be different because of
forwarding). Now, suppose after alerting but before answer the call is
forwarded, so that the party who answers is not the party identified in the
contact header in the 180 response. What should go in the contact header of
the 200 response: the same value that was in the 180 contact header; or a
different value reflecting the identity of the answering party? It is not
clear to me whether the contact header is allowed to or should change in
this situation.

Because if is not always feasible, because of privacy, forwarding etc. for
the contact header issued by a gateway always to reflect the identity of the
remote party in the legacy network, I see the contact header primarily as a
means of identifying the gateway, rather than the remote party. That is why
the qsig2sip draft is silent on this issue - the choice of contact header is
an implementation matter for the gateway.

Regards,

John

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: 13 May 2003 00:12
> To: Elwell, John
> Cc: 'Drage, Keith (Keith)'; 'sip@ietf.org'
> Subject: Re: [Sip] UPDATE and P-Asserted-ID
> 
> 
> John,
> 
> I believe you still should generate a new Contact header, even if you 
> are only changing the display name.
> 
> thanks,
> -rohan
> 
> On Thursday, May 8, 2003, at 01:21 AM, Elwell, John wrote:
> 
> > Keith,
> >
> > Thanks for your response. However, I do not wish to cover target 
> > refresh. As
> > I understand from section 12 of RFC3261, local and remote 
> targets are
> > derived from Contact headers. The scenario I have in mind is where 
> > something
> > happens behind a gateway. However, the gateway remains the same and
> > therefore it does not need to generate a new Contact header.
> >
> > John (john.elwell@siemens.com)
> >
> >> -----Original Message-----
> >> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> >> Sent: 07 May 2003 19:11
> >> To: Elwell, John; 'sip@ietf.org'
> >> Subject: RE: [Sip] UPDATE and P-Asserted-ID
> >>
> >>
> >> I believe the thinking when the document was developed was
> >> essentially to allow it in methods that either initiated a
> >> dialog, or were sent outside of a dialog. There was therefore
> >> no intent to cover reINVITE at that time, although I do not
> >> think it was explicitly excluded in the RFC. I am not sure
> >> how the BYE got added in respect of this, maybe someone else
> >> can clarify. The NOTIFY is there in respect of a NOTIFY that
> >> confirms a SUBSCRIBE dialog.
> >>
> >> This would allow the exchange of identities between the end
> >> users under these circumstances.
> >>
> >> From what you are indicating, it looks like you would want to
> >> extend the above to cover target refresh requests.
> >>
> >> Keith
> >>
> >> Keith Drage
> >> Lucent Technologies
> >> Tel: +44 1793 776249
> >> Email: drage@lucent.com
> >>
> >>> -----Original Message-----
> >>> From: Elwell, John [mailto:john.elwell@siemens.com]
> >>> Sent: 07 May 2003 10:20
> >>> To: 'sip@ietf.org'
> >>> Subject: [Sip] UPDATE and P-Asserted-ID
> >>>
> >>>
> >>> In RFC3325 section 9.1 it indicates that P-Asserted-ID is not
> >>> allowed in
> >>> UPDATE. Does anyone know the reason for this? A useful
> >>> application would be
> >>> to indicate a changed asserted identity when a change of
> >>> party occurs in a
> >>> legacy network (e.g., due to transfer). If not allowed in 
> UPDATE, a
> >>> re-INVITE would need to be used, but this introduces an
> >> extra message.
> >>>
> >>> John Elwell (john.elwell@siemens.com)
> >>>
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol
> >>> Use sip-implementors@cs.columbia.edu for questions on current sip
> >>> Use sipping@ietf.org for new developments on the 
> application of sip
> >>>
> >>
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP 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: Flemming Andreasen <fandreas@cisco.com>
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To: hajer ferjani <fer_hajer@yahoo.fr>
CC: sip@ietf.org
Subject: Re: [Sip] integration of sip and cops
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You may want to take a look at RFC 3313.

-- Flemming

hajer ferjani wrote:

> hi all,
> anybody knows if there's some work on the integration
> of sip and cops protocols accordingly to the QoS
> architectures : IntServ and DiffServ.
> thanks for any help.
> regards.
>
> hajer.
>
> ___________________________________________________________
> Do You Yahoo!? -- Une adresse @yahoo.fr gratuite et en français !
> Yahoo! Mail : http://fr.mail.yahoo.com
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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

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From: "Dean Willis" <dwillis@dynamicsoft.com>
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Date: Tue, 13 May 2003 20:10:08 -0500
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Subject: [Sip] Service route discovery draft revised
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Bernie and I revised the Service Route Discovery draft to align with the
current policy on example domain-name usage as requested by the AD. I also
added a small clarifier to the applicability statement as recently requested
on-list. This clarifier is "The service route constructed by the registrar
is the same for all contacts associated with a single address-of-record.
This mechanism does not provide for contact-specific service routes."

The draft has been submitted to internet-drafts, and in the interim is
available from:

http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-scvrtdisco-04.html
http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-scvrtdisco-04.txt
http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-scvrtdisco-04.xml


Note that this draft is in state "IESG Evalution -- AD Followup" according
to the Draft Tracker.

Barring having done something stupid in the processing of the changes (and
given the Blessing of the AD), I believe we're ready to proceed with this
draft. If I understand the process correctly, AD/IESG review the changes as
requested, and if acceptable the doc goes to the RFC editor.

--
Dean

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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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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		: Session Initiation Protocol Extension Header Field fo
                          Service Route Discovery During Registration
	Author(s)	: D. Willis, B. Hoeneisen
	Filename	: draft-ietf-sip-scvrtdisco-04.txt
	Pages		: 18
	Date		: 2003-5-14
	
This document defines a SIP extension header field used in
conjunction with responses to REGISTER requests to provide a
mechanism by which a registrar may inform a registering UA of a
service route that the UA may use to request outbound services from
the registrar's domain.

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'???? ???'" <thomas.lee@polypix.com>, <sip@ietf.org>
Subject: RE: [Sip] Push-To-Talk with SIP
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Rumor (I overheard people talking about it in a bar near a 3GPP meeting last
night) has it that there have been proposals to pursue this specification
work in OMA (www.openmobilealliance.org). Of course, we can't really verify
such rumors due to the closed IPR policy of the OMA. 
 
I CAN say that I've been encouraging OMA to open up.
 
--
Dean

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of ???? ???
Sent: Wednesday, May 14, 2003 8:47 PM
To: sip@ietf.org
Subject: [Sip] Push-To-Talk with SIP


Hello,
 
Nokia, Siemens and Ericsson have announced that they are developing a
push-to-talk application for GSM networks with GPRS and for future 3G
networks. They want to use SIP for signaling but I couldn't find any more
detailed information.
 
Does someone of you know some detail about the PTT signaling with SIP?
Do they care about arbitration if two people want to talk to the same group
at the same time? Are there some drafts or discussions about this topics?
 
regards,
thomas lee.
 
 


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<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D647475916-15052003>Rumor=20
(I overheard people talking about it in a bar near a 3GPP meeting last =
night)=20
has it that there have been proposals to pursue this specification work =
in OMA=20
(<A =
href=3D"http://www.openmobilealliance.org">www.openmobilealliance.org</A>=
). Of=20
course,&nbsp;we can't really verify such rumors due to the closed IPR =
policy of=20
the OMA. </SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D647475916-15052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D647475916-15052003>I CAN=20
say that I've been encouraging OMA to open up.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D647475916-15052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D647475916-15052003>--</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D647475916-15052003>Dean</SPAN></FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
  sip-admin@ietf.org [mailto:sip-admin@ietf.org] <B>On Behalf Of =
</B>????=20
  ???<BR><B>Sent:</B> Wednesday, May 14, 2003 8:47 PM<BR><B>To:</B>=20
  sip@ietf.org<BR><B>Subject:</B> [Sip] Push-To-Talk with=20
  SIP<BR><BR></FONT></DIV>
  <DIV><SPAN class=3D235004001-15052003><FONT =
size=3D2>Hello,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D235004001-15052003><FONT =
size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D235004001-15052003><FONT size=3D2>Nokia, Siemens =
and Ericsson=20
  have announced that they are developing a push-to-talk application for =
GSM=20
  networks with GPRS and for future 3G networks. They want to use SIP =
for=20
  signaling but I couldn't find any more detailed=20
  information.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D235004001-15052003><FONT =
size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D235004001-15052003><FONT size=3D2>Does someone of =
you know some=20
  detail about the PTT signaling with SIP?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D235004001-15052003><FONT size=3D2>Do they care =
about=20
  arbitration if two people want to talk to the same group at the same =
time? Are=20
  there some drafts or discussions about this =
topics?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D235004001-15052003><FONT =
size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D235004001-15052003><FONT =
size=3D2>regards,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D235004001-15052003><FONT size=3D2>thomas=20
  lee.</FONT></SPAN></DIV>
  <DIV><FONT size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV></BLOCKQUOTE></BODY></HTML>

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From: "Shane McCormack" <smccormack@tssg.org>
To: "Sipping@Ietf. Org" <sipping@ietf.org>, "Sip@Ietf. Org" <sip@ietf.org>
Date: Tue, 13 May 2003 17:09:48 +0100
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Subject: [Sip] PIDF format for spatial location
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Hi,

I have mailed the list before on this subject but I am still looking for a
bit more clarification on this subject. I need to retreive spatial location
in pidf format for work I am doing on my thesis.
I know that work is being done in the Geopriv working group on a format for
sending such information. Just as a matter of doing things correctly could
somebody with any knowledge help me in
by telling me the additional headers necessary, and any other thoughts so i
can put the correct information in my implementation and my thesis.

Thank you

Shane McCormack

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

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





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<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D713380516-13052003>Hi,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D713380516-13052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D713380516-13052003>I&nbsp;have mailed=20
the list before on this subject but I am still looking for&nbsp;a bit=20
more&nbsp;clarification on this subject. I need to retreive spatial =
location in=20
pidf format for work I am doing on my thesis.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D713380516-13052003>I know =
that work is=20
being done in the Geopriv working group on a format for sending such=20
information. Just as a matter of doing things correctly could somebody =
with any=20
knowledge help me in</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D713380516-13052003>by =
telling me the=20
additional headers necessary, and any other thoughts so i can put the =
correct=20
information in my implementation and my thesis. </SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D713380516-13052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D713380516-13052003>Thank=20
you</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D713380516-13052003></SPAN></FONT>&nbsp;</DIV><FONT size=3D2>
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-IE=20
style=3D"COLOR: black; mso-ansi-language: EN-IE"><FONT size=3D3>Shane=20
McCormack</FONT></SPAN></P>
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-IE=20
style=3D"COLOR: black; mso-ansi-language: EN-IE"><FONT size=3D3>Research =

Assistant,</FONT></SPAN><FONT size=3D3><SPAN style=3D"COLOR: =
black"><BR></SPAN><SPAN=20
lang=3DEN-IE style=3D"COLOR: black; mso-ansi-language: EN-IE">TSSG, =
Confederation=20
House,</SPAN></FONT><FONT size=3D3><SPAN style=3D"COLOR: black"> =
<BR></SPAN><SPAN=20
lang=3DEN-IE style=3D"COLOR: black; mso-ansi-language: EN-IE">Waterford =
Business=20
Park, Cork Road,</SPAN></FONT><FONT size=3D3><SPAN style=3D"COLOR: =
black">=20
<BR></SPAN><SPAN lang=3DEN-IE=20
style=3D"COLOR: black; mso-ansi-language: EN-IE">Waterford, Co.=20
Waterford.</SPAN><SPAN lang=3DEN-IE style=3D"COLOR: black"> </SPAN><SPAN =

style=3D"COLOR: black; mso-fareast-font-family: 'Arial Unicode MS'; =
mso-color-alt: windowtext"><?xml:namespace=20
prefix =3D o ns =3D "urn:schemas-microsoft-com:office:office"=20
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mso-ansi-language: EN-IE; mso-fareast-font-family: 'Times New Roman'; =
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Subject: [Sip] SIP and PAM
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Hi
Just wondering if there is any thought on using SIP with the PAM
specification. I know SIMPLE is obviously similar and better when using SIP
but PAM brings alot more with it than just presence. PAM as you know allows
for the retreival, from some sort of database system, of a users
preferences, lists of assigned agents, personal information etc. Can SIP
provide these sort of information retreival without implementing some sort
of PAM interface or is it a viable implementation to couple SIP/SIMPLE with
PAM.

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

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





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<DIV><FONT face=3DArial size=3D2><SPAN class=3D907024415-14052003>Hi=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D907024415-14052003>Just =
wondering if=20
there is any thought on using SIP with the PAM specification. I know =
SIMPLE is=20
obviously similar and better when using SIP but PAM brings alot more =
with it=20
than just presence. PAM as you know allows for the retreival, from some =
sort of=20
database system, of a users preferences, lists of assigned agents, =
personal=20
information etc. Can SIP provide these sort of information retreival =
without=20
implementing some sort of PAM interface or is it a viable implementation =
to=20
couple SIP/SIMPLE with PAM.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D907024415-14052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D907024415-14052003>Regards</SPAN></FONT></DIV>
<DIV><FONT size=3D2><SPAN lang=3DEN-IE=20
style=3D"COLOR: black; mso-ansi-language: EN-IE"><FONT size=3D3>Shane=20
McCormack</FONT></SPAN></DIV>
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-IE=20
style=3D"COLOR: black; mso-ansi-language: EN-IE"><FONT size=3D3>Research =

Assistant,</FONT></SPAN><FONT size=3D3><SPAN style=3D"COLOR: =
black"><BR></SPAN><SPAN=20
lang=3DEN-IE style=3D"COLOR: black; mso-ansi-language: EN-IE">TSSG, =
Confederation=20
House,</SPAN></FONT><FONT size=3D3><SPAN style=3D"COLOR: black"> =
<BR></SPAN><SPAN=20
lang=3DEN-IE style=3D"COLOR: black; mso-ansi-language: EN-IE">Waterford =
Business=20
Park, Cork Road,</SPAN></FONT><FONT size=3D3><SPAN style=3D"COLOR: =
black">=20
<BR></SPAN><SPAN lang=3DEN-IE=20
style=3D"COLOR: black; mso-ansi-language: EN-IE">Waterford, Co.=20
Waterford.</SPAN><SPAN lang=3DEN-IE style=3D"COLOR: black"> </SPAN><SPAN =

style=3D"COLOR: black; mso-fareast-font-family: 'Arial Unicode MS'; =
mso-color-alt: windowtext"><?xml:namespace=20
prefix =3D o ns =3D "urn:schemas-microsoft-com:office:office"=20
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+353 87 9955505</SPAN><SPAN=20
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From: "Mauvielle, Cedric" <Cedric.Mauvielle@eads-telecom.com>
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Hello

I want to know if some SIP/H323 software Gateways are available. We are
looking for partners and so we need name of compagnies which can work with
us.

So, if you know some, can you send me informations about :
- price
- technical overview (communications max, OS,...)
- if we can customize some paramaters (H323 ports for example)
- if we can test your system

Regards,

Cédric M.

--------------------------------------
EADS Defence and Security Network
Rue Jean Pierre Timbaud
Montigny le Bretonneux
78063 Saint Quentin en Yvelines Cedex
France

poste : +33 (0) 1-34-60-75-88
cedric.mauvielle@eads-telecom.com
---------------------------------------

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



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Subject: RE: [Sip] SIP and PAM
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From: "Goud, Venkat Ramana (Venkat) ** CTR **" <goud@avaya.com>
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Cc: "Shane McCormack" <smccormack@tssg.org>
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Most of the interfaces of PAM deal with data manipulation and retrieval and there is no straightforward SIP mapping to these methods.You can implement few of the PAM interfaces using SIMPLE but not many - but you will be breaking semantics!. 
There was a plan in the PAM forum to describe ways of mapping between SIP and PAM but I dont know the details.

Venkat

-----Original Message-----
From: Shane McCormack [mailto:smccormack@tssg.org]
Sent: Wednesday, May 14, 2003 11:49 AM
To: Sip@Ietf. Org
Subject: [Sip] SIP and PAM


Hi 
Just wondering if there is any thought on using SIP with the PAM specification. I know SIMPLE is obviously similar and better when using SIP but PAM brings alot more with it than just presence. PAM as you know allows for the retreival, from some sort of database system, of a users preferences, lists of assigned agents, personal information etc. Can SIP provide these sort of information retreival without implementing some sort of PAM interface or is it a viable implementation to couple SIP/SIMPLE with PAM.

Regards
Shane McCormack
Research Assistant,
TSSG, Confederation House, 
Waterford Business Park, Cork Road, 
Waterford, Co. Waterford. 
Phone: +353 51 302927     Fax: +353 51 302901 
Mobile: +353 87 9955505 
http://www.tssg.org 
MSN: mccormackshane@hotmail.com 
_______________________________________________
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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 mailnull@www1.ietf.org  Fri May 16 12:08:26 2003
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From: "Fred Robinson" <fred.robinson@sipfusion.com>
To: "Mauvielle, Cedric" <Cedric.Mauvielle@eads-telecom.com>, <sip@ietf.org>
Subject: RE: [Sip] SIP/H323 Gateways products
Date: Fri, 16 May 2003 09:57:43 -0600
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Cedric,
Check out the SHOUTip product from NET.com for SIP-H323 conversion;
http://www.net.com/products/voice_packet/



Fred Robinson
2855 Rock Creek Circle #131

Superior, CO 80027
fred.robinson@sipfusion.com
office: 720.304.8707
mobile: 303.881.7590



-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
Mauvielle, Cedric
Sent: Thursday, May 15, 2003 7:08 AM
To: 'sip@ietf.org'
Subject: [Sip] SIP/H323 Gateways products


Hello

I want to know if some SIP/H323 software Gateways are available. We are
looking for partners and so we need name of compagnies which can work with
us.

So, if you know some, can you send me informations about :
- price
- technical overview (communications max, OS,...)
- if we can customize some paramaters (H323 ports for example)
- if we can test your system

Regards,

Cédric M.

--------------------------------------
EADS Defence and Security Network
Rue Jean Pierre Timbaud
Montigny le Bretonneux
78063 Saint Quentin en Yvelines Cedex
France

poste : +33 (0) 1-34-60-75-88
cedric.mauvielle@eads-telecom.com
---------------------------------------

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

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



From mailnull@www1.ietf.org  Mon May 19 00:49:16 2003
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From: "Rajarshi Chakraborty" <rajarshi_chakraborty@hotmail.com>
To: drage@lucent.com, ashishn@mahindrabt.com, asanth01@sprintspectrum.com,
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Subject: [Sip] SIP mobility in WLAN
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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




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Date: Mon, 19 May 2003 04:03:07 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: "Elwell, John" <john.elwell@siemens.com>
CC: "'Rohan Mahy'" <rohan@cisco.com>,
        "'Drage, Keith (Keith)'" <drage@lucent.com>,
        "'sip@ietf.org'" <sip@ietf.org>
Subject: Re: [Sip] UPDATE and P-Asserted-ID
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I agree that Contact is not the right place to identify the user. 
Rather, it is a form of identification for a device.

There is really a bigger issue here, which is the scope of the 
information that can be updated by a target refresh request (of which 
UPDATE is one). RFC 3261 refers only to specific pieces of the state 
of a dialog which can be updated (cseq, contact headers). There are 
other pieces of state held for the lifetime of the dialog, such as the 
  UA capabilities (from the Supported header field), methods allowed 
(Allow header field), and so on. Can these be updated by a target 
refresh request?

RFC 3261 is silent on this point, and I think the question is similar 
to whether an implementation would even update the P-Asserted-ID 
header field in a re-INVITE (forgetting UPDATE even).

Our thinking was really that this information was static, and 
therefore would be set by the initial INVITE and not changed 
afterwards, although the spec never explicitly states this. The issue 
here is that folks believe P-Asserted-ID is NOT static (i.e., because 
of a call transfer), and therefore would need to change. I believe 
there are actually a few cases where it would change, such as various 
3pcc cases too.

So, considering the general problem, there are a few solutions:

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.

-Jonathan R.


Elwell, John wrote:
> Rohan,
> 
> The qsig2sip draft is silent on the subject of the contact header. So too is
> the ISUP-SIP draft (except on the subject of contact headers in 3xx
> response). It is not clear to me that a contact header should necessarily
> reflect the identity of the remote party in the legacy network. Therefore
> there may be no need to change the contact header when transfer occurs in
> the legacy network.
> 
> Consider the following case. An INVITE request leaves the SIP network at an
> egress gateway to QSIG. The gateway receives a QSIG ALERTING message and
> sends back a 180 response containing a contact header. It would be possible
> for this contact header to reflect the identity of the party to whom the
> QSIG SETUP message was directed or, except where prevented by privacy, the
> identity of the party actually being alerted (can be different because of
> forwarding). Now, suppose after alerting but before answer the call is
> forwarded, so that the party who answers is not the party identified in the
> contact header in the 180 response. What should go in the contact header of
> the 200 response: the same value that was in the 180 contact header; or a
> different value reflecting the identity of the answering party? It is not
> clear to me whether the contact header is allowed to or should change in
> this situation.
> 
> Because if is not always feasible, because of privacy, forwarding etc. for
> the contact header issued by a gateway always to reflect the identity of the
> remote party in the legacy network, I see the contact header primarily as a
> means of identifying the gateway, rather than the remote party. That is why
> the qsig2sip draft is silent on this issue - the choice of contact header is
> an implementation matter for the gateway.
> 
> Regards,
> 
> John
> 
> 
>>-----Original Message-----
>>From: Rohan Mahy [mailto:rohan@cisco.com]
>>Sent: 13 May 2003 00:12
>>To: Elwell, John
>>Cc: 'Drage, Keith (Keith)'; 'sip@ietf.org'
>>Subject: Re: [Sip] UPDATE and P-Asserted-ID
>>
>>
>>John,
>>
>>I believe you still should generate a new Contact header, even if you 
>>are only changing the display name.
>>
>>thanks,
>>-rohan
>>
>>On Thursday, May 8, 2003, at 01:21 AM, Elwell, John wrote:
>>
>>
>>>Keith,
>>>
>>>Thanks for your response. However, I do not wish to cover target 
>>>refresh. As
>>>I understand from section 12 of RFC3261, local and remote 
>>
>>targets are
>>
>>>derived from Contact headers. The scenario I have in mind is where 
>>>something
>>>happens behind a gateway. However, the gateway remains the same and
>>>therefore it does not need to generate a new Contact header.
>>>
>>>John (john.elwell@siemens.com)
>>>
>>>
>>>>-----Original Message-----
>>>>From: Drage, Keith (Keith) [mailto:drage@lucent.com]
>>>>Sent: 07 May 2003 19:11
>>>>To: Elwell, John; 'sip@ietf.org'
>>>>Subject: RE: [Sip] UPDATE and P-Asserted-ID
>>>>
>>>>
>>>>I believe the thinking when the document was developed was
>>>>essentially to allow it in methods that either initiated a
>>>>dialog, or were sent outside of a dialog. There was therefore
>>>>no intent to cover reINVITE at that time, although I do not
>>>>think it was explicitly excluded in the RFC. I am not sure
>>>>how the BYE got added in respect of this, maybe someone else
>>>>can clarify. The NOTIFY is there in respect of a NOTIFY that
>>>>confirms a SUBSCRIBE dialog.
>>>>
>>>>This would allow the exchange of identities between the end
>>>>users under these circumstances.
>>>>
>>>>From what you are indicating, it looks like you would want to
>>>>extend the above to cover target refresh requests.
>>>>
>>>>Keith
>>>>
>>>>Keith Drage
>>>>Lucent Technologies
>>>>Tel: +44 1793 776249
>>>>Email: drage@lucent.com
>>>>
>>>>
>>>>>-----Original Message-----
>>>>>From: Elwell, John [mailto:john.elwell@siemens.com]
>>>>>Sent: 07 May 2003 10:20
>>>>>To: 'sip@ietf.org'
>>>>>Subject: [Sip] UPDATE and P-Asserted-ID
>>>>>
>>>>>
>>>>>In RFC3325 section 9.1 it indicates that P-Asserted-ID is not
>>>>>allowed in
>>>>>UPDATE. Does anyone know the reason for this? A useful
>>>>>application would be
>>>>>to indicate a changed asserted identity when a change of
>>>>>party occurs in a
>>>>>legacy network (e.g., due to transfer). If not allowed in 
>>
>>UPDATE, a
>>
>>>>>re-INVITE would need to be used, but this introduces an
>>>>
>>>>extra message.
>>>>
>>>>>John Elwell (john.elwell@siemens.com)
>>>>>
>>>>>
>>>>>_______________________________________________
>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>This list is for NEW development of the core SIP Protocol
>>>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>>Use sipping@ietf.org for new developments on the 
>>
>>application of sip
>>
>>>_______________________________________________
>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>This list is for NEW development of the core SIP Protocol
>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>Use sipping@ietf.org for new developments on the application of sip
>>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
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

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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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CC: Sanjay Sinha <sanjsinh@cisco.com>, sip@ietf.org
Subject: Re: [Sip] Question on RFC 3263
References: <3EADA6A1.5010302@cisco.com> <3EAE8F5D.C3C1FE84@lmf.ericsson.se> <3EAE95E3.5000906@cisco.com> <3EAFA36A.5492523C@lmf.ericsson.se> <3EAFD79D.7070302@cisco.com> <3EB165FB.90FDF134@lmf.ericsson.se>
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Christer Holmberg wrote:
> Hi,
> 
> 
>>>>>> 1) Since ACK for 200 response indicates a new transaction
>>>>>> and follows rules of sec. 4.2 to find out the ip address
>>>>>> of next hop element to send ACK to, how does the client
>>>>>> transaction detect failure for ACK and move to next
>>>>>> record in the list if used with UDP and the next hop does
>>>>>> not generate transport errors.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>> 
>>>>> [CHH] It is true that the remote server will not be able to
>>>>> send eg a 503 response for an ACK, but if the failure
>>>>> occurs on the transport level you will be notified >it the
>>>>> same way as you would for any other request (ICMP errors
>>>>> etc). These are sent independently on if the remote server
>>>>> generates SIP responses or not.
>>>>> 
>>>>> 
>>>>> 
>>>> 
>>>> [SS] But the client transport layer will not get ICMP errors
>>>> if the first record from the list has crashed or has it's
>>>> interface shutdown.
>>>> 
>>>> 
>>> 
>>> [CHH] Well, if the ACK doesn't reach the server you will get
>>> retransmits of the 200 OK (unless they are supposed to be sent
>>> from the same interface which has crashed, of course)..
>>> 
>> 
>> [SS] Right, but how does the client's transaction figure out that
>> 200 retransmissions are because of Ack going to the wrong
>> interface from the list  vs. Ack being lost in the
>> network/network delay and whether it should cycle to the next
>> record or not.
> 
> 
> [CHH] I guess you will have to use logic based on local policy to
> decide when to "switch" from one address to another, if the 200
> responses don't stop coming, even if you send ACK...

I think that this is a bad idea. I believe the answer is simpler - 
there is no timeout-based mechanism for detecting failure for ACK. 
Only ICMP or TCP connection problems will work. Thats just a 
limitation of the way ACK works. Relying on e2e messaging (like the 
200 OK) to infer problems at a specific hop is not going to work.


> 
> 
>>>>> 2) For PRACK, if the client transaction layer times out
>>>>> waiting for 200 OK to PRACK before moving to the next
>>>>> record, the server transaction may run out of 18x retries
>>>>> waiting for PRACK and disconnect the call, if the next
>>>>> record happens to be the one that sent 18x.

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.

-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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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: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'Rohan Mahy'" <rohan@cisco.com>,
        "'Drage, Keith (Keith)'" <drage@lucent.com>,
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Jonathan,

Thanks for your input. I too would prefer 2.

John

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: 19 May 2003 09:03
> To: Elwell, John
> Cc: 'Rohan Mahy'; 'Drage, Keith (Keith)'; 'sip@ietf.org'
> Subject: Re: [Sip] UPDATE and P-Asserted-ID
> 
> 
> I agree that Contact is not the right place to identify the user. 
> Rather, it is a form of identification for a device.
> 
> There is really a bigger issue here, which is the scope of the 
> information that can be updated by a target refresh request (of which 
> UPDATE is one). RFC 3261 refers only to specific pieces of the state 
> of a dialog which can be updated (cseq, contact headers). There are 
> other pieces of state held for the lifetime of the dialog, 
> such as the 
>   UA capabilities (from the Supported header field), methods allowed 
> (Allow header field), and so on. Can these be updated by a target 
> refresh request?
> 
> RFC 3261 is silent on this point, and I think the question is similar 
> to whether an implementation would even update the P-Asserted-ID 
> header field in a re-INVITE (forgetting UPDATE even).
> 
> Our thinking was really that this information was static, and 
> therefore would be set by the initial INVITE and not changed 
> afterwards, although the spec never explicitly states this. The issue 
> here is that folks believe P-Asserted-ID is NOT static (i.e., because 
> of a call transfer), and therefore would need to change. I believe 
> there are actually a few cases where it would change, such as various 
> 3pcc cases too.
> 
> So, considering the general problem, there are a few solutions:
> 
> 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.
> 
> -Jonathan R.
> 
> 
> Elwell, John wrote:
> > Rohan,
> > 
> > The qsig2sip draft is silent on the subject of the contact 
> header. So too is
> > the ISUP-SIP draft (except on the subject of contact headers in 3xx
> > response). It is not clear to me that a contact header 
> should necessarily
> > reflect the identity of the remote party in the legacy 
> network. Therefore
> > there may be no need to change the contact header when 
> transfer occurs in
> > the legacy network.
> > 
> > Consider the following case. An INVITE request leaves the 
> SIP network at an
> > egress gateway to QSIG. The gateway receives a QSIG 
> ALERTING message and
> > sends back a 180 response containing a contact header. It 
> would be possible
> > for this contact header to reflect the identity of the 
> party to whom the
> > QSIG SETUP message was directed or, except where prevented 
> by privacy, the
> > identity of the party actually being alerted (can be 
> different because of
> > forwarding). Now, suppose after alerting but before answer 
> the call is
> > forwarded, so that the party who answers is not the party 
> identified in the
> > contact header in the 180 response. What should go in the 
> contact header of
> > the 200 response: the same value that was in the 180 
> contact header; or a
> > different value reflecting the identity of the answering 
> party? It is not
> > clear to me whether the contact header is allowed to or 
> should change in
> > this situation.
> > 
> > Because if is not always feasible, because of privacy, 
> forwarding etc. for
> > the contact header issued by a gateway always to reflect 
> the identity of the
> > remote party in the legacy network, I see the contact 
> header primarily as a
> > means of identifying the gateway, rather than the remote 
> party. That is why
> > the qsig2sip draft is silent on this issue - the choice of 
> contact header is
> > an implementation matter for the gateway.
> > 
> > Regards,
> > 
> > John
> > 
> > 
> >>-----Original Message-----
> >>From: Rohan Mahy [mailto:rohan@cisco.com]
> >>Sent: 13 May 2003 00:12
> >>To: Elwell, John
> >>Cc: 'Drage, Keith (Keith)'; 'sip@ietf.org'
> >>Subject: Re: [Sip] UPDATE and P-Asserted-ID
> >>
> >>
> >>John,
> >>
> >>I believe you still should generate a new Contact header, 
> even if you 
> >>are only changing the display name.
> >>
> >>thanks,
> >>-rohan
> >>
> >>On Thursday, May 8, 2003, at 01:21 AM, Elwell, John wrote:
> >>
> >>
> >>>Keith,
> >>>
> >>>Thanks for your response. However, I do not wish to cover target 
> >>>refresh. As
> >>>I understand from section 12 of RFC3261, local and remote 
> >>
> >>targets are
> >>
> >>>derived from Contact headers. The scenario I have in mind is where 
> >>>something
> >>>happens behind a gateway. However, the gateway remains the same and
> >>>therefore it does not need to generate a new Contact header.
> >>>
> >>>John (john.elwell@siemens.com)
> >>>
> >>>
> >>>>-----Original Message-----
> >>>>From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> >>>>Sent: 07 May 2003 19:11
> >>>>To: Elwell, John; 'sip@ietf.org'
> >>>>Subject: RE: [Sip] UPDATE and P-Asserted-ID
> >>>>
> >>>>
> >>>>I believe the thinking when the document was developed was
> >>>>essentially to allow it in methods that either initiated a
> >>>>dialog, or were sent outside of a dialog. There was therefore
> >>>>no intent to cover reINVITE at that time, although I do not
> >>>>think it was explicitly excluded in the RFC. I am not sure
> >>>>how the BYE got added in respect of this, maybe someone else
> >>>>can clarify. The NOTIFY is there in respect of a NOTIFY that
> >>>>confirms a SUBSCRIBE dialog.
> >>>>
> >>>>This would allow the exchange of identities between the end
> >>>>users under these circumstances.
> >>>>
> >>>>From what you are indicating, it looks like you would want to
> >>>>extend the above to cover target refresh requests.
> >>>>
> >>>>Keith
> >>>>
> >>>>Keith Drage
> >>>>Lucent Technologies
> >>>>Tel: +44 1793 776249
> >>>>Email: drage@lucent.com
> >>>>
> >>>>
> >>>>>-----Original Message-----
> >>>>>From: Elwell, John [mailto:john.elwell@siemens.com]
> >>>>>Sent: 07 May 2003 10:20
> >>>>>To: 'sip@ietf.org'
> >>>>>Subject: [Sip] UPDATE and P-Asserted-ID
> >>>>>
> >>>>>
> >>>>>In RFC3325 section 9.1 it indicates that P-Asserted-ID is not
> >>>>>allowed in
> >>>>>UPDATE. Does anyone know the reason for this? A useful
> >>>>>application would be
> >>>>>to indicate a changed asserted identity when a change of
> >>>>>party occurs in a
> >>>>>legacy network (e.g., due to transfer). If not allowed in 
> >>
> >>UPDATE, a
> >>
> >>>>>re-INVITE would need to be used, but this introduces an
> >>>>
> >>>>extra message.
> >>>>
> >>>>>John Elwell (john.elwell@siemens.com)
> >>>>>
> >>>>>
> >>>>>_______________________________________________
> >>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>>This list is for NEW development of the core SIP Protocol
> >>>>>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>>>>Use sipping@ietf.org for new developments on the 
> >>
> >>application of sip
> >>
> >>>_______________________________________________
> >>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>This list is for NEW development of the core SIP Protocol
> >>>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>>Use sipping@ietf.org for new developments on the application of sip
> >>
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> 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
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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] UPDATE and P-Asserted-ID
Date: Mon, 19 May 2003 14:45:31 +0300
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Option 2 is ok for me, but we have to soften the SHOULD requirement to add contact headers in requests within dialog (section 12.2) and that it SHOULD contain the same value as the previous request within dialog.

Regards,
Hisham

> -----Original Message-----
> From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Monday, May 19, 2003 11:03 AM
> To: Elwell, John
> Cc: 'Rohan Mahy'; 'Drage, Keith (Keith)'; 'sip@ietf.org'
> Subject: Re: [Sip] UPDATE and P-Asserted-ID
> 
> 
> I agree that Contact is not the right place to identify the user. 
> Rather, it is a form of identification for a device.
> 
> There is really a bigger issue here, which is the scope of the 
> information that can be updated by a target refresh request (of which 
> UPDATE is one). RFC 3261 refers only to specific pieces of the state 
> of a dialog which can be updated (cseq, contact headers). There are 
> other pieces of state held for the lifetime of the dialog, 
> such as the 
>   UA capabilities (from the Supported header field), methods allowed 
> (Allow header field), and so on. Can these be updated by a target 
> refresh request?
> 
> RFC 3261 is silent on this point, and I think the question is similar 
> to whether an implementation would even update the P-Asserted-ID 
> header field in a re-INVITE (forgetting UPDATE even).
> 
> Our thinking was really that this information was static, and 
> therefore would be set by the initial INVITE and not changed 
> afterwards, although the spec never explicitly states this. The issue 
> here is that folks believe P-Asserted-ID is NOT static (i.e., because 
> of a call transfer), and therefore would need to change. I believe 
> there are actually a few cases where it would change, such as various 
> 3pcc cases too.
> 
> So, considering the general problem, there are a few solutions:
> 
> 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.
> 
> -Jonathan R.
> 
> 
> Elwell, John wrote:
> > Rohan,
> > 
> > The qsig2sip draft is silent on the subject of the contact 
> header. So too is
> > the ISUP-SIP draft (except on the subject of contact headers in 3xx
> > response). It is not clear to me that a contact header 
> should necessarily
> > reflect the identity of the remote party in the legacy 
> network. Therefore
> > there may be no need to change the contact header when 
> transfer occurs in
> > the legacy network.
> > 
> > Consider the following case. An INVITE request leaves the 
> SIP network at an
> > egress gateway to QSIG. The gateway receives a QSIG 
> ALERTING message and
> > sends back a 180 response containing a contact header. It 
> would be possible
> > for this contact header to reflect the identity of the 
> party to whom the
> > QSIG SETUP message was directed or, except where prevented 
> by privacy, the
> > identity of the party actually being alerted (can be 
> different because of
> > forwarding). Now, suppose after alerting but before answer 
> the call is
> > forwarded, so that the party who answers is not the party 
> identified in the
> > contact header in the 180 response. What should go in the 
> contact header of
> > the 200 response: the same value that was in the 180 
> contact header; or a
> > different value reflecting the identity of the answering 
> party? It is not
> > clear to me whether the contact header is allowed to or 
> should change in
> > this situation.
> > 
> > Because if is not always feasible, because of privacy, 
> forwarding etc. for
> > the contact header issued by a gateway always to reflect 
> the identity of the
> > remote party in the legacy network, I see the contact 
> header primarily as a
> > means of identifying the gateway, rather than the remote 
> party. That is why
> > the qsig2sip draft is silent on this issue - the choice of 
> contact header is
> > an implementation matter for the gateway.
> > 
> > Regards,
> > 
> > John
> > 
> > 
> >>-----Original Message-----
> >>From: Rohan Mahy [mailto:rohan@cisco.com]
> >>Sent: 13 May 2003 00:12
> >>To: Elwell, John
> >>Cc: 'Drage, Keith (Keith)'; 'sip@ietf.org'
> >>Subject: Re: [Sip] UPDATE and P-Asserted-ID
> >>
> >>
> >>John,
> >>
> >>I believe you still should generate a new Contact header, 
> even if you 
> >>are only changing the display name.
> >>
> >>thanks,
> >>-rohan
> >>
> >>On Thursday, May 8, 2003, at 01:21 AM, Elwell, John wrote:
> >>
> >>
> >>>Keith,
> >>>
> >>>Thanks for your response. However, I do not wish to cover target 
> >>>refresh. As
> >>>I understand from section 12 of RFC3261, local and remote 
> >>
> >>targets are
> >>
> >>>derived from Contact headers. The scenario I have in mind is where 
> >>>something
> >>>happens behind a gateway. However, the gateway remains the same and
> >>>therefore it does not need to generate a new Contact header.
> >>>
> >>>John (john.elwell@siemens.com)
> >>>
> >>>
> >>>>-----Original Message-----
> >>>>From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> >>>>Sent: 07 May 2003 19:11
> >>>>To: Elwell, John; 'sip@ietf.org'
> >>>>Subject: RE: [Sip] UPDATE and P-Asserted-ID
> >>>>
> >>>>
> >>>>I believe the thinking when the document was developed was
> >>>>essentially to allow it in methods that either initiated a
> >>>>dialog, or were sent outside of a dialog. There was therefore
> >>>>no intent to cover reINVITE at that time, although I do not
> >>>>think it was explicitly excluded in the RFC. I am not sure
> >>>>how the BYE got added in respect of this, maybe someone else
> >>>>can clarify. The NOTIFY is there in respect of a NOTIFY that
> >>>>confirms a SUBSCRIBE dialog.
> >>>>
> >>>>This would allow the exchange of identities between the end
> >>>>users under these circumstances.
> >>>>
> >>>>From what you are indicating, it looks like you would want to
> >>>>extend the above to cover target refresh requests.
> >>>>
> >>>>Keith
> >>>>
> >>>>Keith Drage
> >>>>Lucent Technologies
> >>>>Tel: +44 1793 776249
> >>>>Email: drage@lucent.com
> >>>>
> >>>>
> >>>>>-----Original Message-----
> >>>>>From: Elwell, John [mailto:john.elwell@siemens.com]
> >>>>>Sent: 07 May 2003 10:20
> >>>>>To: 'sip@ietf.org'
> >>>>>Subject: [Sip] UPDATE and P-Asserted-ID
> >>>>>
> >>>>>
> >>>>>In RFC3325 section 9.1 it indicates that P-Asserted-ID is not
> >>>>>allowed in
> >>>>>UPDATE. Does anyone know the reason for this? A useful
> >>>>>application would be
> >>>>>to indicate a changed asserted identity when a change of
> >>>>>party occurs in a
> >>>>>legacy network (e.g., due to transfer). If not allowed in 
> >>
> >>UPDATE, a
> >>
> >>>>>re-INVITE would need to be used, but this introduces an
> >>>>
> >>>>extra message.
> >>>>
> >>>>>John Elwell (john.elwell@siemens.com)
> >>>>>
> >>>>>
> >>>>>_______________________________________________
> >>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>>This list is for NEW development of the core SIP Protocol
> >>>>>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>>>>Use sipping@ietf.org for new developments on the 
> >>
> >>application of sip
> >>
> >>>_______________________________________________
> >>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>This list is for NEW development of the core SIP Protocol
> >>>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>>Use sipping@ietf.org for new developments on the application of sip
> >>
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> 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
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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] UPDATE and P-Asserted-ID
Date: Mon, 19 May 2003 08:57:13 -0600
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From: "Hearty, John" <John.Hearty@Level3.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Elwell, John" <john.elwell@siemens.com>
Cc: "Rohan Mahy" <rohan@cisco.com>, "Drage, Keith (Keith)" <drage@lucent.com>,
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I'd be interested in seeing all "such headers" listed, and consider security implications carefully before covering them with a blanket rule.


> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Monday, May 19, 2003 2:03 AM
> To: Elwell, John
> Cc: 'Rohan Mahy'; 'Drage, Keith (Keith)'; 'sip@ietf.org'
> Subject: Re: [Sip] UPDATE and P-Asserted-ID
> 
> 
> I agree that Contact is not the right place to identify the user. 
> Rather, it is a form of identification for a device.
> 
> There is really a bigger issue here, which is the scope of the 
> information that can be updated by a target refresh request (of which 
> UPDATE is one). RFC 3261 refers only to specific pieces of the state 
> of a dialog which can be updated (cseq, contact headers). There are 
> other pieces of state held for the lifetime of the dialog, 
> such as the 
>   UA capabilities (from the Supported header field), methods allowed 
> (Allow header field), and so on. Can these be updated by a target 
> refresh request?
> 
> RFC 3261 is silent on this point, and I think the question is similar 
> to whether an implementation would even update the P-Asserted-ID 
> header field in a re-INVITE (forgetting UPDATE even).
> 
> Our thinking was really that this information was static, and 
> therefore would be set by the initial INVITE and not changed 
> afterwards, although the spec never explicitly states this. The issue 
> here is that folks believe P-Asserted-ID is NOT static (i.e., because 
> of a call transfer), and therefore would need to change. I believe 
> there are actually a few cases where it would change, such as various 
> 3pcc cases too.
> 
> So, considering the general problem, there are a few solutions:
> 
> 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.
> 
> -Jonathan R.
> 
> 
> Elwell, John wrote:
> > Rohan,
> > 
> > The qsig2sip draft is silent on the subject of the contact 
> header. So too is
> > the ISUP-SIP draft (except on the subject of contact headers in 3xx
> > response). It is not clear to me that a contact header 
> should necessarily
> > reflect the identity of the remote party in the legacy 
> network. Therefore
> > there may be no need to change the contact header when 
> transfer occurs in
> > the legacy network.
> > 
> > Consider the following case. An INVITE request leaves the 
> SIP network at an
> > egress gateway to QSIG. The gateway receives a QSIG 
> ALERTING message and
> > sends back a 180 response containing a contact header. It 
> would be possible
> > for this contact header to reflect the identity of the 
> party to whom the
> > QSIG SETUP message was directed or, except where prevented 
> by privacy, the
> > identity of the party actually being alerted (can be 
> different because of
> > forwarding). Now, suppose after alerting but before answer 
> the call is
> > forwarded, so that the party who answers is not the party 
> identified in the
> > contact header in the 180 response. What should go in the 
> contact header of
> > the 200 response: the same value that was in the 180 
> contact header; or a
> > different value reflecting the identity of the answering 
> party? It is not
> > clear to me whether the contact header is allowed to or 
> should change in
> > this situation.
> > 
> > Because if is not always feasible, because of privacy, 
> forwarding etc. for
> > the contact header issued by a gateway always to reflect 
> the identity of the
> > remote party in the legacy network, I see the contact 
> header primarily as a
> > means of identifying the gateway, rather than the remote 
> party. That is why
> > the qsig2sip draft is silent on this issue - the choice of 
> contact header is
> > an implementation matter for the gateway.
> > 
> > Regards,
> > 
> > John
> > 
> > 
> >>-----Original Message-----
> >>From: Rohan Mahy [mailto:rohan@cisco.com]
> >>Sent: 13 May 2003 00:12
> >>To: Elwell, John
> >>Cc: 'Drage, Keith (Keith)'; 'sip@ietf.org'
> >>Subject: Re: [Sip] UPDATE and P-Asserted-ID
> >>
> >>
> >>John,
> >>
> >>I believe you still should generate a new Contact header, 
> even if you 
> >>are only changing the display name.
> >>
> >>thanks,
> >>-rohan
> >>
> >>On Thursday, May 8, 2003, at 01:21 AM, Elwell, John wrote:
> >>
> >>
> >>>Keith,
> >>>
> >>>Thanks for your response. However, I do not wish to cover target 
> >>>refresh. As
> >>>I understand from section 12 of RFC3261, local and remote 
> >>
> >>targets are
> >>
> >>>derived from Contact headers. The scenario I have in mind is where 
> >>>something
> >>>happens behind a gateway. However, the gateway remains the same and
> >>>therefore it does not need to generate a new Contact header.
> >>>
> >>>John (john.elwell@siemens.com)
> >>>
> >>>
> >>>>-----Original Message-----
> >>>>From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> >>>>Sent: 07 May 2003 19:11
> >>>>To: Elwell, John; 'sip@ietf.org'
> >>>>Subject: RE: [Sip] UPDATE and P-Asserted-ID
> >>>>
> >>>>
> >>>>I believe the thinking when the document was developed was
> >>>>essentially to allow it in methods that either initiated a
> >>>>dialog, or were sent outside of a dialog. There was therefore
> >>>>no intent to cover reINVITE at that time, although I do not
> >>>>think it was explicitly excluded in the RFC. I am not sure
> >>>>how the BYE got added in respect of this, maybe someone else
> >>>>can clarify. The NOTIFY is there in respect of a NOTIFY that
> >>>>confirms a SUBSCRIBE dialog.
> >>>>
> >>>>This would allow the exchange of identities between the end
> >>>>users under these circumstances.
> >>>>
> >>>>From what you are indicating, it looks like you would want to
> >>>>extend the above to cover target refresh requests.
> >>>>
> >>>>Keith
> >>>>
> >>>>Keith Drage
> >>>>Lucent Technologies
> >>>>Tel: +44 1793 776249
> >>>>Email: drage@lucent.com
> >>>>
> >>>>
> >>>>>-----Original Message-----
> >>>>>From: Elwell, John [mailto:john.elwell@siemens.com]
> >>>>>Sent: 07 May 2003 10:20
> >>>>>To: 'sip@ietf.org'
> >>>>>Subject: [Sip] UPDATE and P-Asserted-ID
> >>>>>
> >>>>>
> >>>>>In RFC3325 section 9.1 it indicates that P-Asserted-ID is not
> >>>>>allowed in
> >>>>>UPDATE. Does anyone know the reason for this? A useful
> >>>>>application would be
> >>>>>to indicate a changed asserted identity when a change of
> >>>>>party occurs in a
> >>>>>legacy network (e.g., due to transfer). If not allowed in 
> >>
> >>UPDATE, a
> >>
> >>>>>re-INVITE would need to be used, but this introduces an
> >>>>
> >>>>extra message.
> >>>>
> >>>>>John Elwell (john.elwell@siemens.com)
> >>>>>
> >>>>>
> >>>>>_______________________________________________
> >>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>>This list is for NEW development of the core SIP Protocol
> >>>>>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>>>>Use sipping@ietf.org for new developments on the 
> >>
> >>application of sip
> >>
> >>>_______________________________________________
> >>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>This list is for NEW development of the core SIP Protocol
> >>>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>>Use sipping@ietf.org for new developments on the application of sip
> >>
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> 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
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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Thread-Topic: SIP mobility in WLAN
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From: "Santharam, Arun [NTWK SVCS]" <asanth01@sprintspectrum.com>
To: "Rajarshi Chakraborty" <rajarshi_chakraborty@hotmail.com>,
        <drage@lucent.com>, <ashishn@mahindrabt.com>, <sip@ietf.org>
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Subject: [Sip] RE: SIP mobility in WLAN
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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
> 
> 
> 
> 
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Santharam, Arun [NTWK SVCS] writes:
 > 
 > 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.

I think it's a little bit more complicated than
this. As with most things, it often makes sense to
have logic at many different layers depending on
the actual situation you find yourself in. For
example, since there isn't route optimization for
IPv4, a reinvite with the new correct CoA would
rid yourself of the dog-leg through the home
agent. But it might complicate QoS reservations
and/or treatment, etc.

Which is a round about way of saying "it depends".

	 Mike
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From: Michael Thomas <mat@cisco.com>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: "Elwell, John" <john.elwell@siemens.com>, "'Rohan Mahy'" <rohan@cisco.com>,
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Subject: Re: [Sip] UPDATE and P-Asserted-ID
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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
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Subject: [Sip] Out of sequence datagrams
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Hi,
Please refer to the snippet from "12.2.2 UAS Behavior" of RFC3261

....
 If the remote sequence number is empty, it MUST be set to the value
   of the sequence number in the CSeq header field value in the request.
   If the remote sequence number was not empty, but the sequence number
   of the request is lower than the remote sequence number, the request
   is out of order 
....

Because the ordering of UDP datagrams isn't guarenteed, it is possible for
the above to occur. For example NOTIFY with CSEQ 2 is received after the BYE
with CSeq 3. Since, from the application perpective, the dialog has ended,
there appears no way the NOTIFY can be processed by the UAS. Any ideas much
appreciated.

Fred O'Leary
MTS Sylantro Systems Corp.
(408)626-3040

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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Out of sequence datagrams
Date: Tue, 20 May 2003 15:11:45 -0400
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> Please refer to the snippet from 
> "12.2.2 UAS Behavior" of RFC3261
> 
> ....
>  If the remote sequence number is empty, 
>  it MUST be set to the value of the sequence 
>  number in the CSeq header field value in 
>  the request.
>  If the remote sequence number was not empty, 
>  but the sequence number of the request is lower 
>  than the remote sequence number, the request
>  is out of order 
> ....
> 
> Because the ordering of UDP datagrams isn't 
> guarenteed, it is possible for the above to occur. 
> For example NOTIFY with CSEQ 2 is received after 
> the BYE with CSeq 3. Since, from the application 
> perpective, the dialog has ended, there appears no 
> way the NOTIFY can be processed by the UAS. 

If order is important, requests should
be queued until the prior request has been 
acknowledged.  However concerning your BYE
and NOTIFY example, the BYE only terminates
the call associated with the dialog.  The
BYE does not necessarily terminate subscriptions 
associated with the dialog.  Thus if the NOTIFY
relates to the implicit subscription caused
by a REFER, the notifies can continue to be
sent after the BYE has released the call
associated with the dialog.

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Date: Tue, 20 May 2003 13:01:47 -0700
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Cc: "'Dean Willis'" <dean.willis@softarmor.com>, jon.peterson@neustar.biz,
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        "'Robert Sparks'" <rsparks@dynamicsoft.com>, rohan@cisco.com,
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Subject: [Sip] WGLC for event-list extension
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Hello Everyone,

The SIMPLE Working Group already issued a last call for this, but the 
SIP Working Group has been asked to issue a short working group last 
call for this extension as well, since it contains an option-tag (the 
tokens used in the Require and Supported headers).

So, I would like to begin an abbreviated last call in SIP for:

http://www.ietf.org/internet-drafts/draft-ietf-simple-event-list-03.txt

This WGLC ends June 6, 2003

thanks,
-rohan

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From: "Fred O'Leary" <Fred.Oleary@sylantro.com>
To: brett@broadsoft.com, sip@ietf.org
Subject: RE: [Sip] Out of sequence datagrams
Date: Tue, 20 May 2003 14:16:15 -0700
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This is what suspected. Yes the NOTIFY is a result of the implicit
SUBSCRIPTION caused by the REFER. So from an implementation perspective,
what is/should be the CSEQ relationship between NOTIFY CSEQ and the INVITE
dialog? (If there is none, the UAS would need to differentiate NOTIFY CSEQ's
from the INVITE/BYE even though the call identifers are the same.)

-----Original Message-----
From: Brett Tate [mailto:brett@broadsoft.com]
Sent: Tuesday, May 20, 2003 12:12 PM
To: sip@ietf.org
Subject: RE: [Sip] Out of sequence datagrams


> Please refer to the snippet from 
> "12.2.2 UAS Behavior" of RFC3261
> 
> ....
>  If the remote sequence number is empty, 
>  it MUST be set to the value of the sequence 
>  number in the CSeq header field value in 
>  the request.
>  If the remote sequence number was not empty, 
>  but the sequence number of the request is lower 
>  than the remote sequence number, the request
>  is out of order 
> ....
> 
> Because the ordering of UDP datagrams isn't 
> guarenteed, it is possible for the above to occur. 
> For example NOTIFY with CSEQ 2 is received after 
> the BYE with CSeq 3. Since, from the application 
> perpective, the dialog has ended, there appears no 
> way the NOTIFY can be processed by the UAS. 

If order is important, requests should
be queued until the prior request has been 
acknowledged.  However concerning your BYE
and NOTIFY example, the BYE only terminates
the call associated with the dialog.  The
BYE does not necessarily terminate subscriptions 
associated with the dialog.  Thus if the NOTIFY
relates to the implicit subscription caused
by a REFER, the notifies can continue to be
sent after the BYE has released the call
associated with the dialog.

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

_______________________________________________
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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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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Out of sequence datagrams
Date: Tue, 20 May 2003 18:57:48 -0400
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> This is what suspected. Yes the NOTIFY is 
> a result of the implicit SUBSCRIPTION caused 
> by the REFER. So from an implementation 
> perspective, what is/should be the CSEQ relationship 
> between NOTIFY CSEQ and the INVITE dialog? 

They subscription and call share the same 
dialog.  Thus the cseq must continue to 
increment upon each sent request.

Your prior example of a NOTIFY with cseq
value 2 being received after a BYE with
cseq value 3 could produce several results.

If the UAS does not care about refer related
notifies after processing a BYE, it could
return a 481 to avoid a resend or future
notifies.

It the notify contained a subscription-state 
of terminated, the UAS could return a 200,
481, or 500.  If the 500 is returned, the
UAC might not actually bump the cseq and
resend the request; thus it might lead to
a delay in subscription cleanup on the referrer.

If the UAS does still care about the notifies
and subscription-state is not terminated, 
it should return a 500 response to inform 
the UAC that it was received out of order.  
Since the 500 isn't only used for incorrectly 
order cseq, adding a retry-after header with
a small value communicates that the UAS would 
like the UAC to try again if needed.

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From: "Sumit Arora , Noida" <sumit@acmet.com>
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Date: Fri, 16 May 2003 18:56:33 +0530
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Subject: [Sip] Doubt in Sec-8.2 ,rfc-3261(SIP)
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    Dear sir,
 
     I am having doubt on these lines from rfc-3261 ,
     Sec-8.2, UAS behavior
 
   
  " Note that request processing is atomic.  If a request is accepted,
    all state changes associated with it MUST be performed.  If it is
    rejected, all state changes MUST NOT be performed."
 
   Is this above MUST and MUST not requirement , Talking 
   about  the state machine in Section - 17.2 Server Transaction.
 
   Kindly reply.
 
  Thanx
  -Sumit Arora -

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Hi all, please:

 

How is used SIP for applications of chats?

 

How I can be informed on the matter?

 

Thank you 

 

Manuel


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DES-TRAD =
style=3D'font-size:
10.0pt;font-family:Arial'>Hi all, please:</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>How is used SIP for applications of =
chats?</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>How I can be informed on the =
matter?</span></font></p>

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

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

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

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

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From mailnull@www1.ietf.org  Wed May 21 09:53:56 2003
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Subject: [Sip] TCP timeout
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Hi,

I was wondering how to handle TCP timeouts with SIP.
If a connection is opened to a _non existing_ destination, the TCP stack 
(using non-blocking sockets and select) will only inform about an error 
after a TCP timeout (that can be considerably long on some systems).
(Existing destination with no listening port usually facilitates ICMP and 
"tells" that there is no listener).

My understanding is that SIP handles that by implementing invite timeout 
that is T1*64 (which is also quite long).

Here are the issues I'm struggling with:
1 - Does SIP has any other mechanism to handle such scenarios (H323 
facilitates a 4 seconds timeout on every outgoing call)?
2 - Doesn't this scenario compromises part of rfc3263 - locating sip 
servers, because the user is not likely to wait that long until a second 
server is reached.

Thanks,
James.

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Subject: [Sip] WLAN & SIP
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Is there any draft on SIP usage in WLAN arena ?
Any help would be appreciated.

Prabhas
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Subject: Re: [Sip] WGLC for auth-id body
From: Cullen Jennings <fluffy@cisco.com>
To: sip-ietf <sip@ietf.org>, Jon Peterson <jon.peterson@neustar.biz>
CC: Dean Willis <dean.willis@softarmor.com>, <Gonzalo.Camarillo@ericsson.com>,
        Rohan Mahy <rohan@cisco.com>
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I have a concern about SIP security (I've always wanted to start and email
that way :-) 

If A sends a request to B and B puts an AIB in the response, I think the
response AIB should have a To: B and not have a From: B. The reason is
fairly simple - Imagine A is an attacker and is watching some call B is
having with C. A wants to insert a message into this call flow - It knows
the Call-ID and appropriate tags and such. A can guess a CSeq that will
likely work. Now A sends a request to B with all this stuff and B sends A a
signed AIB that A can now use to send to C and impersonate B.

On the minor NITs level ...

It would be nice to add some text on tags in To From and if they should be
inserted or not and if  they must match.

Use a binary transfer encoding instead of base64 in all of the examples. I
know this is a pain in the RFC but I think it is what we should use in SIP.

Point out that detached signatures SHOULD be used.

Clarify if the signature is computed over

    ---------
    Content-Type: ...
    Content-Disposition: ...

    From:...
    To:....   
    -------

Or over 

    ---------
    From:...
    To:....   
    -------

The outer SIP message in example 3 does not have a Content-Length.

I like the draft.
 


On 5/13/03 5:39 PM, "Rohan Mahy" <rohan@cisco.com> 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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From: =?iso-8859-1?Q?Jos=E9_Soler_Lucas?= <jsoler@com.dtu.dk>
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Subject: [Sip] Doubt about SIP Specific Event Notification behaviour
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------_=_NextPart_001_01C32060.0BEA9CA0
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Hi everyone,
=20
trying to imagine a service based on the SUBSCRIBE - NOTIFY methods of =
SIP
described in RFC 3265 I have came across a doubt that I was not able to =
find
a respose to through the different documents I have checked. Could =
anyone
out there please give me any hint on the matter?:
=20
Imagine I have a SIP client that subscribes to a certain event =
happening in
a server. The subscription is acepted and performs properly. After a =
moment,
while the subscription is still active, there is an internal problem in =
the
server that prevents further NOTIFYes towards the subscribing client. =
The
question is:
=20
a) Does the server send a 5xx message towards the client? or
b) Does the server send a NOTIFY with a "pending" status indicating a
temporal suspension of the subscription? or
c) Does the server send a NOTIFY with a "terminated" status forcing the =
end
of the subscription?
d) Should the server send rather [5xx] followed by
[NOTIFY(status=3Dpending/terminated)]?=20
=20
If the answer is (a), what should the behavior of the client then be, =
to
assume the subscription as terminated?
=20
Thanks in advance for your help.
Best regards,
=20
jos=E9

--------------------------------------------=20
Jos=E9 Soler Lucas, M.Sc.Telecom.E.=20
PhD Student=20
Networks Competence Area=20
Research Center COM (Communications, Optics and Materials)=20
Technical University of Denmark=20
Building 343, DK 2800 Kgs. Lyngby, Denmark.=20
Phone: +45 45 25 38 08 (direct)=20
Fax:   +45 45 93 65 81=20
E-mail: jsoler@com.dtu.dk=20
Web:  <http://www.com.dtu.dk/> http://www.com.dtu.dk/=20
----=20
Cellular: +45 22 27 48 22=20
--------------------------------------------=20



=20

------_=_NextPart_001_01C32060.0BEA9CA0
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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">


<META content=3D"MSHTML 6.00.2800.1141" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>Hi=20
everyone,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D031112212-22052003>trying to imagine a=20
service based on the SUBSCRIBE - NOTIFY methods of SIP described in RFC =
3265 I=20
have came across a doubt that I was not able to find a respose to =
through the=20
different documents I have checked. Could anyone out there please give =
me any=20
hint on the matter?:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D031112212-22052003>Imagine I have a SIP=20
client that subscribes to a certain event happening in a server. The=20
subscription is acepted and performs properly. After a moment, while =
the=20
subscription is still active, there is an internal problem in the =
server that=20
prevents further NOTIFYes towards the subscribing client. The question=20
is:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>a) =
Does the server=20
send a 5xx message towards the client? or</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>b) =
Does the server=20
send a NOTIFY with a "pending" status indicating a temporal suspension =
of the=20
subscription? or</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>c) =
Does the server=20
send a NOTIFY with a "terminated" status forcing the end of the=20
subscription?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>d) =
Should the server=20
send rather [5xx] followed by=20
[NOTIFY(status=3Dpending/terminated)]?&nbsp;</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>If =
the answer is=20
(a), what should the behavior of the client then be, to assume the =
subscription=20
as terminated?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D031112212-22052003>Thanks in advance=20
for your help.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>Best=20
regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D031112212-22052003>jos=E9</SPAN></FONT></DIV>
<P><FONT face=3D"Courier New"><FONT=20
size=3D2>--------------------------------------------</FONT></FONT> =
<BR><FONT=20
face=3D"Courier New" size=3D2>Jos=E9 Soler Lucas, =
M.Sc.Telecom.E.</FONT> <BR><FONT=20
face=3D"Courier New" size=3D2>PhD Student</FONT> <BR><FONT =
face=3D"Courier New"=20
size=3D2>Networks Competence Area</FONT> <BR><FONT face=3D"Courier New" =

size=3D2>Research Center COM (Communications, Optics and =
Materials)</FONT>=20
<BR><FONT face=3D"Courier New" size=3D2>Technical University of =
Denmark</FONT>=20
<BR><FONT face=3D"Courier New" size=3D2>Building 343, DK 2800 Kgs. =
Lyngby,=20
Denmark.</FONT> <BR><FONT face=3D"Courier New" size=3D2>Phone: +45 45 =
25 38 08=20
(direct)</FONT> <BR><FONT face=3D"Courier New" =
size=3D2>Fax:&nbsp;&nbsp; +45 45 93=20
65 81</FONT> <BR><FONT face=3D"Courier New" size=3D2>E-mail:=20
jsoler@com.dtu.dk</FONT> <BR><FONT face=3D"Courier New" size=3D2>Web: =
<A=20
href=3D"http://www.com.dtu.dk/" target=3D_blank><FONT=20
color=3D#000000>http://www.com.dtu.dk/</FONT></A></FONT> <BR><FONT=20
face=3D"Courier New" size=3D2>----</FONT> <BR><FONT face=3D"Courier =
New"=20
size=3D2>Cellular: +45 22 27 48 22</FONT> <BR><FONT face=3D"Courier =
New"=20
size=3D2>--------------------------------------------</FONT> =
</P><BR><BR>
<DIV>&nbsp;</DIV></BODY></HTML>

------_=_NextPart_001_01C32060.0BEA9CA0--
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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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 Sorry. I made a wrong formulation of the question. It should be like =
that:
=20
"...internal problem in the server that prevents to have knowledge of =
the
events the client is interested on and so notify about them properly to =
the
client despite the event is happening. The question is...."
=20
Thanks and sorry for the duplication,
=20
jose

-----Original Message-----
From: Jos=E9 Soler Lucas=20
Sent: 22. maj 2003 14:46
To: 'sip@ietf.org'
Subject: Doubt about SIP Specific Event Notification behaviour


Hi everyone,
=20
trying to imagine a service based on the SUBSCRIBE - NOTIFY methods of =
SIP
described in RFC 3265 I have came across a doubt that I was not able to =
find
a respose to through the different documents I have checked. Could =
anyone
out there please give me any hint on the matter?:
=20
Imagine I have a SIP client that subscribes to a certain event =
happening in
a server. The subscription is acepted and performs properly. After a =
moment,
while the subscription is still active, there is an internal problem in =
the
server that prevents further NOTIFYes towards the subscribing client. =
The
question is:
=20
a) Does the server send a 5xx message towards the client? or
b) Does the server send a NOTIFY with a "pending" status indicating a
temporal suspension of the subscription? or
c) Does the server send a NOTIFY with a "terminated" status forcing the =
end
of the subscription?
d) Should the server send rather [5xx] followed by
[NOTIFY(status=3Dpending/terminated)]?=20
=20
If the answer is (a), what should the behavior of the client then be, =
to
assume the subscription as terminated?
=20
Thanks in advance for your help.
Best regards,
=20
jos=E9

--------------------------------------------=20
Jos=E9 Soler Lucas, M.Sc.Telecom.E.=20
PhD Student=20
Networks Competence Area=20
Research Center COM (Communications, Optics and Materials)=20
Technical University of Denmark=20
Building 343, DK 2800 Kgs. Lyngby, Denmark.=20
Phone: +45 45 25 38 08 (direct)=20
Fax:   +45 45 93 65 81=20
E-mail: jsoler@com.dtu.dk=20
Web:  <http://www.com.dtu.dk/> http://www.com.dtu.dk/=20
----=20
Cellular: +45 22 27 48 22=20
--------------------------------------------=20



=20


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charset=3Diso-8859-1">


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<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D734045112-22052003>&nbsp;Sorry. I made a wrong formulation of =
the=20
question. It should be like that:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D734045112-22052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D734045112-22052003>"...internal problem in the server that =
prevents to=20
have knowledge of the events the client is interested on and so notify =
about=20
them properly to the client despite the event is happening. The =
question=20
is...."</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D734045112-22052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D734045112-22052003>Thanks=20
and sorry for the duplication,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D734045112-22052003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D734045112-22052003>jose</SPAN></FONT></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Jos=E9 Soler =
Lucas=20
  <BR><B>Sent:</B> 22. maj 2003 14:46<BR><B>To:</B>=20
  'sip@ietf.org'<BR><B>Subject:</B> Doubt about SIP Specific Event =
Notification=20
  behaviour<BR><BR></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>Hi=20
  everyone,</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D031112212-22052003>trying to imagine=20
  a service based on the SUBSCRIBE - NOTIFY methods of SIP described in =
RFC 3265=20
  I have came across a doubt that I was not able to find a respose to =
through=20
  the different documents I have checked. Could anyone out there please =
give me=20
  any hint on the matter?:</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D031112212-22052003>Imagine I have a=20
  SIP client that subscribes to a certain event happening in a server. =
The=20
  subscription is acepted and performs properly. After a moment, while =
the=20
  subscription is still active, there is an internal problem in the =
server that=20
  prevents further NOTIFYes towards the subscribing client. The =
question=20
  is:</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>a) =
Does the server=20
  send a 5xx message towards the client? or</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>b) =
Does the server=20
  send a NOTIFY with a "pending" status indicating a temporal =
suspension of the=20
  subscription? or</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>c) =
Does the server=20
  send a NOTIFY with a "terminated" status forcing the end of the=20
  subscription?</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>d) =
Should the=20
  server send rather [5xx] followed by=20
  [NOTIFY(status=3Dpending/terminated)]?&nbsp;</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D031112212-22052003>If =
the answer is=20
  (a), what should the behavior of the client then be, to assume the=20
  subscription as terminated?</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D031112212-22052003>Thanks in advance=20
  for your help.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D031112212-22052003>Best=20
  regards,</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D031112212-22052003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D031112212-22052003>jos=E9</SPAN></FONT></DIV>
  <P><FONT face=3D"Courier New"><FONT=20
  size=3D2>--------------------------------------------</FONT></FONT> =
<BR><FONT=20
  face=3D"Courier New" size=3D2>Jos=E9 Soler Lucas, =
M.Sc.Telecom.E.</FONT> <BR><FONT=20
  face=3D"Courier New" size=3D2>PhD Student</FONT> <BR><FONT =
face=3D"Courier New"=20
  size=3D2>Networks Competence Area</FONT> <BR><FONT face=3D"Courier =
New"=20
  size=3D2>Research Center COM (Communications, Optics and =
Materials)</FONT>=20
  <BR><FONT face=3D"Courier New" size=3D2>Technical University of =
Denmark</FONT>=20
  <BR><FONT face=3D"Courier New" size=3D2>Building 343, DK 2800 Kgs. =
Lyngby,=20
  Denmark.</FONT> <BR><FONT face=3D"Courier New" size=3D2>Phone: +45 45 =
25 38 08=20
  (direct)</FONT> <BR><FONT face=3D"Courier New" =
size=3D2>Fax:&nbsp;&nbsp; +45 45 93=20
  65 81</FONT> <BR><FONT face=3D"Courier New" size=3D2>E-mail:=20
  jsoler@com.dtu.dk</FONT> <BR><FONT face=3D"Courier New" size=3D2>Web: =
<A=20
  href=3D"http://www.com.dtu.dk/" target=3D_blank><FONT=20
  color=3D#000000>http://www.com.dtu.dk/</FONT></A></FONT> <BR><FONT=20
  face=3D"Courier New" size=3D2>----</FONT> <BR><FONT face=3D"Courier =
New"=20
  size=3D2>Cellular: +45 22 27 48 22</FONT> <BR><FONT face=3D"Courier =
New"=20
  size=3D2>--------------------------------------------</FONT> =
</P><BR><BR>
  <DIV>&nbsp;</DIV></BLOCKQUOTE></BODY></HTML>

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From: "Roberto Tealdi" <roby.tea@tin.it>
To: <sip@ietf.org>, <sipping@ietf.org>
Subject: [SIP] Hierarchical Registration
Date: Thu, 22 May 2003 15:12:45 +0200
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Hello All,

I'm new on SIP, I have a question about registration procedures, =
expecially the Hierarchical Registration described in Shulzrinne's =
article "Application-Layer Mobility Using SIP".

Suppose this scenario, there are 2 domains (Visited Network and Home =
Network) and there is a SIP registrar proxy in each domain;=20

1) a user in the visited network wants to register himself  at his home =
network. UAC sends a REGISTER message; in this message the Contact =
header is set with the current IP address of the UAC. The message has a =
digital signature in order to provide integrity and authentication, such =
digital signature include the following headers: To, From, Contact, =
Date, C-Seq, Call-ID.

2) Suppose to implement the hierarchical registration as described in =
some Schulzrinne's articles: the SIP outbound proxy in the visited =
domain when receives this message has to change the Contact header =
putting in it its own address, before forwarding the REGISTER message to =
the user's home network. This is done in order to receive and inspect =
(by the proxy in the visited network) all incoming messages for the user =
visiting in. But if the Contact header is changed, the digital signature =
is compromised.

The UAC should know the "correct" Contact value before doing the digital =
signature in his REGISTER messages.

How can the proxy do to "inform" the UAC about this "correct" value? Are =
there existing mechanism? Is possible to do it with 4xx messages?

Note: the UAC signs his messages with his private key, the visited proxy =
do not know this key.

I was supposing that this would be a possible solution (see the =
Scenario.txt attach):

The Visited Proxy in the "4xx Contact Not Acceptable" notify the UAC the =
correct value to put in the Contact header, and in the successive =
REGISTER message the UAC fill the header with this new value. The =
Visited Register does not need to made any changes in the REGISTER =
message so the digital signature is still valid.

It would be possible this solution or there are other mechanisms?

Thank you very much for Your helps.
Roby.


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>Hello All,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I'm new on SIP, I have a question about =

registration procedures, expecially the Hierarchical Registration =
described in=20
Shulzrinne's article "<FONT face=3D"Times New Roman" size=3D4><FONT =
face=3DArial=20
size=3D2>Application-Layer Mobility Using =
SIP".</FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Suppose this scenario, there are 2 =
domains (Visited=20
Network and Home Network) and there is a SIP registrar proxy in each =
domain;=20
<BR><BR>1) a user in&nbsp;the visited network wants to register =
himself&nbsp; at=20
his home network. UAC sends a REGISTER message; in this message the =
Contact=20
header is set with the current IP address of the UAC. The message has a =
digital=20
signature in order to provide integrity and authentication,&nbsp;such =
digital=20
signature include the following headers: To, From, Contact, Date, C-Seq, =

Call-ID.<BR><BR>2) Suppose to implement the hierarchical registration as =

described in&nbsp;some Schulzrinne's articles: the SIP outbound proxy in =
the=20
visited domain when receives this message has to change the Contact =
header=20
putting in it&nbsp;its own address, before forwarding the REGISTER =
message to=20
the user's home network.&nbsp;This is&nbsp;done in order to receive and =
inspect=20
(by the proxy in the visited network) all incoming messages&nbsp;for the =
user=20
visiting in. But if the Contact header is changed, the digital signature =
is=20
compromised.<BR><BR>The UAC should know the "correct" Contact value =
before doing=20
the digital signature in his REGISTER messages.<BR><BR>How can the proxy =
do to=20
"inform" the UAC about this "correct" value? Are there existing =
mechanism? Is=20
possible to do it with 4xx messages?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Note: the UAC signs his messages with =
his private=20
key, the visited proxy do not know this key.<BR><BR>I was supposing that =
this=20
would be a possible solution (see the Scenario.txt attach):<BR><BR>The =
Visited=20
Proxy in the "4xx Contact Not Acceptable" notify&nbsp;the UAC the =
correct value=20
to put in the Contact header, and in the successive REGISTER message the =
UAC=20
fill the header with this new value. The Visited Register does not need =
to made=20
any changes in the REGISTER message so the digital signature is still=20
valid.</FONT></DIV><FONT face=3DArial size=3D2></FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><FONT face=3DArial =
size=3D2></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><FONT face=3DArial size=3D2>It would be =
possible this=20
solution or there are other mechanisms?</DIV>
<DIV><BR>Thank you very much for Your=20
helps.<BR>Roby.<BR></DIV></FONT></FONT></BODY></HTML>

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	filename="Scenario.txt"
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UAC                                        Visited Registrar             =
               Home Registrar
   |               REGISTER                       |                      =
                    |
   |--------------------------------------------->|                      =
                    |
   |                                              |                      =
                    |
   |      4xx Contact Address Not Acceptable      |                      =
                    |
   |<---------------------------------------------|					     |    =20
   |                                              |                      =
                    |
   |     REGISTER (with new Contact)              |                      =
                    |
   |--------------------------------------------->|                      =
                    |
   |                                              |                  =
REGISTER                |
   |                                              =
|----------------------------------------->|
   |                                              |                      =
                    |
   |                                              |                200 =
OK                    |
   |                                              =
|<-----------------------------------------|
   |                200 OK                        |                      =
                    |
   |<---------------------------------------------|                      =
                    |
   |                                              |                      =
                    |

------=_NextPart_000_0021_01C32074.99469220--


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



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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Michael Thomas'" <mat@cisco.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Cc: "Elwell, John" <john.elwell@siemens.com>,
        "'Rohan Mahy'"
	 <rohan@cisco.com>,
        "'Drage, Keith (Keith)'" <drage@lucent.com>,
        "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] UPDATE and P-Asserted-ID
Date: Fri, 23 May 2003 11:21:26 +0100
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Jonathan's approach seems reasonable, and since P-Asserted-Id just
represents the identity of the originator of the *message* 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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<BODY>

<P><FONT SIZE=3D2>Jonathan's approach seems reasonable, and since =
P-Asserted-Id just represents the identity of the originator of the =
*message* 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-&gt;X gateway.</FONT></P>

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

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

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

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Michael Thomas [<A =
HREF=3D"mailto:mat@cisco.com">mailto:mat@cisco.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 19 May 2003 21:18</FONT>
<BR><FONT SIZE=3D2>&gt; To: Jonathan Rosenberg</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: Elwell, John; 'Rohan Mahy'; 'Drage, Keith =
(Keith)'; 'sip@ietf.org'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [Sip] UPDATE and =
P-Asserted-ID</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Jonathan Rosenberg writes:</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; 1. Each header field that conveys =
dialog-associated state would </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; indicate whether or not that state =
is updated by a target </FONT>
<BR><FONT SIZE=3D2>&gt; refresh request.</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; 2. We specify a blanket rule for all =
such headers that </FONT>
<BR><FONT SIZE=3D2>&gt; their state is </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; updated by a target refresh request =
(absence of a header </FONT>
<BR><FONT SIZE=3D2>&gt; would imply </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; no update).</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; 3. We specify a blanket rule for all =
such headers that </FONT>
<BR><FONT SIZE=3D2>&gt; their state is </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; never updated by a target refresh =
request.</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; 4. We specify a default rule for all =
such headers, but allow </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; individual headers to override =
it.</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; In the interests of simplicity I =
would tend to lean </FONT>
<BR><FONT SIZE=3D2>&gt; towards (2). The </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; best place for this, of course, is =
rfc 3261, in which case </FONT>
<BR><FONT SIZE=3D2>&gt; we should </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; treat this as a bug against the =
spec.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; This seems quite reasonable to me as well =
--</FONT>
<BR><FONT SIZE=3D2>&gt; special casing &quot;initial&quot; from =
subsequent dialog</FONT>
<BR><FONT SIZE=3D2>&gt; seems pretty fraught. The only reason that I =
can</FONT>
<BR><FONT SIZE=3D2>&gt; think of why you might want to do that is when =
the</FONT>
<BR><FONT SIZE=3D2>&gt; state assembly/teardown of a mid-session =
change</FONT>
<BR><FONT SIZE=3D2>&gt; was so onerous and/or ambiguous to other =
state</FONT>
<BR><FONT SIZE=3D2>&gt; that it was prudent to set up the state once =
and</FONT>
<BR><FONT SIZE=3D2>&gt; leave it be. It doesn't seem to me that that's =
the</FONT>
<BR><FONT SIZE=3D2>&gt; case here, but I'm hardly an expert.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Mike</FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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Subject: [Sip] Notifier Privacy Mechanism puzzle
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Hi all, 

   I have a question about section 5.2 Notifier Privacy Mechanism in RFC3265 

   <quote> 

  The mere act of returning a 200 or certain 4xx and 6xx responses to
  SUBSCRIBE requests may, under certain circumstances, create privacy
  concerns by revealing sensitive policy information. 

   </quote> 

   I do not unstand why it says a 4xx or 6xx response can reveal sensitive 
policy information, for example, the notifier send a 403 Forbidden or 401 
Unauthorization , can this cause policy problem? I think no. If it can , 
anybody give me some explanations? 

  Thanks. 
_______________________________________________
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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Subject: RE: [Sip] Notifier Privacy Mechanism puzzle
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You might not want watchers to know that you have rejected their subscription to your presence info. Sending a 4xx - 6xx to the subscription will reveal such information. Sending a 2xx followed by a NOTIFY with bogus info may not reveal that sensitive policy information.

/Hisham

> -----Original Message-----
> From: ext Y01317@njupt.edu.cn [mailto:Y01317@njupt.edu.cn]
> Sent: Monday, May 26, 2003 12:07 PM
> To: sip@ietf.org
> Subject: [Sip] Notifier Privacy Mechanism puzzle
> 
> 
> Hi all, 
> 
>    I have a question about section 5.2 Notifier Privacy 
> Mechanism in RFC3265 
> 
>    <quote> 
> 
>   The mere act of returning a 200 or certain 4xx and 6xx responses to
>   SUBSCRIBE requests may, under certain circumstances, create privacy
>   concerns by revealing sensitive policy information. 
> 
>    </quote> 
> 
>    I do not unstand why it says a 4xx or 6xx response can 
> reveal sensitive 
> policy information, for example, the notifier send a 403 
> Forbidden or 401 
> Unauthorization , can this cause policy problem? I think no. 
> If it can , 
> anybody give me some explanations? 
> 
>   Thanks. 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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 May 26 22:49:40 2003
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From: =?iso-8859-1?q?suresh=20t?= <sureshmaillist@yahoo.co.uk>
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hi all.. 

i got one doubt reg. NAT traversal of the SIP
messages...pls clarify me ASAP...

suppose if the user1 is using local Ip address and 
registers with the proxy with Ip address and port(the 
proxy registers the NATed IP and port(dynamic)). 
  
and suppose it is a chat like application, the user2 
wants to connect user1 through some private protocol, 
which is using TCP, but after a long time... 

in this case, the translation table(NAT) in the Router

of user1 is timed out and has been purged. then the 
user2 cannot access the user1's system. 

how can i solve the above problem..?

ur suggestion is highly appreciated.   

thanks in advance 
suresh 



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From mailnull@www1.ietf.org  Tue May 27 00:47:10 2003
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From: "Sujith Kumar" <sujith_km@hotmail.com>
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Subject: Re: [Sip] problem with NAT
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Hi Suresh,
             The sip registration should have a time out and after the time 
out, the sip client needs to register again. The SIP register timeout should 
be less than (or equal) the Transition table entry time out.
I hope, this will solve your problem,.

Regds,
Sujith



>From: suresh t <sureshmaillist@yahoo.co.uk>
>To: sip@ietf.org
>Subject: [Sip] problem with NAT
>Date: Tue, 27 May 2003 03:44:49 +0100 (BST)
>
>hi all..
>
>i got one doubt reg. NAT traversal of the SIP
>messages...pls clarify me ASAP...
>
>suppose if the user1 is using local Ip address and
>registers with the proxy with Ip address and port(the
>proxy registers the NATed IP and port(dynamic)).
>
>and suppose it is a chat like application, the user2
>wants to connect user1 through some private protocol,
>which is using TCP, but after a long time...
>
>in this case, the translation table(NAT) in the Router
>
>of user1 is timed out and has been purged. then the
>user2 cannot access the user1's system.
>
>how can i solve the above problem..?
>
>ur suggestion is highly appreciated.
>
>thanks in advance
>suresh
>
>
>
>__________________________________________________
>It's Samaritans' Week. Help Samaritans help others.
>Call 08709 000032 to give or donate online now at 
>http://www.samaritans.org/support/donations.shtm
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP 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: Ryan Kereliuk <ryker@ryker.org>
To: Sujith Kumar <sujith_km@hotmail.com>
Cc: sip@ietf.org
Subject: Re: [Sip] problem with NAT
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On Tue, May 27, 2003 at 10:13:25AM +0530, Sujith Kumar wrote:
>
>             The sip registration should have a time out and after the time 
> out, the sip client needs to register again. The SIP register timeout 
> should be less than (or equal) the Transition table entry time out.
> I hope, this will solve your problem,.

Your comment is generally correct assuming a UA that uses UDP and supports
both symmetric signalling and very short REGISTER expiry times (and once
you add media into the picture it gets more complex).  Interop experience
shows that quite a few UAs don't support one or the other.

The original poster asked about TCP, however.  Unlike UDP, the NAT/PAT
bindings for TCP can be (and often are) torn down when the TCP state 
machine reaches a final state.  Unless you're using persistent and   
reusable connections you're going to run into trouble.

Let me point out the ICE draft, draft-rosenberg-sipping-ice-00.html,
which AFAIK is the latest attempt to reconcile SIP with the realities
of NAT/PAT.  I'm not sure what kind of traction it's getting but I think
it will be quite a while before something like this is part of common
SIP implementations.

_______________________________________________
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Date: Wed, 21 May 2003 10:48:02 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
In-reply-to: <28B9108644E492469B911D587AABC97C023DEB80@PKDWB01C.ad.sprint.com>
To: "'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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Subject: [Sip] RE: [Sipping] RE: SIP mobility in WLAN
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> If we follow the SIP implementation on Mobile IP then the SIP 
> stack should not have to bother about the mobility aspect.

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

> -----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
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> > 
> > 
> > 
> > 
> > *********************************************************
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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/listinfo/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
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Subject: [Sip] FW: How SIP sessions are maintained in Push to talk?
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> Hi,
> 
> In push to talk services: Based on who is available, the user can simply push a button and start a Push-To-Talk dialog in a walkie-talkie or chat room-like mode. 
> Whenever a user presses a button, he can start speaking instantly. Terminator need not answer this call.
> 
> If we assume that sessions are maintained in PTT through SIP, then whenever a user presses a button, an invite message is sent to the terminator. Then without getting an 200OK message how can a session be established between the two parties?
> 
> Regards,
> Deepti 
> 
_______________________________________________
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This list is for NEW development of the core SIP Protocol
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>
> 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
> > >
> > >
> > >
> > >
> > > *********************************************************
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> > >
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> > > protected by law. If you are not the intended recipient, you should
> > > delete this message and are hereby notified that
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> > > message, or the taking of any action based on it,
> > > is strictly prohibited.
> > >
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> > >
> > >
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> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/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
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>

Prabhas
+++++++++++++
CSEL Technology Enterprise Programme
London Business School
Baker Street, London NW1 6DD
*****************************************
There is no wisdom greater than kindness.
*****************************************
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Hi,

>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?

Regards,

Christer Holmberg
Ericsson Finland
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Prabhas Ranjan Sinha'" <prsinha@unity.ncsu.edu>, <sip@ietf.org>
Subject: RE: [Sip] WLAN & SIP
Date: Tue, 27 May 2003 11:08:36 -0500
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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

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Thread-Topic: a question on section 18.2.2 in RFC 3261
Thread-Index: AcMkdT8nW9n8Rno5QTSsh0t4fXQ44w==
From: "Wang, Jin-Gen" <Jin-Gen.Wang@Level3.com>
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Subject: [Sip] a question on section 18.2.2 in RFC 3261
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Section 18.2.2 specifies:

"18.2.2 Sending Responses

   The server transport uses the value of the top Via header field in
   order to determine where to send a response.  It MUST follow the
   following process:

      o  If the "sent-protocol" is a reliable transport protocol such as
         TCP or SCTP, or TLS over those, the response MUST be sent using
         the existing connection to the source of the original request
         that created the transaction, if that connection is still open.
         This requires the server transport to maintain an association
         between server transactions and transport connections.  If that
         connection is no longer open, the server SHOULD open a
         connection to the IP address in the "received" parameter, if
         present, using the port in the "sent-by" value, or the default
         port for that transport, if no port is specified.  If that
         connection attempt fails, the server SHOULD use the procedures
         in [4] for servers in order to determine the IP address and
         port to open the connection and send the response to.
"

Since the Via header has enough information (IP address + port + transport protocol) for an application to send back response messages, what is the design idea behind the above decision to request an application to send back responses using an existing connection if the connection is still open, other than (may be) optimization in implementation? In other words, from SIP protocol point of view, why is such request a MUST requirement?

Thanks.

Jin-Gen Wang
Level 3 Communications, Inc.

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Date: Tue, 27 May 2003 12:00:43 -0600
From: Ryan Kereliuk <ryker@ryker.org>
To: "Wang, Jin-Gen" <Jin-Gen.Wang@Level3.com>
Cc: sip@ietf.org
Subject: Re: [Sip] a question on section 18.2.2 in RFC 3261
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On Tue, May 27, 2003 at 11:27:28AM -0600, Wang, Jin-Gen wrote:
> 
> Since the Via header has enough information (IP address + port +
> transport protocol) for an application to send back response messages,
> what is the design idea behind the above decision to request an
> application to send back responses using an existing connection if the
> connection is still open, other than (may be) optimization in
> implementation? In other words, from SIP protocol point of view, why is
> such request a MUST requirement?

There are a variety of reasons why this is good including traversal of
of NAT/PAT devices.  I think the motivation for the text as written,
however, was to head off potential scaling and performance problems
associated with doubling the number of TCP/SCTP/TLS connections in use.

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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: Corrado Derenale <derenale@polito.it>
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Subject: [Sip] SIP and EAP
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This is a cryptographically signed message in MIME format.

--------------ms020104090105060907060505
Content-Type: text/plain; charset=us-ascii; format=flowed
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Dear all,
	I saw the Torval proposal for authenticate SIP messages with EAP.

The IETF document draft-torvinen-http-eap-02.txt

Was that proposal been inserted in a new draft or RFC?

We are working to find an extensible authentication schema for SIP and 
are we now evaluating that old proposal.

Thank you in advance,

Corrado Derenale.

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Date: Fri, 30 May 2003 23:34:12 +0530
From: "Nataraju A.B." <natarajuab@huawei.com>
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Hello All,

I have a doubt why do we really need 3-way handshaking for INVITE
transacation.
whereas most of the others employ 2-way handshaking (BYE/200,
UPDATE/200...).

According to my knowledge one of the reason could be offer-answer mechanism
involved in INVITE transactions.

Thanx in advance,

Regards,
-Nataraju A.B.
 Huawei Technologies Ind Pvt Ltd,
 Bangalore-560008,
 +98455.95744

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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] Why do we need 3-way handshake for INVITE transaction....
To: "Nataraju A.B." <natarajuab@huawei.com>
Cc: sip@ietf.org, sip-admin@ietf.org
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From: "Vishal Phirke" <vishal_phirke@nmss.com>
Date: Fri, 30 May 2003 15:03:53 -0400
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Apart from offer-answer mechanism, INVITE transactions differ from other
transactions in following ways:
1. They take more time and often an user intervention is required.
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.
Hence, it is very important to do reliable handshake at the time of session
establishment.
-Vishal


                                                                                                           
                      "Nataraju A.B."                                                                      
                      <natarajuab@huawe        To:       sip@ietf.org                                      
                      i.com>                   cc:                                                         
                      Sent by:                 Subject:  [Sip] Why do we need 3-way handshake for INVITE   
                      sip-admin@ietf.or         transaction....                                            
                      g                                                                                    
                                                                                                           
                                                                                                           
                      05/30/2003 02:04                                                                     
                      PM                                                                                   
                                                                                                           
                                                                                                           




Hello All,

I have a doubt why do we really need 3-way handshaking for INVITE
transacation.
whereas most of the others employ 2-way handshaking (BYE/200,
UPDATE/200...).

According to my knowledge one of the reason could be offer-answer mechanism
involved in INVITE transactions.

Thanx in advance,

Regards,
-Nataraju A.B.
 Huawei Technologies Ind Pvt Ltd,
 Bangalore-560008,
 +98455.95744

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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  Fri May 30 15:21:00 2003
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Date: Sat, 31 May 2003 00:45:53 +0530
From: "Nataraju A.B." <natarajuab@huawei.com>
Subject: Re: [Sip] Why do we need 3-way handshake for INVITE transaction....
To: Vishal Phirke <vishal_phirke@nmss.com>
Cc: sip@ietf.org, sip-admin@ietf.org
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vishal,

does ur 2nd point mean that the SIP application  might select one of the 'n'
replies to the INVITE request (if it was not intenede for multi-party call).
even in this case it is logical to derive that application will selecet the
first response and reject the later 2xx responses OR will the application
wait for some time before sending the ACK back to callee.

Regards,
-Nataraju A.B.
----- Original Message -----
From: "Vishal Phirke" <vishal_phirke@nmss.com>
To: "Nataraju A.B." <natarajuab@huawei.com>
Cc: <sip@ietf.org>; <sip-admin@ietf.org>
Sent: Saturday, May 31, 2003 12:33 AM
Subject: Re: [Sip] Why do we need 3-way handshake for INVITE transaction....


>
> Apart from offer-answer mechanism, INVITE transactions differ from other
> transactions in following ways:
> 1. They take more time and often an user intervention is required.
> 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.
> Hence, it is very important to do reliable handshake at the time of
session
> establishment.
> -Vishal
>
>
>
>                       "Nataraju A.B."
>                       <natarajuab@huawe        To:       sip@ietf.org
>                       i.com>                   cc:
>                       Sent by:                 Subject:  [Sip] Why do we
need 3-way handshake for INVITE
>                       sip-admin@ietf.or         transaction....
>                       g
>
>
>                       05/30/2003 02:04
>                       PM
>
>
>
>
>
>
> Hello All,
>
> I have a doubt why do we really need 3-way handshaking for INVITE
> transacation.
> whereas most of the others employ 2-way handshaking (BYE/200,
> UPDATE/200...).
>
> According to my knowledge one of the reason could be offer-answer
mechanism
> involved in INVITE transactions.
>
> Thanx in advance,
>
> Regards,
> -Nataraju A.B.
>  Huawei Technologies Ind Pvt Ltd,
>  Bangalore-560008,
>  +98455.95744
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>
>
>
>

_______________________________________________
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To: "Vishal Phirke" <vishal_phirke@nmss.com>
Cc: "Nataraju A.B." <natarajuab@huawei.com>, sip@ietf.org, sip-admin@ietf.org
Subject: Re: [Sip] Why do we need 3-way handshake for INVITE transaction....
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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.

regds
arjun

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Subject: Re: [Sip] Why do we need 3-way handshake for INVITE transaction....
To: sip@ietf.org
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From: "Vishal Phirke" <vishal_phirke@nmss.com>
Date: Fri, 30 May 2003 16:07:28 -0400
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Well...As per my understanding, the dialogs will be established as soon as
UA receives 200 OK. The application will generally accept first 200 OK,
sending ACK and will reject others by sending BYE. What I was trying to get
to is sending an ACK will make sure that both parties know that the other
party is ready for session. However, for other requests, which are within
dialog, the end points already know that the other end point is ready to
accept its requests.
-Vishal


                                                                                                           
                      "Nataraju A.B."                                                                      
                      <natarajuab@huawe        To:       Vishal Phirke <vishal_phirke@nmss.com>            
                      i.com>                   cc:       sip@ietf.org, sip-admin@ietf.org                  
                      Sent by:                 Subject:  Re: [Sip] Why do we need 3-way handshake for      
                      sip-admin@ietf.or         INVITE transaction....                                     
                      g                                                                                    
                                                                                                           
                                                                                                           
                      05/30/2003 03:15                                                                     
                      PM                                                                                   
                                                                                                           
                                                                                                           




vishal,

does ur 2nd point mean that the SIP application  might select one of the
'n'
replies to the INVITE request (if it was not intenede for multi-party
call).
even in this case it is logical to derive that application will selecet the
first response and reject the later 2xx responses OR will the application
wait for some time before sending the ACK back to callee.

Regards,
-Nataraju A.B.
----- Original Message -----
From: "Vishal Phirke" <vishal_phirke@nmss.com>
To: "Nataraju A.B." <natarajuab@huawei.com>
Cc: <sip@ietf.org>; <sip-admin@ietf.org>
Sent: Saturday, May 31, 2003 12:33 AM
Subject: Re: [Sip] Why do we need 3-way handshake for INVITE
transaction....


>
> Apart from offer-answer mechanism, INVITE transactions differ from other
> transactions in following ways:
> 1. They take more time and often an user intervention is required.
> 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.
> Hence, it is very important to do reliable handshake at the time of
session
> establishment.
> -Vishal
>
>
>
>                       "Nataraju A.B."
>                       <natarajuab@huawe        To:       sip@ietf.org
>                       i.com>                   cc:
>                       Sent by:                 Subject:  [Sip] Why do we
need 3-way handshake for INVITE
>                       sip-admin@ietf.or         transaction....
>                       g
>
>
>                       05/30/2003 02:04
>                       PM
>
>
>
>
>
>
> Hello All,
>
> I have a doubt why do we really need 3-way handshaking for INVITE
> transacation.
> whereas most of the others employ 2-way handshaking (BYE/200,
> UPDATE/200...).
>
> According to my knowledge one of the reason could be offer-answer
mechanism
> involved in INVITE transactions.
>
> Thanx in advance,
>
> Regards,
> -Nataraju A.B.
>  Huawei Technologies Ind Pvt Ltd,
>  Bangalore-560008,
>  +98455.95744
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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: "Nataraju A.B." <natarajuab@huawei.com>
Subject: Re: [Sip] Why do we need 3-way handshake for INVITE transaction....
To: Arjun Roychowdhury <aroychow@hns.com>,
        Vishal Phirke <vishal_phirke@nmss.com>
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----- 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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Subject: Re: [Sip] Why do we need 3-way handshake for INVITE transaction....
To: "Arjun Roychowdhury" <aroychow@hns.com>
Cc: "Nataraju A.B." <natarajuab@huawei.com>, sip@ietf.org, sip-admin@ietf.org,
        <sip-implementors@cs.columbia.edu>
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From: "Vishal Phirke" <vishal_phirke@nmss.com>
Date: Fri, 30 May 2003 18:13:24 -0400
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I think the problem with the forking is, when an INVITE is forked and you
actually want to setup multiple calls.
In such a case application will send INVITE without SDP body and each 200
OK will carry an offer in SDP to which application can respond via using
ACK with separate media ports for each connection. If 3-way handshake is
not supported, setting up multiple calls with one INVITE is not possible.
-Vishal


                                                                                                           
                      "Arjun                                                                               
                      Roychowdhury"            To:       "Vishal Phirke" <vishal_phirke@nmss.com>          
                      <aroychow@hns.com        cc:       "Nataraju A.B." <natarajuab@huawei.com>,          
                      >                         sip@ietf.org, sip-admin@ietf.org                           
                      Sent by:                 Subject:  Re: [Sip] Why do we need 3-way handshake for      
                      sip-admin@ietf.or         INVITE transaction....                                     
                      g                                                                                    
                                                                                                           
                                                                                                           
                      05/30/2003 03:57                                                                     
                      PM                                                                                   
                                                                                                           
                                                                                                           




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.

regds
arjun

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Subject: Re: [Sip] Doubt about SIP Specific Event Notification behaviour
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The spec doesnt call this out, since I have no idea why such a thing 
would happen. But the right thing to do is undoubtedly (c), have the 
server terminate the subscription by sending a NOTIFY with a 
terminated state.

-Jonathan R.

José Soler Lucas wrote:
> Hi everyone,
>  
> trying to imagine a service based on the SUBSCRIBE - NOTIFY methods of 
> SIP described in RFC 3265 I have came across a doubt that I was not able 
> to find a respose to through the different documents I have checked. 
> Could anyone out there please give me any hint on the matter?:
>  
> Imagine I have a SIP client that subscribes to a certain event happening 
> in a server. The subscription is acepted and performs properly. After a 
> moment, while the subscription is still active, there is an internal 
> problem in the server that prevents further NOTIFYes towards the 
> subscribing client. The question is:
>  
> a) Does the server send a 5xx message towards the client? or
> b) Does the server send a NOTIFY with a "pending" status indicating a 
> temporal suspension of the subscription? or
> c) Does the server send a NOTIFY with a "terminated" status forcing the 
> end of the subscription?
> d) Should the server send rather [5xx] followed by 
> [NOTIFY(status=pending/terminated)]? 
>  
> If the answer is (a), what should the behavior of the client then be, to 
> assume the subscription as terminated?
>  
> Thanks in advance for your help.
> Best regards,
>  
> josé
> 
> --------------------------------------------
> José Soler Lucas, M.Sc.Telecom.E.
> PhD Student
> Networks Competence Area
> Research Center COM (Communications, Optics and Materials)
> Technical University of Denmark
> Building 343, DK 2800 Kgs. Lyngby, Denmark.
> Phone: +45 45 25 38 08 (direct)
> Fax:   +45 45 93 65 81
> E-mail: jsoler@com.dtu.dk
> Web: http://www.com.dtu.dk/
> ----
> Cellular: +45 22 27 48 22
> --------------------------------------------
> 
> 
> 
>  

-- 
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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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Ryan Kereliuk <ryker@ryker.org>
CC: "Wang, Jin-Gen" <Jin-Gen.Wang@level3.com>, sip@ietf.org
Subject: Re: [Sip] a question on section 18.2.2 in RFC 3261
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The reason was simple - pretty much all protocols where you send a 
request on TCP expect a response on TCP under normal conditions 
(normal being, the connection still exists). Anything else is going to 
be more complicated for no gain.

-Jonathan R.

Ryan Kereliuk wrote:
> On Tue, May 27, 2003 at 11:27:28AM -0600, Wang, Jin-Gen wrote:
> 
>>Since the Via header has enough information (IP address + port +
>>transport protocol) for an application to send back response messages,
>>what is the design idea behind the above decision to request an
>>application to send back responses using an existing connection if the
>>connection is still open, other than (may be) optimization in
>>implementation? In other words, from SIP protocol point of view, why is
>>such request a MUST requirement?
> 
> 
> There are a variety of reasons why this is good including traversal of
> of NAT/PAT devices.  I think the motivation for the text as written,
> however, was to head off potential scaling and performance problems
> associated with doubling the number of TCP/SCTP/TLS connections in use.
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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To: Vishal Phirke <vishal_phirke@nmss.com>
CC: Arjun Roychowdhury <aroychow@hns.com>,
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Subject: Re: [Sip] Why do we need 3-way handshake for INVITE transaction....
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Vishal Phirke wrote:
> I think the problem with the forking is, when an INVITE is forked and you
> actually want to setup multiple calls.
> In such a case application will send INVITE without SDP body and each 200
> OK will carry an offer in SDP to which application can respond via using
> ACK with separate media ports for each connection. If 3-way handshake is
> not supported, setting up multiple calls with one INVITE is not possible.

This is not the case. The INVITE can contain an offer, and if forked, 
you get multiple answers. The caller then gets media from multiple 
sources to the same port, which it can disambiguate using SSRC. If it 
wants to move each dialog to a different port, it can then do a 
re-invite if it wants to.

The reason the three way handshake is needed has everything to do with 
the long delays in responding to an invite.

-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
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From mailnull@www1.ietf.org  Sat May 31 02:02:48 2003
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To: "Christer Holmberg (LMF)" <Christer.Holmberg@lmf.ericsson.se>
CC: Sanjay Sinha <sanjsinh@cisco.com>, sip@ietf.org
Subject: Re: [Sip] Question on RFC 3263
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Christer Holmberg (LMF) wrote:
> Hi,
> 
> 
>> 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? In any case, as I said genreally a 
call is not terminated because a 18x never gets a prack.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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From mailnull@www1.ietf.org  Sat May 31 09:46:49 2003
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Subject: [Sip] Question on rfc 3313 : sip extension for media authorization
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Hi all and thanks for previous help ,

in section 7 of rfc 3313 it is said that "the use of
media authorization makes it possible to control the
usage of network resources". 
Isn't this possible by means of RSVP which carry
enough information to identify flows and resources
being used ?

also, it seems to me that the integration of Qos
admission control with call signaling brings
additional session setup delay and additionnal
messages exchange, since there is exchange of REQ/DEC
messages within the phase of resource reservation by
RSVP (refer to section 6. Examples)and authorization
could by performed at this stage.
Please clarify me. Is there something wrong or missing
with my understanding ?
any help appreciated.

Great thanks and Regards.
Hajer.

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Subject: [Sip] question on draft-sinnreich-sip-qos-osp-01
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Hi all,
somebody knows what is the current state of
draft-sinnreich-sip-qos-osp-01 ? Or if there's some
further work on the linkage between call setup using
sip and end-to-end Qos setup.

thanks in advance.
Hajer.

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