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Subject: [Sip] SigComp States
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I'm studying now SigComp and wondering about some issues:

(1) Quoting from draft-ietf-roch-sigcomp-sip-00.txt (section 3): =93Usual=
ly,=20
any states created during the lifetime of a compartment will be "logicall=
y"=20
deleted when the compartment is closed. A logical deletion becomes a=20
physical one when all the compartments that created the (same) state are=20
closed.=94  When do compartments share the same state? When do different=20
compartments become related to the same state?

(2) Quoting from RFC 3320 (section 6.2): =93Each instance of the UDVM can=
 pass=20
up to four state creation requests to the state handler=85=94 Why does a =
UDVM=20
need more than a single allocated state item?

(3) Continuing the quote from the previous question: =93When the state ha=
ndler=20
receives a state creation request from the UDVM it takes the following=20
steps: =85  3. If the state creation request contains a state_identifier =
that=20
already exists=85=94 Is there any clear example to this kind of situation=
?

I would appreciate any piece of information.

Thanks,
John.

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Subject: [Sip] RE: Comments on publish-01
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Hi Ben,

 > -----Original Message-----
 > From: ext Ben Campbell [mailto:bcampbell@dynamicsoft.com]
 > Sent: 28 October, 2003 00:38
 > To: Niemi Aki (NMP/Helsinki)
 > Cc: sip@ietf.org
 > Subject: Comments on publish-01
 >=20
 >=20
 > Hi,
 >=20
 > I have some comments on publish-01. These are not=20
 > necessarily specific=20
 > to 01 changes--I apologize if I bring up old concerns again,=20
 > or mention=20
 > something now that I should have mentioned previously.
 >=20
 >=20
 > Thanks!
 >=20
 > Ben.
 >=20
 > --------------------------------------------------------------
 >=20
 > 4.3: Why do we require all publish packages to support partial=20
 > publication? This seems more a presence requirement than a generic=20
 > publish requirement.

This is true.=20
=20
 > 4.4: Why do we need to specify default expiration times? Why=20
 > is this not=20
 > just local policy of the ESC?

This is also quite a good question. I think it would be enough to just =
specify a default expiration time for the PUBLISH method, and leave =
everything else up to the ESC.
=20
 > 5 (last paragraph): I _think_ the intent is to say you=20
 > should not send a=20
 > new request to a particular ESC when a previous one is=20
 > pending for the=20
 > _same_ ESC, correct? This seems to be the spirit of the=20
 > text, but there=20
 > is room for someone to interpret this to say that I cannot=20
 > send a new=20
 > publish to any ESC if I have not gotten a response from a=20
 > particular ESC.

I'll try to clarify the language.
=20
 > 5.2 Paragraph 1: Still says To header indicates the AoR. I am pretty=20
 > sure we had consensus in SIMPLE that this should be the=20
 > r-URI. (Although=20
 > I recognize we have had a new controversy on the SIP list=20
 > about this.)

Yes, this was simply overlooked while editing.

 > Paragraph 2: Can it _ever_ be carried anywhere else? If not,=20
 > we should=20
 > drop the word "typically"

I think it might, for example using content indirection, or perhaps the =
event state could be in the Request-URI. In any case, there doesn't seem =
to be that much harm if we leave the options open with "typically".=20
=20
 > 5.3: This seems to conflict with the meaning of 4.4. Nothing is=20
 > mentioned in 5.3 about a default expiration.

Will clarify this. It should mention the default expiration.
=20
 > 5.4 para 5, 1st sentence:
 >=20
 > "The If-Match header field containing an entity-tag=20
 > preconditions the=20
 > PUBLISH request to refresh a specific instance of event state in the=20
 > ESC." Sentence is confusing to me. Are we using the word=20
 > "preconditions"=20
 > as a verb?

I will reword.
=20
 > Same paragraph: Nit: I suspect that "i.e." (which translates=20
 > as "that=20
 > is", really should mean "for example", as there are other situations=20
 > other than expired state that could cause this result. (For=20
 > example, the=20
 > state never existed at all.)

Correct, will change this to "e.g.".
=20
 > Last paragraph: I think this should be MUST NOT. Doesn't the=20
 > presence of=20
 > a body explicily make this into a modify request, instead of=20
 > a refresh?

Yes, will use MUST NOT.=20
=20
 > 5.7 First paragraph: Should use r-uri, not To header.

Correct.

 > 6 step 8 last para: Do we want to limit this to explictly=20
 > 500? In the=20
 > example you give, it may be that 504 makes more sense.

Hoe about something like:

       The processing of the PUBLISH request MUST be atomic. If internal
       errors (such as the inability to access a back-end database)
       occur before processing is complete, the publication MUST NOT
       succeed, and the ESC MUST fail with an appropriate error =
response,
       such as 504 (Server Time-out).
=20
 > Table 1: Accept explicitly disallowed in requests? What if a future=20
 > publish package uses bodies in responses? Would it not make sense to=20
 > allow Accept in the request for such a package?

That makes sense for Accept and maybe even all Accept-*. I will add an =
"o" to all of them.

 > Security Considerations: Do we need to comment on the interaction=20
 > between PUBLISH and the possibility of e2e signatures on presence=20
 > documents? I don't know if that belongs here or not.

There is already some text in Section 9.4 - do we need more than that?

Cheers,
Aki
=20
 >=20
 >=20
 >=20

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aki.niemi@nokia.com wrote:

[...]

>  > 6 step 8 last para: Do we want to limit this to explictly 
>  > 500? In the 
>  > example you give, it may be that 504 makes more sense.
> 
> Hoe about something like:
> 
>        The processing of the PUBLISH request MUST be atomic. If internal
>        errors (such as the inability to access a back-end database)
>        occur before processing is complete, the publication MUST NOT
>        succeed, and the ESC MUST fail with an appropriate error response,
>        such as 504 (Server Time-out).

I think that works.

[...]

>  > Security Considerations: Do we need to comment on the interaction 
>  > between PUBLISH and the possibility of e2e signatures on presence 
>  > documents? I don't know if that belongs here or not.
> 
> There is already some text in Section 9.4 - do we need more than that?

I think that is fine--I just missed it the first time.

Thanks!

Ben.


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Subject: [Sip] redirect token in caller preferences extension?
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Hello

I try to implement caller preferences extension in already existing software (which is "ser" and Kphone). I'm not quite sure how UAC should apply "redirect" token. As I see it, when UAC is satisfied with redirection it should remove this token from the following INVITES otherwise it would be redirected forever. Is my thinking correct?

regards

Marcin Dzieweczynski



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From exim@www1.ietf.org  Mon Nov  3 19:06:26 2003
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: <rogan@cisco.com>,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: [Sip] Draft agenda, SIP, IETF 58 version 1
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The following is the proposed agenda for the upcoming SIP meeting. Please
comment if we've missed something or you wish to make a suggestion.

Note that the time slot might move, depedning on overall IETF agenda
requirements.

Hyperlinked version at:
http://www.softarmor.com/sipwg/meets/ietf58/agenda.html


Text follows:

Wednesday, November  12, 1300-1500

1300 Call to Order and Agenda Bash
    Chairs
1305   Status
    Chairs
1315 Congestion Safety
    Dean Willis
    draft-ietf-sip-congest-safe-02
1325  RFC 3312
    Gonzalo Camarillo
    draft-camarillo-sip-rfc3312-update-00.txt
1330 Publish
    Aki Niemi
    draft-ietf-sip-publish-01
1345 Request History
    Mary Barnes
    draft-ietf-sip-history-info-01
1400 Non-Invite Transactions
    Robert Sparks
    draft-sparks-sip-noninvite-01.txt
1415 GRUU
    Jonathan Rosenberg
    draft-rosenberg-sip-gruu-00.txt
    draft-rosenberg-sipping-gruu-reqs-01.txt
1435 Join
    Rohan Mahy
    draft-ietf-sip-join-02.txt
1445 REFER Semantics
    Rohan Mahy  
    draft-olson-sipping-refer-extensions
    draft-mahy-sip-remote-cc


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This list is for NEW development of the core SIP Protocol
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From exim@www1.ietf.org  Tue Nov  4 00:30:35 2003
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To: Marcin Dzieweczynski <mardz581@student.liu.se>
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Subject: Re: [Sip] redirect token in caller preferences extension?
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Marcin Dzieweczynski wrote:

> Hello
> 
> I try to implement caller preferences extension in already existing
> software (which is "ser" and Kphone). I'm not quite sure how UAC
> should apply "redirect" token. As I see it, when UAC is satisfied
> with redirection it should remove this token from the following
> INVITES otherwise it would be redirected forever. Is my thinking
> correct?

Actually, I think the spec is merely ambiguous on this point.

What is supposed to happen is that you get continually redirected to 
new URIs until you reach the UAS. The UAS itself would answer or 
reject the call (it could also redirect of course). Thus, the call 
should still work even if the UAC never removes the redirect token. 
However, the spec says that UAS behavior should be as if it were a 
proxy. This makes sense for the feature preferences, but not for the 
request handling preferences. The spec needs to be clarified that it 
applies to feature preferences and the queue request handling 
directive only. I'll make sure it gets clarified.

Of course, its also reasonable for a UAC to purposefully use the 
redirect preference once, and then remove it later. It depends on the 
specific feature that is being implemented.

-Jonathan R.


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


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From exim@www1.ietf.org  Tue Nov  4 10:06:41 2003
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: Marcin Dzieweczynski <mardz581@student.liu.se>, sip@ietf.org
Subject: Re: [Sip] redirect token in caller preferences extension?
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Marcin,

In retrospect I will agree with Jonathan that it might be useful to 
clarify this further. I guess this is justified by the fact that you 
asked the question.

But it always seemed obvious to me that it should work as Jonathan 
describes. Fundamentally, inserting the "redirect" request-disposition 
is a way for the UAC to say "I want to make all the decisions about 
which alternatives get tried."

If that was confusing, what about other cases? Is it clear that a proxy 
should ignore this option when it is acting on a Route header?

	Paul

Jonathan Rosenberg wrote:
> 
> 
> Marcin Dzieweczynski wrote:
> 
>> Hello
>>
>> I try to implement caller preferences extension in already existing
>> software (which is "ser" and Kphone). I'm not quite sure how UAC
>> should apply "redirect" token. As I see it, when UAC is satisfied
>> with redirection it should remove this token from the following
>> INVITES otherwise it would be redirected forever. Is my thinking
>> correct?
> 
> 
> Actually, I think the spec is merely ambiguous on this point.
> 
> What is supposed to happen is that you get continually redirected to new 
> URIs until you reach the UAS. The UAS itself would answer or reject the 
> call (it could also redirect of course). Thus, the call should still 
> work even if the UAC never removes the redirect token. However, the spec 
> says that UAS behavior should be as if it were a proxy. This makes sense 
> for the feature preferences, but not for the request handling 
> preferences. The spec needs to be clarified that it applies to feature 
> preferences and the queue request handling directive only. I'll make 
> sure it gets clarified.
> 
> Of course, its also reasonable for a UAC to purposefully use the 
> redirect preference once, and then remove it later. It depends on the 
> specific feature that is being implemented.
> 
> -Jonathan R.
> 
> 


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



From exim@www1.ietf.org  Tue Nov  4 14:19:29 2003
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Subject: RE: [Sip] Draft agenda, SIP, IETF 58 version 1
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From: "Henry Chen" <hjlechen@cisco.com>
To: "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
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I think Rohan Mahy's draft-mahy-sip-remote-cc is a good candidate.

Thanks,

Henry Chen


-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]=20
Sent: Monday, November 03, 2003 6:04 PM
To: sip@ietf.org
Cc: rogan@cisco.com; 'Gonzalo Camarillo'
Subject: [Sip] Draft agenda, SIP, IETF 58 version 1



The following is the proposed agenda for the upcoming SIP meeting.
Please comment if we've missed something or you wish to make a
suggestion.

Note that the time slot might move, depedning on overall IETF agenda
requirements.

Hyperlinked version at:
http://www.softarmor.com/sipwg/meets/ietf58/agenda.html


Text follows:

Wednesday, November  12, 1300-1500

1300 Call to Order and Agenda Bash
    Chairs
1305   Status
    Chairs
1315 Congestion Safety
    Dean Willis
    draft-ietf-sip-congest-safe-02
1325  RFC 3312
    Gonzalo Camarillo
    draft-camarillo-sip-rfc3312-update-00.txt
1330 Publish
    Aki Niemi
    draft-ietf-sip-publish-01
1345 Request History
    Mary Barnes
    draft-ietf-sip-history-info-01
1400 Non-Invite Transactions
    Robert Sparks
    draft-sparks-sip-noninvite-01.txt
1415 GRUU
    Jonathan Rosenberg
    draft-rosenberg-sip-gruu-00.txt
    draft-rosenberg-sipping-gruu-reqs-01.txt
1435 Join
    Rohan Mahy
    draft-ietf-sip-join-02.txt
1445 REFER Semantics
    Rohan Mahy =20
    draft-olson-sipping-refer-extensions
    draft-mahy-sip-remote-cc


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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: [Sip] HELP - problem with accessing provisioning GUI
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Dear all,

If I try to access Provisioning GUI (VOCAL 1.5) by executing 
"/usr/local/vocal/bin/provgui localhost 6005", I always get a error message 
shown below:

----------------------------------------------------------------------------------------------------------------------------
Could not open connection to pserver because:

vocal.comm.VPPLowLevelException: The program could not open a socket to the 
server localhost on port 6005.

Reason: connection refused.
------------------------------------------------------------------------------------------------------------------------------

My VOCAL runs on Linux Redhat 7.3, and J2se 1.4.1 is installed.

Anyone knows what's wrong? Thanks a lot in advance!

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From exim@www1.ietf.org  Wed Nov  5 12:06:29 2003
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Subject: Re: [Sip] GRUU Mechanism I-D
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Paul Kyzivat <pkyzivat@cisco.com>, sip@ietf.org, rohan@cisco.com
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I do think it would be very good medicine for the protocol to
deprecate all dialog reuse.

My personal preference for modifying REFER is to deprecate the
implicit subscription behavior altogether (by strengthening the
mechanisms proposed in Sean's and Rohan's drafts perhaps).

I am uncomfortable with the idea of an implicit subscription
created with an entirely new dialog id - this would require
putting a new dialog correlation algorithm into all implementations.
I'm sure we could invent something that worked, but is the complexity
worth the gain when just removing the implicit subscription solves
the problem and comes with with so much better behavior properties
given the GRUU.

RjS


> >>
> >> This is a good point. Of course, the REAL problem here is that the 
> >> REFER request ought to go on the dialog, followed by a separate 
> >> SUBSCRIBE to the GRUU, in order to track progress of the refer. That 
> >> is, its the implicit subscriptions which makes this complicated.
> >>
> >> That said, we may need to except REFER from this rule. Bleh.
> > 
> > 
> > If we need to exempt REFER, then we have admitted that we can't phase 
> > out shared dialogs. Then, if we have to continue to support shared 
> > dialogs, why do we want to tell people not to use them?
> 
> If there is no way to fix this for REFER, I would have to agree with 
> you. However, if there is a possibility to update REFER to take 
> advantage of gruu, and thus avoid dialog reuse, then it makes sense. 
> AFAICT, the fix to REFER is not hard at all. The recipient of the 
> REFER, who creates the NOTIFY, would simply send it to the gruu of the 
> referor, and use a different dialog ID. There is still an implicit 
> subscription, but the NOTIFY creates a new, separate dialog.



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Subject: Re: [Sip] GRUU Mechanism I-D
References: <3F94E9A9.5020707@dynamicsoft.com> <3F953BD9.7040505@cisco.com>	 <3F95F1A2.2060107@dynamicsoft.com> <3F968FCA.1060907@cisco.com>	 <3F973D54.3070506@dynamicsoft.com> <1068051884.1584.47.camel@RjS.localdomain>
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I would be happy if dialog reuse had never been there. But at this point 
I think it is impossible to avoid supporting it. And once the effort has 
been made to support it, deprecating it in favor of a new approach is 
just that much more work, with more backwards compatibiilty alternatives 
to consider, etc. At this point I would prefer to live with the mistake.

	Paul

Robert Sparks wrote:
> I do think it would be very good medicine for the protocol to
> deprecate all dialog reuse.
> 
> My personal preference for modifying REFER is to deprecate the
> implicit subscription behavior altogether (by strengthening the
> mechanisms proposed in Sean's and Rohan's drafts perhaps).
> 
> I am uncomfortable with the idea of an implicit subscription
> created with an entirely new dialog id - this would require
> putting a new dialog correlation algorithm into all implementations.
> I'm sure we could invent something that worked, but is the complexity
> worth the gain when just removing the implicit subscription solves
> the problem and comes with with so much better behavior properties
> given the GRUU.
> 
> RjS
> 
> 
> 
>>>>This is a good point. Of course, the REAL problem here is that the 
>>>>REFER request ought to go on the dialog, followed by a separate 
>>>>SUBSCRIBE to the GRUU, in order to track progress of the refer. That 
>>>>is, its the implicit subscriptions which makes this complicated.
>>>>
>>>>That said, we may need to except REFER from this rule. Bleh.
>>>
>>>
>>>If we need to exempt REFER, then we have admitted that we can't phase 
>>>out shared dialogs. Then, if we have to continue to support shared 
>>>dialogs, why do we want to tell people not to use them?
>>
>>If there is no way to fix this for REFER, I would have to agree with 
>>you. However, if there is a possibility to update REFER to take 
>>advantage of gruu, and thus avoid dialog reuse, then it makes sense. 
>>AFAICT, the fix to REFER is not hard at all. The recipient of the 
>>REFER, who creates the NOTIFY, would simply send it to the gruu of the 
>>referor, and use a different dialog ID. There is still an implicit 
>>subscription, but the NOTIFY creates a new, separate dialog.
> 
> 
> 
> 


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Brett Tate wrote:
> 
> Most of the current dialog reuse issues could be 
> resolved by a new header which indicates a
> real call-id and a parameter which indicates
> that this is a request to establish a new 
> call/subscription/... on this dialog.  

I'm not clear what that helps.

> A new response is also needed to indicate that 
> I am aware of this dialog however I am not aware
> of the call/subscription/... associated with
> the request (a 481 which doesn't release the 
> dialog).

When would this be used? Perhaps if a NOTIFY was received for which no 
corresponding subscription is known? I agree that is a potential 
problem, and a suitable error code would be helpful. AFAIK that is the 
only case where such a problem can arise.

> To avoid some complexities, potentially we could
> also restrict the number of calls sharing a
> dialog to be 1 at any given time.  Or potentially
> we could add a statement that a call cannot be
> originated on an already established dialog.

As things stand there can only be one call on a dialog at a time. The 
only action that could establish a new one (an INVITE) will be 
interpretted as a reINVITE for the existing call.

In theory you could establish a dialog with an invite, share it with a 
SUBSCRIBE, terminate the first call with a BYE, and then create a new 
call (with another INVITE) sharing the dialog with the subscription.

A rational use of this capability escapes me at the moment, but once the 
mechanisms are there to support sharing at all this kind of capability 
more or less comes for free, so why worry about it?

The use of GRUUs certainly provides an alternative to sharing of dialogs 
for certain applications. I think it makes sense to just permit them to 
coexist and see how things evolve.

	Paul


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Subject: RE: [Sip] GRUU Mechanism I-D
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> > Most of the current dialog reuse issues could be 
> > resolved by a new header which indicates a
> > real call-id and a parameter which indicates
> > that this is a request to establish a new 
> > call/subscription/... on this dialog.  
> 
> I'm not clear what that helps.

A and B both know about a subscription dialog.
A thinks that a call is currently active on the
dialog; B does not.  A sends a re-invite for
the call; B rings the phone since it does not
know if it was an invite or re-invite.

> > A new response is also needed to indicate that 
> > I am aware of this dialog however I am not aware
> > of the call/subscription/... associated with
> > the request (a 481 which doesn't release the 
> > dialog).
> 
> When would this be used? 
> Perhaps if a NOTIFY was received for which no 
> corresponding subscription is known? I agree 
> that is a potential problem, and a suitable 
> error code would be helpful. AFAIK that is the 
> only case where such a problem can arise.

A and B both know about a subscription dialog.
A thinks that a call is currently active on the
dialog; B does not.  A sends a re-invite/update 
for the call; B uses the response code to inform
that the call is not active.  A can now choose
to send an INVITE to request the re-establishment
of the call still on the subscription dialog.

> > To avoid some complexities, potentially we could
> > also restrict the number of calls sharing a
> > dialog to be 1 at any given time.  Or potentially
> > we could add a statement that a call cannot be
> > originated on an already established dialog.
> 
> As things stand there can only be one call on 
> a dialog at a time. The only action that could 
> establish a new one (an INVITE) will be 
> interpretted as a reINVITE for the existing call.
> 
> In theory you could establish a dialog with 
> an invite, share it with a SUBSCRIBE, terminate the 
> first call with a BYE, and then create a new 
> call (with another INVITE) sharing the dialog 
> with the subscription.
> 
> A rational use of this capability escapes me

I am sure that someone will define an event
package that allows calls to be 
established/re-established 
sharing the subscription's dialog.

> at the moment, but once the mechanisms are there 
> to support sharing at all this kind of capability 
> more or less comes for free, so why worry about it?

Mainly so A can distinguish a received re-invite 
from invite when A recently sent a BYE or lost
awareness of the call but not the dialog.

However I guess that it would not be that common
if UPDATE is used to refresh the Session-Expires
stuff.   And I guess a retry-after response could 
be used for an invite received in proximity of 
sending bye.

> The use of GRUUs certainly provides an alternative 
> to sharing of dialogs for certain applications. 
> I think it makes sense to just permit them to 
> coexist and see how things evolve.

I agree.


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Subject: RE: [Sip] GRUU Mechanism I-D
Date: Thu, 6 Nov 2003 12:12:41 +0200
Message-ID: <2038BCC78B1AD641891A0D1AE133DBB701797392@esebe019.ntc.nokia.com>
Thread-Topic: [Sip] GRUU Mechanism I-D
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I share Paul's view.  There are currently too many implementations =
supporting dialog reuse.

The only suggestion I have is that for any new SIP extension in the =
future that requires either dialog reuse or GRUU, we use the GRUU =
approach.

/Hisham

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
> Paul Kyzivat
> Sent: Wednesday, November 05, 2003 8:00 PM
> To: Robert Sparks
> Cc: Jonathan Rosenberg; sip@ietf.org; rohan@cisco.com
> Subject: Re: [Sip] GRUU Mechanism I-D
>=20
>=20
> I would be happy if dialog reuse had never been there. But at=20
> this point=20
> I think it is impossible to avoid supporting it. And once the=20
> effort has=20
> been made to support it, deprecating it in favor of a new approach is=20
> just that much more work, with more backwards compatibiilty=20
> alternatives=20
> to consider, etc. At this point I would prefer to live with=20
> the mistake.
>=20
> 	Paul
>=20
> Robert Sparks wrote:
> > I do think it would be very good medicine for the protocol to
> > deprecate all dialog reuse.
> >=20
> > My personal preference for modifying REFER is to deprecate the
> > implicit subscription behavior altogether (by strengthening the
> > mechanisms proposed in Sean's and Rohan's drafts perhaps).
> >=20
> > I am uncomfortable with the idea of an implicit subscription
> > created with an entirely new dialog id - this would require
> > putting a new dialog correlation algorithm into all implementations.
> > I'm sure we could invent something that worked, but is the=20
> complexity
> > worth the gain when just removing the implicit subscription solves
> > the problem and comes with with so much better behavior properties
> > given the GRUU.
> >=20
> > RjS
> >=20
> >=20
> >=20
> >>>>This is a good point. Of course, the REAL problem here is=20
> that the=20
> >>>>REFER request ought to go on the dialog, followed by a separate=20
> >>>>SUBSCRIBE to the GRUU, in order to track progress of the=20
> refer. That=20
> >>>>is, its the implicit subscriptions which makes this complicated.
> >>>>
> >>>>That said, we may need to except REFER from this rule. Bleh.
> >>>
> >>>
> >>>If we need to exempt REFER, then we have admitted that we=20
> can't phase=20
> >>>out shared dialogs. Then, if we have to continue to support shared=20
> >>>dialogs, why do we want to tell people not to use them?
> >>
> >>If there is no way to fix this for REFER, I would have to=20
> agree with=20
> >>you. However, if there is a possibility to update REFER to take=20
> >>advantage of gruu, and thus avoid dialog reuse, then it=20
> makes sense.=20
> >>AFAICT, the fix to REFER is not hard at all. The recipient of the=20
> >>REFER, who creates the NOTIFY, would simply send it to the=20
> gruu of the=20
> >>referor, and use a different dialog ID. There is still an implicit=20
> >>subscription, but the NOTIFY creates a new, separate dialog.
> >=20
> >=20
> >=20
> >=20
>=20
>=20
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Apologies if some of these comments below have already been made on the =
list. There are no major comments.

- General:
   - I think a section should be dedicated to etags. This section would =
explain how they work. At the moment, it is all over the place.

- Abstract:
   - The 1st sentence in the second paragraph repeats what is said in =
the first paragraph.

- Section 1:=20
   - The 1st sentence in the 3rd paragraph repeats what is said in the =
first paragraph.

- Section 3:
   - Second paragraph repeats definitions section. Maybe this paragraph =
can go away and references added to definitions section.

- Section 4.3:
   - Why is it a MUST for the content-type to provide partial =
publication? I think MAY is enough

- Section 5:
   - The note: Why would a PUBLISH fork? If would only fork if the user =
has explicitly registered 2 different addresses or the proxy is =
configured with 2 ESCs. In any case, I don't get why the PUBLISH may not =
be appropriately presented. I think this note can go away.
   - PUBLISH does not create a dialog. Can we have some text here why =
this is so. I just to remind us in 2 years time when we start asking the =
question again.
   - Why is it explicitly forbidden for a PUBLISH to contain a contact. =
It could be as simple as saying that the ESC ignores the contact header.

- Section 5.2:
   - Request-URI should be used instead of To-header
   - It should be clarified that an etag associates an EPA with the ESC =
for that boot cycle or until the EPA removes a publication (or something =
like that) and does not change for that duration. (I was momentarily =
confused and thought that the etag changes with every PUBLISH from the =
same EPA).
   - I think I understand the etag mechanism, but is there a chance that =
a state already exists for a EPA before that initial PUBLISH (a crash =
and reboot or something). How does the ESC behave in this case? Does the =
ESC know it is the same ESC publishing and therefore should remove the =
state that was created before this new PUBLISH?

- Section 5.4 and 5.5:
   - "the 200 (OK) response to a PUBLISH request from the ESC contains =
..." appears a few times. This confused me into thinking that you are =
talking about the 200 of the refresh/modification PUBLISH. I think you =
mean the "previous" PUBLISH.

- Section 5.5:
   - The note should be normative text.

- Section 5.6
   - "EPAs which support the PUBLISH method SHOULD support this =
mechanism for explicitly removing event state." Should the SHOULD be a =
MUST?

Regards,
Hisham

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  This draft presents an excellent explanation of how and why the
  use of UDP in SIP can (and therefor by Murphys Law, will :-) cause
  network operation problems.  In doing so, it makes a good case for
  SIP having been designed differently, possibly not using UDP at
  all, but acknowleges that it is too late for that.  It goes on to
  propose mechanisms by which this oversight can now be corrected by
  defining a congestion-safe mode of operation and adding a protocol
  feature to require operation in that mode.

  Unfortunately, I don't think that creating such a mechanism is
  useful, because there is little or no incentive for any
  significant number of systems to use it.  The cat is not only out
  of the bag in that SIP was designed to support UDP, but later
  standards (SIP use of NAPTR and DNSSRV in RFC 3263) have provided
  for explicit administrative control mechanisms by which users
  (administrators) can (perhaps unwisely) prefer the use of UDP.

  The proposed 'congestion-managed' option would allow a system to
  send messages that conflict with policy as set by network
  administration at some later point in the message path; to
  illustrate:

    UAC     Proxy       Proxy     Proxy      UAC
     A -----> B --------> C ------> D ------> E

  UAC A sends an INVITE to E with 'Proxy-Require: congestion-managed',
  initially handing it off to Proxy B.  Using normal SIP routing
  mechanisms, that message finds itself in due course at Proxy D
  (the local proxy on E's network).  The administrator of that
  network has (for whatever reason) chosen to configure SIP on that
  network to prefer (or even be limited to) UDP.  This potential
  conflict raises two questions:

   - Why should a user elsewhere in the network (A) be empowered to
     exercise any control of routing policy within the local network
     (E's network)?

   - Why would A want to send a request using this mechanism, given
     that doing so introduces a new potential failure mode, but does
     not improve that chances that this individual request will
     succeed?

  Granted - A acted (as any IETFer would naturally be inclined to
  do) with the best of intentions, with the health of the Internet
  itself in mind.  If everyone did as A did, and all systems
  supported it, the network would be healthier and happier, and SIP
  (along with everything else) would work better.

  BUT.

  If one takes the view of a product manufacturer, or a local
  network administrator - why would you construct or configure your
  system such that lack of support for that new and improved world
  in just one system outside your control would cause your calls to
  fail?  Remember that your only way to make them work when that
  (inevitably) happens is to resend them without
  'congestion-managed', so you're going to end up potentially
  contributing to congestion problems anyway - but your first
  attempt failed even if no congestion problem actually existed.

  I do think that there is a problem to be solved - I think that the
  draft makes that case well, and has some good rules for system
  developers and administrators on how best to configure systems.
  Rather than a protocol extension, I would prefer to see this work
  redirected to producing a BCP document on how to configure SIP
  requests to prevent congestion problems - the incentives of network
  and SIP administrators are all in alignment there.

-- 
Scott Lawrence        
  Pingtel Corp.   


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From exim@www1.ietf.org  Thu Nov  6 10:16:46 2003
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Date: Thu, 06 Nov 2003 15:13:31 +0000
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Subject: [Sip] Questions about SigComp over SIP
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Hi,

I would like to develop SigComp over SIP, and find some issues unclear:

1.	Quoting from RFC3486 (Compressing the Session Initiation Protocol): =93=
The=20
presence of this parameter (comp=3Dsigcomp) in a URI indicates that the=20
request has to be compressed using SigComp, as defined in [2]=94 (section=
 no.=20
2).
Then, below in section no.7 it continues with: =93If a SIP client sends a=
=20
compressed request and the client transaction times out without having=20
received any response, the client SHOULD retry the same request without=20
using compression.=94  This quotation instructs us to send the same messa=
ge=20
without specifying the following details:
=B7	Is the parameter =91comp=3Dsigcomp=92 has to be removed from the orig=
inal=20
request URI before re-sending the uncompressed SIP request message?
=B7	In case that a SIP message has to be re-sent uncompressed, should it=20
contain different Via branch value (as it is done after transmission=20
failure, when using DNS) or different CSeq?


2.	Is there any example of SigComp state that is created dynamically and =
is=20
used by more than a single compartment? I=92m looking for an example of s=
tate=20
that is not =91global=92 or created during initialization process (such a=
s state=20
that keeps the dictionary or state that keeps compression/decompression=20
algorithm).


3.	Do the compression/decompression algorithms instruct us which states a=
nd=20
how many of them have to be created in order to support the SigComp featu=
re?=20
or Does the states creation depend on the way that SigComp is implemented=
?

I would appreciate answers to any of my questions.

Thank You,
John.

_________________________________________________________________
Tired of spam? Get advanced junk mail protection with MSN 8.=20
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From exim@www1.ietf.org  Thu Nov  6 10:51:43 2003
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Subject: Re: [Sip] GRUU Mechanism I-D
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Cc: Robert Sparks <rsparks@dynamicsoft.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Paul Kyzivat <pkyzivat@cisco.com>, sip@ietf.org
To: Adam Roach <adam@dynamicsoft.com>
From: Rohan Mahy <rohan@cisco.com>
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Frankly, I think this medicine will kill the patient.  While not 
pretty, what we have *is* implemented and implementable. While not what 
we would have chosen another go-around, I think we should stick with 
the behavior we have.

thanks,
-rohan


On Wednesday, November 5, 2003, at 12:56 PM, Adam Roach wrote:

>> I do think it would be very good medicine for the protocol to
>> deprecate all dialog reuse.
>
> Hear, hear!
>
>> My personal preference for modifying REFER is to deprecate the
>> implicit subscription behavior altogether (by strengthening the
>> mechanisms proposed in Sean's and Rohan's drafts perhaps).
>
> I agree with this course of action, for the reasons you cite,
> and others.
>
> /a


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From exim@www1.ietf.org  Thu Nov  6 11:02:26 2003
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From: Johan Bilien <jobi@via.ecp.fr>
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Subject: [Sip] Certificates handling in SIP/TLS
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Hi,

I have a few questions regarding the certificates handling when using
SIP over TLS:

  . When the mutual authentication isn't used, should the TLS connection
  between the proxy and the UA be kept alive after the first
  registration ? Otherwise, how can the proxy contact the client, for
  example to forward an incoming INVITE: if the initial connection was
  closed, the UA will need a TLS listening connection, thus act as a TLS
  server and need a certificate.

  . When using mutual authentication, to which identity should the UA
  certificate's CN refer ? To the FQDN of the UA, or to the SIP URI ? In
  the last case, it would allow to use the same certificate as in MIKEY
  for instance.

Thanks,
-- 
Johan Bilien

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Subject: [Sip] Last Call: 'Caller Preferences for the Session Initiation Protocol
 (SIP)' to Proposed Standard
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The IESG has received a request from the Session Initiation Protocol WG to 
consider the following document:

- 'Caller Preferences for the Session Initiation Protocol (SIP) '
   <draft-ietf-sip-callerprefs-10.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2003-11-28.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-callerprefs-10.txt


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Brett Tate wrote:
>>>Most of the current dialog reuse issues could be 
>>>resolved by a new header which indicates a
>>>real call-id and a parameter which indicates
>>>that this is a request to establish a new 
>>>call/subscription/... on this dialog.  
>>
>>I'm not clear what that helps.
> 
> 
> A and B both know about a subscription dialog.
> A thinks that a call is currently active on the
> dialog; B does not.  A sends a re-invite for
> the call; B rings the phone since it does not
> know if it was an invite or re-invite.
> 
> 
>>>A new response is also needed to indicate that 
>>>I am aware of this dialog however I am not aware
>>>of the call/subscription/... associated with
>>>the request (a 481 which doesn't release the 
>>>dialog).
>>
>>When would this be used? 
>>Perhaps if a NOTIFY was received for which no 
>>corresponding subscription is known? I agree 
>>that is a potential problem, and a suitable 
>>error code would be helpful. AFAIK that is the 
>>only case where such a problem can arise.
> 
> 
> A and B both know about a subscription dialog.
> A thinks that a call is currently active on the
> dialog; B does not.  A sends a re-invite/update 
> for the call; B uses the response code to inform
> that the call is not active.  A can now choose
> to send an INVITE to request the re-establishment
> of the call still on the subscription dialog.

OK, I see. Taking both of your comments above - your goal is to 
distinguish "call refresh" from "new call" and "subscription refresh" 
from "new subscription".

It isn't really needed for subscriptions (or for REFER) because there 
are already unique event ids that distinguish them.

But for the call there is indeed ambiguity - you can only have one call 
at a time, but in race conditions there could be confusion about whether 
an INVITE is or is not a reINVITE. I'm sure we can find some relatively 
simple way to resolve that.

>>>To avoid some complexities, potentially we could
>>>also restrict the number of calls sharing a
>>>dialog to be 1 at any given time.  Or potentially
>>>we could add a statement that a call cannot be
>>>originated on an already established dialog.
>>
>>As things stand there can only be one call on 
>>a dialog at a time. The only action that could 
>>establish a new one (an INVITE) will be 
>>interpretted as a reINVITE for the existing call.
>>
>>In theory you could establish a dialog with 
>>an invite, share it with a SUBSCRIBE, terminate the 
>>first call with a BYE, and then create a new 
>>call (with another INVITE) sharing the dialog 
>>with the subscription.
>>
>>A rational use of this capability escapes me
> 
> 
> I am sure that someone will define an event
> package that allows calls to be 
> established/re-established 
> sharing the subscription's dialog.

Why does anything need to be defined in an event package for this?

AFAIK it is quite ok today to establish a dialog via a SUBSCRIBE, and 
then send an INVITE within that dialog to establish a new call.

The question is whether this is a *useful* capability.

I guess one good example of a use would be when subscribing to the 
dialog package. If you saw a call that you wanted to affect - perhaps 
join - then it makes perfect sense to send the INVITE thru the same dialog.

Of course this is also the exact use case that GRUUs render unnecessary.

	Paul


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The IESG has received a request from the Session Initiation Protocol WG to 
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On Thu, 2003-11-06 at 08:34, Scott Lawrence wrote:
>   This draft presents an excellent explanation of how and why the
>   use of UDP in SIP can (and therefor by Murphys Law, will :-) cause
>   network operation problems.  In doing so, it makes a good case for
>   SIP having been designed differently, possibly not using UDP at
>   all, but acknowleges that it is too late for that.  It goes on to
>   propose mechanisms by which this oversight can now be corrected by
>   defining a congestion-safe mode of operation and adding a protocol
>   feature to require operation in that mode.

>  Unfortunately, I don't think that creating such a mechanism is
>   useful, because there is little or no incentive for any
>  significant number of systems to use it.  The cat is not only out
>  of the bag in that SIP was designed to support UDP, but later
>  standards (SIP use of NAPTR and DNSSRV in RFC 3263) have provided
>  for explicit administrative control mechanisms by which users
>  (administrators) can (perhaps unwisely) prefer the use of UDP.

. . . 

Excellent summary and commentary.

Here's the deal. Before long, EVERY e2e-secured SIP request or response
is going to be bigger than one packet. The security people aren't going
to give up on e2e.

This gives us two viable alternatives, as I see them:

1) Deprecate UDP usage in SIP.

2) Deprecate SIP.

Since #2 would leave me unemployed :-(, the question I'm interested is
how to do #1.

Three possibilities suggest themselves, and I'd like to get a sense of
consensus from the group on the direction (or other suggestions, if
there are any).

1) Leave the UDP stuff in place, but convince people not to use it by
doing a "UDP Considered Harmful for SIP" BCP,  (which Scott seems to be
suggesting). 

2) Leave the UDP stuff in place, but give people tools to keep it from
being used (which this draft suggests).

3) Rev the spec to 3.0 and rip UDP out by the toenails.

Any other ideas? What is the right thing to do here?

--
Dean

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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] GRUU Mechanism I-D
Date: Thu, 6 Nov 2003 16:44:54 -0500
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> > A and B both know about a subscription dialog.
> > A thinks that a call is currently active on the
> > dialog; B does not.  A sends a re-invite/update 
> > for the call; B uses the response code to inform
> > that the call is not active.  A can now choose
> > to send an INVITE to request the re-establishment
> > of the call still on the subscription dialog.
> 
> OK, I see. Taking both of your comments above - 
> your goal is to distinguish "call refresh" from 
> "new call" 

Yes.

> and "subscription refresh" from "new subscription".

Actually it is being able to reject a notify with a cause
code that reflects that I am aware of the dialog but
not the subscription.  No response code indicates known
dialog but unknown subscription/call.

> But for the call there is indeed ambiguity - 
> you can only have one call at a time, but in race 
> conditions there could be confusion about whether 
> an INVITE is or is not a reINVITE. I'm sure we 
> can find some relatively simple way to resolve that.

I agree.

> >>As things stand there can only be one call on 
> >>a dialog at a time. The only action that could 
> >>establish a new one (an INVITE) will be 
> >>interpretted as a reINVITE for the existing call.
> >>
> >>In theory you could establish a dialog with 
> >>an invite, share it with a SUBSCRIBE, terminate the 
> >>first call with a BYE, and then create a new 
> >>call (with another INVITE) sharing the dialog 
> >>with the subscription.
> >>
> >>A rational use of this capability escapes me
> > 
> > 
> > I am sure that someone will define an event
> > package that allows calls to be 
> > established/re-established 
> > sharing the subscription's dialog.
> 
> Why does anything need to be defined in an 
> event package for this?

I was mainly indicating that nobody has yet
submitted a draft/rfc indicating such functionality.
However I am sure that somebody will start
defining call-me-back-over-this-dialog type
subscription events.

> AFAIK it is quite ok today to establish a dialog 
> via a SUBSCRIBE, and then send an INVITE within 
> that dialog to establish a new call.

Yes.  However many products would reject the invite
unless it has meaning associated with the subscription.

> The question is whether this is a *useful* capability.
>
> I guess one good example of a use would be when 
> subscribing to the dialog package. If you saw a 
> call that you wanted to affect - perhaps 
> join - then it makes perfect sense to send the 
> INVITE thru the same dialog.
> 
> Of course this is also the exact use case that 
> GRUUs render unnecessary.

Agreed.  

That was why I was trying to either:

1) Resolve the issues of 
   originating/re-originating a call 
   over a subscription's dialog.

2) Or mention that such functionality 
   is not approved by the working group
   because GRUU must be used instead.

I prefer option 1 since it will take time to
get the GRUU solution approved for RFC 
and widely deployed.


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To: Dean Willis <dean.willis@softarmor.com>
Cc: sip@ietf.org, slawrence@pingtel.com
References: <1068153082.8295.59.camel@bdsl.greycouncil.com>
Organization: Pingtel Corp. http://www.pingtel.com/
From: Scott Lawrence <slawrence@pingtel.com>
Date: Thu, 06 Nov 2003 16:56:30 -0500
In-Reply-To: <1068153082.8295.59.camel@bdsl.greycouncil.com> (Dean Willis's
 message of "Thu, 06 Nov 2003 15:11:22 -0600")
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Subject: [Sip] Re: Consensus on direction needed
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Dean Willis <dean.willis@softarmor.com> writes:

> The security people aren't going to give up on e2e.

Which is not the same thing as saying that users will use it... but
that's a different discussion.

> Three possibilities suggest themselves, and I'd like to get a sense of
> consensus from the group on the direction (or other suggestions, if
> there are any).
>
> 1) Leave the UDP stuff in place, but convince people not to use it by
> doing a "UDP Considered Harmful for SIP" BCP,  (which Scott seems to be
> suggesting). 
>
> 2) Leave the UDP stuff in place, but give people tools to keep it from
> being used (which this draft suggests).
>
> 3) Rev the spec to 3.0 and rip UDP out by the toenails.
>
> Any other ideas? What is the right thing to do here?

#1 is a good start, but in the absense of an alternative that is
perceived as being equally good, it won't matter.

accordingly, I'd modify #2 slightly to: 

Leave the UDP stuff in place, _and_ provide good alternatives that are
better.

The BCP will be a valuable part of getting the word out that something
else is better by educating administrators & users about the problem
with SIP/UDP.  I do think that we need to refine some other things
further to provide the alternative.

One suggestion I'd make toward that end is that everyone look closely
at draft-mahy-sip-connect-reuse-00 - I think that it goes a long way
to solving some of the barriers to wider adoption of SIP over TCP.
I've a comment or two on that that I'll send separately...

-- 
Scott Lawrence        
  Pingtel Corp.   


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From: Gordon Ledgard <gledgard@iperia.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>, sip@ietf.org,
        slawrence@pingtel.com
Subject: RE: Consensus on direction needed (was Re: [Sip] Re: draft-ietf-s
	ip-congestsafe-02.txt)]
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>> On Thu, 2003-11-06 at 08:34, Scott Lawrence wrote:
>>   This draft presents an excellent explanation of how and why the
>>   use of UDP in SIP can (and therefor by Murphys Law, will :-) cause
>>   network operation problems.  In doing so, it makes a good case for
>>   SIP having been designed differently, possibly not using UDP at
>>   all, but acknowleges that it is too late for that.  It goes on to
>>   propose mechanisms by which this oversight can now be corrected by
>>   defining a congestion-safe mode of operation and adding a protocol
>>   feature to require operation in that mode.
>>
>>  Unfortunately, I don't think that creating such a mechanism is
>>   useful, because there is little or no incentive for any
>>  significant number of systems to use it.  The cat is not only out
>>  of the bag in that SIP was designed to support UDP, but later
>>  standards (SIP use of NAPTR and DNSSRV in RFC 3263) have provided
>>  for explicit administrative control mechanisms by which users
>>  (administrators) can (perhaps unwisely) prefer the use of UDP.

On Thu, 2003-11-06 at 16:12, Dean Willis wrote:

> Excellent summary and commentary.
> 
> Here's the deal. Before long, EVERY e2e-secured SIP request or response
> is going to be bigger than one packet. The security people aren't going
> to give up on e2e.
> 
> This gives us two viable alternatives, as I see them:
> 
> 1) Deprecate UDP usage in SIP.
> 
> 2) Deprecate SIP.
> 
> Since #2 would leave me unemployed :-(, the question I'm interested is
> how to do #1.
> 
> Three possibilities suggest themselves, and I'd like to get a sense of
> consensus from the group on the direction (or other suggestions, if
> there are any).
> 
> 1) Leave the UDP stuff in place, but convince people not to use it by
> doing a "UDP Considered Harmful for SIP" BCP,  (which Scott seems to be
> suggesting). 
> 
> 2) Leave the UDP stuff in place, but give people tools to keep it from
> being used (which this draft suggests).
> 
> 3) Rev the spec to 3.0 and rip UDP out by the toenails.
> 
> Any other ideas? What is the right thing to do here?
> 
> --
> Dean


You could always leave the current definition of SIP alone, and
define a new version of it (call it SIPNEW or something?) that
uses non-UDP transports to support e2e. Why deprecate the old
one if people find it useful for certain problems? You could just 
mandate that old SIP cannot be used in e2e environments, and that
if e2e is ever required for your application, you need to use
the newer SIP.

I've long been of the opinion that there needs to be two flavors of
SIP, one for secured proprietary networks, and one for the internet
at large. (Don't flame me! I understand the current mandate of SIP.
I'm just saying it might be useful.) The motivation for the simpler
SIP would be to improve speed and cut down on message bloat.

Gordon


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Brett Tate wrote:
> 
>>AFAIK it is quite ok today to establish a dialog 
>>via a SUBSCRIBE, and then send an INVITE within 
>>that dialog to establish a new call.
> 
> Yes.  However many products would reject the invite
> unless it has meaning associated with the subscription.

Well, you are probably right about that, though there is no particular 
reason why that ought to be the case.

While it is probably outside the scope of the IETF, it seems plausible 
to me that a request arriving within an existing dialog ought to 
accepted if the exact same request would have been accepted in a new 
dialog. In addition, there MAY be cases where a request in a dialog 
might be accepted with less restrictions if in an existing dialog than 
if received in a new one.

Clearly, if interop problems arise due to differences in authorization 
policies, then the mechanism isn't useful. In that sense using separate 
dialogs and GRUUs may be easier. But that approach will probably also 
have problems of some sort - perhaps around who can use a GRUU or what 
for, GRUU lifetime, etc.

	Paul


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From: Volker Hilt <volkerh@bell-labs.com>
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Subject: Re: [Sip] GRUU Mechanism I-D
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The main issue with deprecating the implicit subscription seems to be 
that it must be done in a backwards compatible way, since a lot of 
implementations already exist. To my understanding, a backwards 
compatible solution would need to:
- enable the UAC to detect, whether the UAS is an "old" UA that uses the 
deprecated implicit subscriptions or a "new" UA that supports explicit 
subscriptions.
- enable the UAS to detect, whether the UAC expects implicit or explicit 
subscriptions.

Would a new option tag help (along the lines of Sean's draft) that is 
mandatory for new implementations? Btw. since explicit subscriptions 
require the UAS to use a gruu, the first case could be covered just by 
looking for the gruu option tag. However, I'm not sure if it is safe to 
assume that a "new" UAC also includes a gruu in its REFER request (in 
this case, we could even use the "gruu" option tag as an indicator).

-- Volker

-- 
Volker Hilt                      Bell Labs / Lucent Technologies
phone: +1 732 332 6432           101 Crawfords Corner Rd
email: volkerh@bell-labs.com     Holmdel, NJ 07733


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To: Gordon Ledgard <gledgard@iperia.com>
Cc: "'Dean Willis'" <dean.willis@softarmor.com>, sip@ietf.org,
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From: Scott Lawrence <slawrence@pingtel.com>
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> On Thu, 2003-11-06 at 16:12, Dean Willis wrote:
>> Here's the deal. Before long, EVERY e2e-secured SIP request or response
>> is going to be bigger than one packet. The security people aren't going
>> to give up on e2e.

Gordon Ledgard <gledgard@iperia.com> writes:

> You could always leave the current definition of SIP alone, and
> define a new version of it (call it SIPNEW or something?) that
> uses non-UDP transports to support e2e. Why deprecate the old
> one if people find it useful for certain problems? You could just 
> mandate that old SIP cannot be used in e2e environments, and that
> if e2e is ever required for your application, you need to use
> the newer SIP.

e2e is not required to cause the congestion-related bad behaviour of
SIP; large messages are not even required.  All that's required is
_something_ that causes packet loss on links that are already running
at a load close enough to capacity, somewhere in the path - after
that, the retransmissions push the network into collapse.

-- 
Scott Lawrence        
  Pingtel Corp.   


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Subject: Re: Consensus on direction needed (was Re: [Sip] Re:	draft-ietf-sip-congestsafe-02.txt)]
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> 3) Rev the spec to 3.0 and rip UDP out by the toenails.

For the historical record, that was roughly SIP 1.0 :-)

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Subject: RE: Consensus on direction needed (was Re: [Sip] Re:draft-ietf-sip-congestsafe-02.txt)]
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> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
> Dean Willis
> Sent: Thursday, November 06, 2003 11:11 PM
> To: sip@ietf.org; slawrence@pingtel.com
> Subject: Consensus on direction needed (was Re: [Sip]
> Re:draft-ietf-sip-congestsafe-02.txt)]
>=20
>=20
> Here's the deal. Before long, EVERY e2e-secured SIP request=20
> or response
> is going to be bigger than one packet. The security people=20
> aren't going
> to give up on e2e.
>=20
> This gives us two viable alternatives, as I see them:
>=20
> 1) Deprecate UDP usage in SIP.
>=20
> 2) Deprecate SIP.
>=20
> Since #2 would leave me unemployed :-(, the question I'm interested is
> how to do #1.
>=20
> Three possibilities suggest themselves, and I'd like to get a sense of
> consensus from the group on the direction (or other suggestions, if
> there are any).
>=20
> 1) Leave the UDP stuff in place, but convince people not to use it by
> doing a "UDP Considered Harmful for SIP" BCP,  (which Scott=20
> seems to be
> suggesting).=20
>=20
> 2) Leave the UDP stuff in place, but give people tools to keep it from
> being used (which this draft suggests).
>=20
> 3) Rev the spec to 3.0 and rip UDP out by the toenails.
>=20
> Any other ideas? What is the right thing to do here?

I prefer 1. I would not deprecate UDP completely, it has its uses. We =
can also extend option 1 so that the default transport protocol is TCP =
if the transport parameter is not present.

Regards,
Hisham

>=20
> --
> Dean
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Subject: RE: Consensus on direction needed (was Re: [Sip] Re: draft-ietf-s
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> Here's the deal. Before long, EVERY e2e-secured SIP request or response
> is going to be bigger than one packet. The security people aren't going
> to give up on e2e.

It would be nice if the security people could find an e2e solution better
suited to SIP's needs, then, I gather.

> This gives us two viable alternatives, as I see them:
> 
> 1) Deprecate UDP usage in SIP.
> 
> 2) Deprecate SIP.
> 
> Since #2 would leave me unemployed :-(, the question I'm interested is
> how to do #1.
> 
> Three possibilities suggest themselves, and I'd like to get a sense of
> consensus from the group on the direction (or other suggestions, if
> there are any).
> 
> 1) Leave the UDP stuff in place, but convince people not to use it by
> doing a "UDP Considered Harmful for SIP" BCP,  (which Scott seems to be
> suggesting). 
> 
> 2) Leave the UDP stuff in place, but give people tools to keep it from
> being used (which this draft suggests).
> 
> 3) Rev the spec to 3.0 and rip UDP out by the toenails.
> 

I'm afraid I remain closest to option 3, though this particular motivation
is far from my main reason reason for feeling so.

Jon Peterson
NeuStar, Inc.

> Any other ideas? What is the right thing to do here?
> 
> --
> Dean
> 

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From exim@www1.ietf.org  Fri Nov  7 03:06:26 2003
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Subject: [Sip] REGISTER requests via NAT
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Excuse me if this topic has been already discussed, I have not found an 
answer in the list archive.

The situation: an organization with 2 offices, each running its own NAT'ed 
network, with a SIP server installed on each NAT gateway. The SIP servers 
insert the Record-Route headers for all requests coming to and from their 
LANs, and they also do media stream proxing: the Record-Route mechanism 
allows user@network1.com to exchange SIP messages and to establish calls 
with any user@network2.com.

Now, the organization decides to register users of these 2 (or more) 
networks within one domain, so employees moving back and force between 
offices can be addressed by their name, user@company.com. The company.com 
SIP registrar is installed, let's say, "outside" both LANs.

The SIP servers in each office properly redirect the REGISTER requests to 
that company.com server. But the Contact: header fields, obviously, contain 
the internal LAN addresses of those users.

As the RFC3261 explicitly specifies that any Record-Route information in 
REGISTER requests should be ignored, is there any other, "legal" way to 
"attach" the routing information to those requests, so it is stored by the 
registrar and all SIP requests made to user@company.com from a different LAN 
(coming to the registrar proxy) will be routed correctly?

Sincerely,
Vladimir

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> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
> Scott Lawrence
> Sent: Thursday, November 06, 2003 4:34 PM
> To: sip@ietf.org
> Subject: [Sip] Re: draft-ietf-sip-congestsafe-02.txt

>=20
>     UAC     Proxy       Proxy     Proxy      UAC
>      A -----> B --------> C ------> D ------> E
>=20
>   UAC A sends an INVITE to E with 'Proxy-Require: congestion-managed',
>   initially handing it off to Proxy B.  Using normal SIP routing
>   mechanisms, that message finds itself in due course at Proxy D
>   (the local proxy on E's network).  The administrator of that
>   network has (for whatever reason) chosen to configure SIP on that
>   network to prefer (or even be limited to) UDP.  This potential
>   conflict raises two questions:
>=20
>    - Why should a user elsewhere in the network (A) be empowered to
>      exercise any control of routing policy within the local network
>      (E's network)?
>=20
>    - Why would A want to send a request using this mechanism, given
>      that doing so introduces a new potential failure mode, but does
>      not improve that chances that this individual request will
>      succeed?

Actually, Proxy D uses the transport protocol that E has indicated in =
its contact-header of REGISTER.

REGISTER
...
Contact: <sip:userb@client.domain2.com;transport=3Dtcp>

This forces D to forward the message to E using TCP.

Proxies B, C, and D can have NAPTR entries without a udp option.

Eg of B network NSPTR records (no UDP option):

;       order   pref   flags         service      regexp     replacement
IN NAPTR 50     50      "s"    "SIPS+D2T"         ""           =
_sips._tcp.domain1.com.
IN NAPTR 90     50      "s"    "SIP+D2T"          ""           =
_sip._tcp.domain1.com

It is true that is a collected effort in that all domains have to =
configure their networks as above.

This is the BCP I would propose.

Regards,
Hisham


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Subject: RE: Consensus on direction needed (was Re: [Sip] Re:draft-ietf-sip-congestsafe-02.txt)]
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Hi Dean,

Inline.

 > -----Original Message-----
 > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
 > Dean Willis
 > Sent: 06 November, 2003 23:11
 > To: sip@ietf.org; slawrence@pingtel.com
 > Subject: Consensus on direction needed (was Re: [Sip]
 > Re:draft-ietf-sip-congestsafe-02.txt)]
 >=20
 >=20
 >=20
 > On Thu, 2003-11-06 at 08:34, Scott Lawrence wrote:
 > >   This draft presents an excellent explanation of how and why the
 > >   use of UDP in SIP can (and therefor by Murphys Law, will=20
 > :-) cause
 > >   network operation problems.  In doing so, it makes a=20
 > good case for
 > >   SIP having been designed differently, possibly not using UDP at
 > >   all, but acknowleges that it is too late for that.  It goes on to
 > >   propose mechanisms by which this oversight can now be=20
 > corrected by
 > >   defining a congestion-safe mode of operation and adding=20
 > a protocol
 > >   feature to require operation in that mode.
 >=20
 > >  Unfortunately, I don't think that creating such a mechanism is
 > >   useful, because there is little or no incentive for any
 > >  significant number of systems to use it.  The cat is not only out
 > >  of the bag in that SIP was designed to support UDP, but later
 > >  standards (SIP use of NAPTR and DNSSRV in RFC 3263) have provided
 > >  for explicit administrative control mechanisms by which users
 > >  (administrators) can (perhaps unwisely) prefer the use of UDP.
 >=20
 > . . .=20
 >=20
 > Excellent summary and commentary.
 >=20
 > Here's the deal. Before long, EVERY e2e-secured SIP request=20
 > or response
 > is going to be bigger than one packet. The security people=20
 > aren't going
 > to give up on e2e.
 >=20
 > This gives us two viable alternatives, as I see them:
 >=20
 > 1) Deprecate UDP usage in SIP.
 >=20
 > 2) Deprecate SIP.
 >=20
 > Since #2 would leave me unemployed :-(, the question I'm=20
 > interested is
 > how to do #1.
 >=20
 > Three possibilities suggest themselves, and I'd like to get=20
 > a sense of
 > consensus from the group on the direction (or other suggestions, if
 > there are any).
 >=20
 > 1) Leave the UDP stuff in place, but convince people not to use it by
 > doing a "UDP Considered Harmful for SIP" BCP,  (which Scott=20
 > seems to be
 > suggesting).=20

Which UDP stuff?

My recollection of this problem and how this work on congest-safe came =
about in the first place is mainly that in RFC 3428, there are strict =
size limits for sending a message. These limits can be circumvented only =
by knowledge that the entire path that the message takes uses TCP (or =
other congestion controlled transport). Currently, there is no other way =
to know this, except by administration, and that only works in closed =
environments.=20

The problem that lead to this restriction as I remember it is that per =
RFC 3261 Ch 18.1.1., in the SIP transport layer, if the size of a =
message exceeds path MTU (or 1300 bytes when the path MTU for the next =
hop is unknown) the request MUST always be sent using TCP or similar =
congestion controlled protocol. But, there is an unfortunate exception =
to this rule in taking into consideration backwards compatibility with =
RFC 2543 implementations - if the TCP connection fails or is not there, =
the request may be retried using UDP.

I think what we really want to do is deprecate *this* behavior.
=20
 > 2) Leave the UDP stuff in place, but give people tools to=20
 > keep it from
 > being used (which this draft suggests).

Assuming the above is the "UDP stuff" then I agree. However, I agree =
that documenting the congestion problems in SIP in general is a good =
idea, and a BCP therefore makes sense as suggested in 1). =20
=20
 > 3) Rev the spec to 3.0 and rip UDP out by the toenails.

Well... ;)

 > Any other ideas? What is the right thing to do here?

I think we should separate these two things here. First, we have the =
general congestion-safetyness of SIP to worry about, and as people have =
already pointed out, a BCP discussing the options in mitigating those =
concerns is a good idea.=20

Secondly, we have the harmful "if-TCP-fails-then-fall-back-to-UDP" =
behavior, which we want to deprecate and in the mean time hack a fix =
around it.

Cheers,
Aki

 > --
 > Dean
 >=20
 > _______________________________________________
 > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > This list is for NEW development of the core SIP Protocol
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sipping@ietf.org for new developments on the application of sip
 >=20

_______________________________________________
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To: <aki.niemi@nokia.com>
Cc: <dean.willis@softarmor.com>, <sip@ietf.org>, <slawrence@pingtel.com>
References: <98C7D2E5BCD2374C9AAE0BCD2E9C1DD902D592A8@esebe013.ntc.nokia.com>
Organization: Pingtel Corp. http://www.pingtel.com/
From: Scott Lawrence <slawrence@pingtel.com>
Date: Fri, 07 Nov 2003 09:50:39 -0500
In-Reply-To: <98C7D2E5BCD2374C9AAE0BCD2E9C1DD902D592A8@esebe013.ntc.nokia.com> (aki
 niemi's message of "Fri, 7 Nov 2003 15:42:37 +0200")
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<aki.niemi@nokia.com> writes:

> Secondly, we have the harmful "if-TCP-fails-then-fall-back-to-UDP"
> behavior, which we want to deprecate and in the mean time hack a fix
> around it.

but it isn't just that - we have placed the control of which transport
is used _explicitly_ in the hands of the user.  Having done that, our
only recourse now as vendors is to make the TCP alternative more
attractive to them than UDP.

-- 
Scott Lawrence        
  Pingtel Corp.   


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From: "Vijay K. Gurbani" <vkg@lucent.com>
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To: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: Consensus on direction needed (was Re: [Sip] Re:	draft-ietf-sip-congestsafe-02.txt)]
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Dean Willis wrote:

> This gives us two viable alternatives, as I see them:
> 
> 1) Deprecate UDP usage in SIP.
> 
> 2) Deprecate SIP.

Deprecate SIP!! What'choo talkin' about, Willis?  (Sorry, always
wanted to say that :-) )

> Three possibilities suggest themselves
> 
> 1) Leave the UDP stuff in place, but convince people not to use it by
> doing a "UDP Considered Harmful for SIP" BCP,  (which Scott seems to be
> suggesting). 
> 
> 2) Leave the UDP stuff in place, but give people tools to keep it from
> being used (which this draft suggests).
> 
> 3) Rev the spec to 3.0 and rip UDP out by the toenails.

If we rev the spec to 3.0, can we guarantee that the only
change in SIP/3.0 will be UDP removal?  What if someone wants
to take this opportunity to put other behavior in the spec?  Do
we run the risk of ending up with a protocol that does not quite
resemble SIP/2.0.  I do not know what the criteria of IESG is
to rev up a protocol version.  The only instance of this
happening that I am actively aware of is the rev from HTTP/1.0
to HTTP/1.1.  But in that particular instance, backward
compatibility was preserved, I believe.

I think it may be too late to deprecating UDP completely; so
much deployed gear uses it.  I have also witnessed many
 >>new<< SIP implementations start out with only supporting UDP
with the (mistaken) belief that it will be easier to code.

> Any other ideas? What is the right thing to do here?

I like Aki's idea of selectively deprecating UDP-related
backwards compatibility issues in rfc3261 which haunt us
at this point, but still maintain UDP as a transport (albeit
not endorse it as the preferred transport, which it was in
rfc2543).

Cheers,

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


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From exim@www1.ietf.org  Fri Nov  7 14:30:26 2003
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From: Cullen Jennings <fluffy@cisco.com>
To: <sip@ietf.org>, Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: [Sip] GRUU, Contacts, and privacy
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When making anonymous calls, the contact should change on every call. For
example of if you don't do this ...  Bill makes an anonymous call from
Monika's phone to Hillary. Hillary, takes the call, is suppositious about
the location and makes note of the contact. Now Hillary calls Monika and
notes the contact. If the two contacts are the same, the privacy of the
original call is lost and Hillary knows the original call was from Monika's
phone. 

Unfortunately, it looks like the grid approach still reveals this
information - the grid on the two calls might be different but the core GRUU
is still the same. 

I have a hard time imagining a simple solution that allowed the UA to
generate the GRUUs and still have fast lookup for the proxy. It seems like a
way is needed for the UA to get many GRUU that it can use and that these
need to be anonymous and uncorelatable by third parties.

One possible solution for this is to send another Register after each call
to get a new GRUU that can be used. This would drive the text to change so
that re-registrations must result in a new GRUU - the current encryption
mechanism with salt would meet this. Presumable, the old GRUU's would
continue to work and be good as long as the contact that generated it was
still valid. 

Cullen




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Dean wrote:
> 
> Three possibilities suggest themselves, and I'd like to get a sense of
> consensus from the group on the direction (or other suggestions, if
> there are any).
> 
> 1) Leave the UDP stuff in place, but convince people not to use it by
> doing a "UDP Considered Harmful for SIP" BCP,  (which Scott seems to be
> suggesting). 
> 
> 2) Leave the UDP stuff in place, but give people tools to keep it from
> being used (which this draft suggests).
> 
> 3) Rev the spec to 3.0 and rip UDP out by the toenails.
> 
> Any other ideas? What is the right thing to do here?

I think there are really two cases here: case #1 between the UA
and its first proxy (where there is very little signaling traffic),
and case #2 between proxies (where there is lots more traffic).  

In case #1: one major problem is the impact TCP has on post-dial 
delay.  While it certainly can be fixed by a custom TCP stack in
the UA, this doesn't generally seem to be happening.  Lacking this, 
UDP is left as the preferred solution here.  Can such modified
parameters for TCP be suggested in a BCP?

In case #2: one problem is the impact TCP has due to head-of-line 
blocking.  This can be fixed with SCTP.  A BCP recommending this
would be good.

So it seems the best solution is a variation of #1, "UDP
Considered Harmful between SIP Proxies".

Another idea, not that I expect much traction.
If a protocol is to "ripped out by its toenails", how about TCP? 
It seems to be stuck in the middle, not perfect for either case.

Bill Marshall
wtm@research.att.com

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'William Marshall'" <wtm@research.att.com>, <sip@ietf.org>
Subject: RE: Consensus on direction needed (was Re: [Sip] Re: draft-ietf-sip-congestsafe-02.txt)]
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Bill said:
> I think there are really two cases here: case #1 between the 
> UA and its first proxy (where there is very little signaling 
> traffic), and case #2 between proxies (where there is lots 
> more traffic).  
> 
> In case #1: one major problem is the impact TCP has on post-dial 
> delay.  While it certainly can be fixed by a custom TCP stack 
> in the UA, this doesn't generally seem to be happening.  
> Lacking this, 
> UDP is left as the preferred solution here.  Can such 
> modified parameters for TCP be suggested in a BCP?

Isn't this pretty much resolved by connection-reuse and a persistent TCP
connection between UA and proxy? This seems to be even sweeter if TLS is
also used.

> In case #2: one problem is the impact TCP has due to head-of-line 
> blocking.  This can be fixed with SCTP.  A BCP recommending 
> this would be good.

Good point. Would it be adequate to put this discussion into the SIP-SCTP
doc,

http://www.softarmor.com/wgdb/docs/draft-ietf-sip-sctp-03.txt

Bill also said: 
> So it seems the best solution is a variation of #1, "UDP 
> Considered Harmful between SIP Proxies".
> 
> Another idea, not that I expect much traction.
> If a protocol is to "ripped out by its toenails", how about TCP? 
> It seems to be stuck in the middle, not perfect for either case.

There's also DCCP to consider.

--
Dean


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CC: "'William Marshall'" <wtm@research.att.com>, sip@ietf.org
Subject: Re: Consensus on direction needed (was Re: [Sip] Re: draft-ietf-sip-congestsafe-02.txt)]
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>>In case #2: one problem is the impact TCP has due to head-of-line 
>>blocking.  This can be fixed with SCTP.  A BCP recommending 
>>this would be good.
> 
> 
> Good point. Would it be adequate to put this discussion into the SIP-SCTP
> doc,
> 
> http://www.softarmor.com/wgdb/docs/draft-ietf-sip-sctp-03.txt
> 

Obligatory discussion reminder: From previous discussions, sometimes 
people assume that SCTP fixes head-of-line blocking if there is a slower 
downstream system for one session. It does not. There seem to be 
actually very few places in SIP-style systems where SCTP head-of-line 
blocking avoidance matters.

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To: Dean Willis <dean.willis@softarmor.com>
CC: sip@ietf.org, "Jain, Rajnish" <rajnishjain@xl.com>
Subject: Re: Consensus on direction needed (was Re: [Sip] Re: draft-ietf-sip-congestsafe-02.txt)]
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Dean Willis wrote:

>>In case #1: one major problem is the impact TCP has on post-dial 
>>delay.  While it certainly can be fixed by a custom TCP stack 
>>in the UA, this doesn't generally seem to be happening.  
>>Lacking this, 
>>UDP is left as the preferred solution here.  Can such 
>>modified parameters for TCP be suggested in a BCP?
> 
> Isn't this pretty much resolved by connection-reuse and a persistent TCP
> connection between UA and proxy? This seems to be even sweeter if TLS is
> also used.

Problem is that the spec is vague on reusing TCP connections.
Rohan has an I-D on connection resue and I had presented one
at the Atlanta IETF that extended that paradigm to longer lived
connections (i.e. beyond the dialog).  Let me see -- the I-D is
still in the archives: 
http://www.ietf.org/internet-drafts/draft-jain-sipping-persistent-conn-reqs-00.txt

If TCP becomes the dominant transport, the specification should
probably provide more direction on connection reuse and connection
longevity (especially with TLS in the picture).

Cheers,

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


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From exim@www1.ietf.org  Fri Nov  7 17:01:29 2003
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To: dean.willis@softarmor.com, sip@ietf.org
Subject: RE: Consensus on direction needed (was Re: [Sip] Re: draft-ietf-sip-congestsafe-02.txt)]
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Dean wrote:
> Bill said:
> > In case #1: one major problem is the impact TCP has on post-dial 
> > delay.  While it certainly can be fixed by a custom TCP stack 
> > in the UA, this doesn't generally seem to be happening.  
> > Lacking this, 
> > UDP is left as the preferred solution here.  Can such 
> > modified parameters for TCP be suggested in a BCP?
> 
> Isn't this pretty much resolved by connection-reuse and a persistent TCP
> connection between UA and proxy? This seems to be even sweeter if TLS is
> also used.

Persistent connections solves one of the TCP problems, the extra round-
trips (SYN-SYNACK-ACK) needed before the INVITE request can be sent.

But the other problem is when an INVITE request is dropped by
the network due to corruption on the transmission link (far more
likely on access links than on backbone connections).  TCP will wait
for a timeout on the sender before sending a query; that timeout
standrad value is too long for telephony-type services.

As an aside, the post-dial-delay issue also seems to be forcing 
IPsec for endpoints over TLS.

Bill Marshall
wtm@research.att.com

-----original message-----
From: "Dean Willis" <dean.willis@softarmor.com>
To: "'William Marshall'" <wtm@research.att.com>, <sip@ietf.org>
Subject: RE: Consensus on direction needed (was Re: [Sip] Re: draft-ietf-sip-congestsafe-02.txt)]
Date: Fri, 7 Nov 2003 15:27:34 -0600

Bill said:
> I think there are really two cases here: case #1 between the 
> UA and its first proxy (where there is very little signaling 
> traffic), and case #2 between proxies (where there is lots 
> more traffic).  
> 
> In case #1: one major problem is the impact TCP has on post-dial 
> delay.  While it certainly can be fixed by a custom TCP stack 
> in the UA, this doesn't generally seem to be happening.  
> Lacking this, 
> UDP is left as the preferred solution here.  Can such 
> modified parameters for TCP be suggested in a BCP?

Isn't this pretty much resolved by connection-reuse and a persistent TCP
connection between UA and proxy? This seems to be even sweeter if TLS is
also used.

> In case #2: one problem is the impact TCP has due to head-of-line 
> blocking.  This can be fixed with SCTP.  A BCP recommending 
> this would be good.

Good point. Would it be adequate to put this discussion into the SIP-SCTP
doc,

http://www.softarmor.com/wgdb/docs/draft-ietf-sip-sctp-03.txt

Bill also said: 
> So it seems the best solution is a variation of #1, "UDP 
> Considered Harmful between SIP Proxies".
> 
> Another idea, not that I expect much traction.
> If a protocol is to "ripped out by its toenails", how about TCP? 
> It seems to be stuck in the middle, not perfect for either case.

There's also DCCP to consider.

--
Dean

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From exim@www1.ietf.org  Fri Nov  7 22:42:34 2003
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To: "'Cullen Jennings'" <fluffy@cisco.com>, <sip@ietf.org>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'Paul Kyzivat'" <pkyzivat@cisco.com>, <oritl@microsoft.com>
Subject: RE: [Sip] GRUU, Contacts, and privacy
Date: Fri, 7 Nov 2003 19:41:44 -0800
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I would like to offer the suggestion that while privacy is an interesting
and useful property, it is not a necessity for many applications that might
otherwise use GRUU. I would go even further and state that is runs counter
to some applications that might want to use a globally routable and unique
URI. Part of the complexity of the grid approach is this requirement for
obfuscation/privacy. If you remove that requirement, it is entirely possible
to come up with a simpler solution. 


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Cullen
Jennings
Sent: Friday, November 07, 2003 11:29 AM
To: sip@ietf.org; Jonathan Rosenberg
Cc: Paul Kyzivat
Subject: [Sip] GRUU, Contacts, and privacy

    
When making anonymous calls, the contact should change on every call. For
example of if you don't do this ...  Bill makes an anonymous call from
Monika's phone to Hillary. Hillary, takes the call, is suppositious about
the location and makes note of the contact. Now Hillary calls Monika and
notes the contact. If the two contacts are the same, the privacy of the
original call is lost and Hillary knows the original call was from Monika's
phone. 

Unfortunately, it looks like the grid approach still reveals this
information - the grid on the two calls might be different but the core GRUU
is still the same. 

I have a hard time imagining a simple solution that allowed the UA to
generate the GRUUs and still have fast lookup for the proxy. It seems like a
way is needed for the UA to get many GRUU that it can use and that these
need to be anonymous and uncorelatable by third parties.

One possible solution for this is to send another Register after each call
to get a new GRUU that can be used. This would drive the text to change so
that re-registrations must result in a new GRUU - the current encryption
mechanism with salt would meet this. Presumable, the old GRUU's would
continue to work and be good as long as the contact that generated it was
still valid. 

Cullen




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From exim@www1.ietf.org  Sat Nov  8 01:36:33 2003
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hisham.khartabil@nokia.com wrote:

> I share Paul's view.  There are currently too many implementations
> supporting dialog reuse.

Well, the issue isn't so much as to whether its been implemented, as 
to whether it is being used. I would actually be surprised if there 
was a lot of usage of it, and in particular, interoperable usage of 
it, given the various problems we have identified with it since 
publication of rfc3261.

Can people comment on any real usage of dialog reuse they are aware 
of? I suspect REFER on an INVITE dialog is the main (if only) one...

Thanks,
Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Subject: [Sip] Re: GRUU, Contacts, and privacy
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Cullen Jennings wrote:

>     
> When making anonymous calls, the contact should change on every call. For
> example of if you don't do this ...  Bill makes an anonymous call from
> Monika's phone to Hillary. Hillary, takes the call, is suppositious about
> the location and makes note of the contact. Now Hillary calls Monika and
> notes the contact. If the two contacts are the same, the privacy of the
> original call is lost and Hillary knows the original call was from Monika's
> phone. 
> 
> Unfortunately, it looks like the grid approach still reveals this
> information - the grid on the two calls might be different but the core GRUU
> is still the same. 

True.

> 
> I have a hard time imagining a simple solution that allowed the UA to
> generate the GRUUs and still have fast lookup for the proxy. It seems like a
> way is needed for the UA to get many GRUU that it can use and that these
> need to be anonymous and uncorelatable by third parties.

In my original I-D on this subject, I had postulated the existence of 
a separate LEASE operation that would allow for explicit requests for 
a GRUU. It made sense to use this method for cases where allocation of 
gruus was decoupled from registrations. Here is an example of such a 
case. GRUUs obtained by a lease may very well require extra state in 
proxies, but that would be OK - the lease operation could be used 
under more exceptional circumstances.

> 
> One possible solution for this is to send another Register after each call
> to get a new GRUU that can be used. This would drive the text to change so
> that re-registrations must result in a new GRUU - the current encryption
> mechanism with salt would meet this. Presumable, the old GRUU's would
> continue to work and be good as long as the contact that generated it was
> still valid. 

Actually, if you *really* wanted privacy, this would still not be 
sufficient. The reason is that your domain is still there, for all to 
see. Indeed, so would the IP address in your media stream. You'd need 
a real privacy service (as described in rfc3323) which would 
presumably obscure everything.

-Jonathan R.


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


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Subject: Re: [Sip] Re: Consensus on direction needed
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Scott Lawrence wrote:

> <aki.niemi@nokia.com> writes:
> 
> 
>>Secondly, we have the harmful "if-TCP-fails-then-fall-back-to-UDP"
>>behavior, which we want to deprecate and in the mean time hack a fix
>>around it.
> 
> 
> but it isn't just that - we have placed the control of which transport
> is used _explicitly_ in the hands of the user.  Having done that, our
> only recourse now as vendors is to make the TCP alternative more
> attractive to them than UDP.

The spec is pretty clear on the behavior for large packets:

If a request is within 200 bytes of the path MTU, or if it is larger
    than 1300 bytes and the path MTU is unknown, the request MUST be sent
    using an RFC 2914 [43] congestion controlled transport protocol, such
    as TCP.


this is not leaving it the hands of the user at all. The problem, as 
Aki has pointed out, is the exception clause in the paragraph that 
follows. Indeed, I would go so far is to say that this isnt the 
problem either. Practically speaking, most implementations are rfc3261 
compliant by now - or so they say. The issue is that many 
implementations are rfc3261 compliant, but have chosen not to 
implement or support tcp. No doubt this is in part due to the 
perceived complexities of tcp (some of which are real - large scale 
connection management is hard). There is also the lack of 
specification on properly doing connection reuse, which we are also 
addressing.

As such, there are really two separate problems here. One is getting 
people to use tcp, because we think its better. The second is being 
able to verify, in any particular request, that it really is being 
sent over congestion controlled transports. Mostly this would be for 
rfc3428 implementations that want to send a big message.

Solving the first problem, is done by a combination of the bcp, along 
with connection-reuse getting wrapped, as Scott has suggested. Dean's 
draft addresses the second. I would anticipate that this mechanism 
would be used infrequently, if ever. Since messaging sessions is 
getting close to done, that would be our preferred mechanism for large 
messages anyway.

There isn't going to be a SIP/3.0 - certainly not one that isnt 
backwards compatible. We don't get that kind of luxury in protocol 
design, not for ones that achieve any amount of success at least. So, 
since it needs to be backwards compatible anyway, we are right back to 
where we started from, and thus there is no point in a rev of the spec 
number.

Thanks,
Jonathan R.


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


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See the Path header extension:
http://www.ietf.org/rfc/rfc3327.txt

-Jonathan R.

Vladimir A. Butenko wrote:

> Excuse me if this topic has been already discussed, I have not found an 
> answer in the list archive.
> 
> The situation: an organization with 2 offices, each running its own 
> NAT'ed network, with a SIP server installed on each NAT gateway. The SIP 
> servers insert the Record-Route headers for all requests coming to and 
> from their LANs, and they also do media stream proxing: the Record-Route 
> mechanism allows user@network1.com to exchange SIP messages and to 
> establish calls with any user@network2.com.
> 
> Now, the organization decides to register users of these 2 (or more) 
> networks within one domain, so employees moving back and force between 
> offices can be addressed by their name, user@company.com. The 
> company.com SIP registrar is installed, let's say, "outside" both LANs.
> 
> The SIP servers in each office properly redirect the REGISTER requests 
> to that company.com server. But the Contact: header fields, obviously, 
> contain the internal LAN addresses of those users.
> 
> As the RFC3261 explicitly specifies that any Record-Route information in 
> REGISTER requests should be ignored, is there any other, "legal" way to 
> "attach" the routing information to those requests, so it is stored by 
> the registrar and all SIP requests made to user@company.com from a 
> different LAN (coming to the registrar proxy) will be routed correctly?
> 
> Sincerely,
> Vladimir
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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


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From: "Vladimir A. Butenko" <vladimir_butenko@stalker.com>
Subject: Re: [Sip] REGISTER requests via NAT
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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On Sat, 08 Nov 2003 03:03:10 -0500
  Jonathan Rosenberg <jdrosen@dynamicsoft.com> wrote:
> See the Path header extension:
> http://www.ietf.org/rfc/rfc3327.txt
> 
> -Jonathan R.

Thank you very much, that's exactly whay I was looking for. Please forgive 
my ignorance :-(.


Sincerely,
Vladimir

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Subject: Re: Consensus on direction needed (was Re: [Sip] Re:
	draft-ietf-sip-congestsafe-02.txt)]
From: Cullen Jennings <fluffy@cisco.com>
To: William Marshall <wtm@research.att.com>,
        Dean Willis <dean.willis@softarmor.com>, <sip@ietf.org>
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On 11/7/03 1:57 PM, "William Marshall" <wtm@research.att.com> wrote:

> As an aside, the post-dial-delay issue also seems to be forcing
> IPsec for endpoints over TLS.
>

I'm curious about this - can you provide some more information and the type
of deployments where this is an issue? 


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Date: Sun, 09 Nov 2003 22:32:18 -0800
Subject: Re: [Sip] Certificates handling in SIP/TLS
From: Cullen Jennings <fluffy@cisco.com>
To: Johan Bilien <jobi@via.ecp.fr>, <sip@ietf.org>
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On 11/6/03 8:01 AM, "Johan Bilien" <jobi@via.ecp.fr> wrote:

> Hi,
> 
> I have a few questions regarding the certificates handling when using
> SIP over TLS:
> 
> . When the mutual authentication isn't used, should the TLS connection
> between the proxy and the UA be kept alive after the first
> registration ? 

Yes - and it it goes down, the UA should re-register. If you are behind many
types of firewalls or gasp worse yet a NAT, you will have to do this even if
you are not using TLS.

> Otherwise, how can the proxy contact the client, for
> example to forward an incoming INVITE: if the initial connection was
> closed, the UA will need a TLS listening connection, thus act as a TLS
> server and need a certificate.
> 
> . When using mutual authentication, to which identity should the UA
> certificate's CN refer ? To the FQDN of the UA, or to the SIP URI ?

It's really the SubjectAltname not the CN that should be used in SIP. This
should refer to the same FQDN as the proxy would be trying to connect to -
this basically means it needs to assert the hostname portion of of whatever
it put in it's contact. This is not really possible if you got your
hostname/IP via DHCP like nearly every device that registers does. I view
mutual TLS as very useful between proxies and to things like gateways or
voice mail servers but not really very practical to most devices where user
register. 

Cullen


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From: Johan Bilien <jobi@via.ecp.fr>
To: sip@ietf.org
Subject: Re: [Sip] Certificates handling in SIP/TLS
Message-ID: <20031110162032.GA29896@via.ecp.fr>
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On Sun, Nov 09, 2003, Cullen Jennings wrote:
> > . When the mutual authentication isn't used, should the TLS connection
> > between the proxy and the UA be kept alive after the first
> > registration ? 
> 
> Yes - and it it goes down, the UA should re-register. If you are behind many
> types of firewalls or gasp worse yet a NAT, you will have to do this even if
> you are not using TLS.

So the client should check continuously that the connection is up ? And
if the connection goes down, and the client does not see it for some
reason, the server will not be able to forward the next INVITE messages.
In some cases, it is not very convenient to have to maintain a
connection alive, for example in case of mobile IP.

On the other hand, having the UA being able to handle TLS-server
connections would allow to re-establish a closed connection in any case.
Using TLS session resumption, re-establishing the connection is very
fast. The need for a UA certificate has other motivations (see below).


> > . When using mutual authentication, to which identity should the UA
> > certificate's CN refer ? To the FQDN of the UA, or to the SIP URI ?
> 
> It's really the SubjectAltname not the CN that should be used in SIP. This
> should refer to the same FQDN as the proxy would be trying to connect to -
> this basically means it needs to assert the hostname portion of of whatever
> it put in it's contact. This is not really possible if you got your
> hostname/IP via DHCP like nearly every device that registers does. I view
> mutual TLS as very useful between proxies and to things like gateways or
> voice mail servers but not really very practical to most devices where user
> register. 

In the case of a UA, having the certificate point to the FQDM doesn't
make much sense to me. On the other hand, having a certificate pointing
to the SIP URI of the user would allow several interesting things:

  . Replace the HTTP Digest client-to-proxy authentication with a TLS
  mutual authentication.

  . Allow a user to switch from one device to another without having to
  use different certificates.

  . Use the same certificate for the signature of a Diffie-Hellman key
  exchange, in a scheme such a MIKEY [1], and maybe for instant
  messaging encryption and signature ...

In this case, the association user-hostname would still be provided in a
secured way through the contact header, in the TLS-encrypted REGISTER. 

[1] http://www.ietf.org/internet-drafts/draft-ietf-msec-mikey-07.txt

Best regards,
-- 
Johan Bilien

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Hi Hisham,

Sorry for the delay in answering. Inline-

ext hisham.khartabil@nokia.com wrote:
> Apologies if some of these comments below have already been made on
> the list. There are no major comments.
> 
> - General: - I think a section should be dedicated to etags. This
> section would explain how they work. At the moment, it is all over
> the place.

This is probably a good idea. I will try to address this in the next rev.

> - Abstract: - The 1st sentence in the second paragraph repeats what
> is said in the first paragraph.

Hmm... I don't quite see how, but perhaps you can suggest a better wording?

> - Section 1: - The 1st sentence in the 3rd paragraph repeats what is
> said in the first paragraph.

I think it is sort of elaborating on the text in the 1st paragraph, but 
I will change it if you can suggest some better wording.

> - Section 3: - Second paragraph repeats definitions section. Maybe
> this paragraph can go away and references added to definitions
> section.

True. How about this: I remove references to presence from the 
definitions and that way chapter 3 adds value in describing how these 
components relate to presence.

> - Section 4.3: - Why is it a MUST for the content-type to provide
> partial publication? I think MAY is enough

Ben commented this before, and this is mainly a presence related thing 
anyway. Some event packages may have no use for such a thing. I'll try 
to reword this so that what is meant here is more in terms of allowing 
more than one EPA at a time.

Then, if that is the case, the content-type may also need to support 
this explicitly.

> - Section 5: - The note: Why would a PUBLISH fork? If would only fork
> if the user has explicitly registered 2 different addresses or the
> proxy is configured with 2 ESCs. In any case, I don't get why the
> PUBLISH may not be appropriately presented. I think this note can go
> away. - PUBLISH does not create a dialog. Can we have some text here
> why this is so. I just to remind us in 2 years time when we start
> asking the question again. - Why is it explicitly forbidden for a
> PUBLISH to contain a contact. It could be as simple as saying that
> the ESC ignores the contact header.

This note is about an OPTIONS forking, in which case the 200 OK may not 
be from an ESC, and thus lack support for the PUBLISH method.

> - Section 5.2: - Request-URI should be used instead of To-header 

Correct, I missed this while editing.

> - It
> should be clarified that an etag associates an EPA with the ESC for
> that boot cycle or until the EPA removes a publication (or something
> like that) and does not change for that duration. (I was momentarily
> confused and thought that the etag changes with every PUBLISH from
> the same EPA). - I think I understand the etag mechanism, but is
> there a chance that a state already exists for a EPA before that
> initial PUBLISH (a crash and reboot or something). How does the ESC
> behave in this case? Does the ESC know it is the same ESC publishing
> and therefore should remove the state that was created before this
> new PUBLISH?

The ESC would not know, and would instead assign a new etag and treat 
this as an initial publication. The original publication would simply 
expire and be garbage-collected.

The etags are just receipts for the EPA on a given publication. 
Modifying, refreshing or removing that state is only possible upon 
presenting this receipt to the ESC.

I gather this sort of text needs to be in that chapter explaining the 
etags usage a bit closer.

> - Section 5.4 and 5.5: - "the 200 (OK) response to a PUBLISH request
> from the ESC contains ..." appears a few times. This confused me into
> thinking that you are talking about the 200 of the
> refresh/modification PUBLISH. I think you mean the "previous"
> PUBLISH.

Right. I need to clarify those passages a bit.

> - Section 5.5: - The note should be normative text.

It ought to be - in Chapter 6 which describes the ESC behavior.

> - Section 5.6 - "EPAs which support the PUBLISH method SHOULD support
> this mechanism for explicitly removing event state." Should the
> SHOULD be a MUST?

You're right. This is a MUST.

Thanks for the comments!

Cheers,
Aki

> Regards, Hisham



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


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From exim@www1.ietf.org  Mon Nov 10 12:11:28 2003
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I'll be posting the slides for our meeting at 

http://www.softarmor.com/sipwg/meets/ietf58/slides/


If you're using slides in the meeting, please send them to me -- preferably
IN ADVANCE so that remote participants can download the materials. In any
case, get them to me by the end of the week if you want them in the
proceedings.


--
Dean


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Subject: Re: [Sip] Comments on history-info
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On Thursday, November 6, 2003, at 01:44 AM, Eric Burger wrote:
> Big question:
> Why don't we use multiple History-Info headers, instead of a 
> (potentially VERY long) single header?

Hey Eric,

These are semantically equivalent and implementations are supposed to 
support receiving BOTH syntaxes.

Just like:

Supported: 100rel, timer, replaces

is equivalent to

Supported: 100rel
Supported: timer
Supported: replaces

AND

Via: SIP/2.0/TCP sip1.example.com, SIP/2.0/TCP sip2.example.com, 
SIP/2.0/TCP sip3.example.com

is equivalent to

Via: SIP/2.0/TCP sip1.example.com
Via: SIP/2.0/TCP sip2.example.com
Via: SIP/2.0/TCP sip3.example.com

(branches omitted for clarity)

Bottom Line: The draft is using the correct BNF.

thanks,
-rohan

> I would argue for multiple headers on two grounds.
>
> First, the idea of a proxy modifying history information doesn't sound 
> smart.  If it screws up, it is hard to figure out who screwed up.
>
> Second, the index parameter frees us from worrying about header order, 
> which is a VERY good thing.
>
>
>
> Overview:
> While the motivation for History-Info is for the initiator of a 
> request to know what happened to a request, it is also of great use to 
> the recipient (End of Section 2 overview paragraph).
>
> Section 1.2:
> How is History-Info any different than Via w.r.t. security?
>
>
>
> Nits:
> Section 2.5, first paragraph, last sentence:
> this would like be a local
> this would likely be a local
>                ^^
>
> Appendix B, message F8, extra space before last angle-bracket in the 
> History-Info line.
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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From exim@www1.ietf.org  Mon Nov 10 14:33:33 2003
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Subject: [Sip] Reading List Addition for IETF 58 meeting, supplementary docs for Request History
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I've been asked to add two relevant documents to the reading list for the
Request History discussion. The amended agenda item is below.

1345 Request History
    Mary Barnes
    draft-ietf-sip-history-info-01
    draft-jennings-sip-voicemail-uri-00.txt
    RFC 3087

--
Dean


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From exim@www1.ietf.org  Mon Nov 10 18:29:27 2003
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Subject: RE: [Sip] Comments on history-info
Date: Mon, 10 Nov 2003 18:27:59 -0500
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Thread-Topic: [Sip] Comments on history-info
Thread-Index: AcOnsIEVSdJ5kcVgRCqNZzYScVDncwALJ6qQ
From: "Eric Burger" <eburger@snowshore.com>
To: "Rohan Mahy" <rohan@cisco.com>
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Whoops!  I should have looked closer at the ABNF.

Maybe we should include an example that has multiple lines.  Otherwise, =
we KNOW people will look at the examples, think everything has to go on =
a single line, and then barf when a fully conformant UA sends a response =
with multiple lines.

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: Mon, November 10, 2003 12:32 PM
> To: Eric Burger
> Cc: SIP List (E-mail)
> Subject: Re: [Sip] Comments on history-info
>=20
>=20
> On Thursday, November 6, 2003, at 01:44 AM, Eric Burger wrote:
> > Big question:
> > Why don't we use multiple History-Info headers, instead of a=20
> > (potentially VERY long) single header?
>=20
> Hey Eric,
>=20
> These are semantically equivalent and implementations are supposed to=20
> support receiving BOTH syntaxes.
>=20
> Just like:
>=20
> Supported: 100rel, timer, replaces
>=20
> is equivalent to
>=20
> Supported: 100rel
> Supported: timer
> Supported: replaces
>=20
> AND
>=20
> Via: SIP/2.0/TCP sip1.example.com, SIP/2.0/TCP sip2.example.com,=20
> SIP/2.0/TCP sip3.example.com
>=20
> is equivalent to
>=20
> Via: SIP/2.0/TCP sip1.example.com
> Via: SIP/2.0/TCP sip2.example.com
> Via: SIP/2.0/TCP sip3.example.com
>=20
> (branches omitted for clarity)
>=20
> Bottom Line: The draft is using the correct BNF.
>=20
> thanks,
> -rohan
>=20
> > I would argue for multiple headers on two grounds.
> >
> > First, the idea of a proxy modifying history information=20
> doesn't sound=20
> > smart.  If it screws up, it is hard to figure out who screwed up.
> >
> > Second, the index parameter frees us from worrying about=20
> header order,=20
> > which is a VERY good thing.
> >
> >
> >
> > Overview:
> > While the motivation for History-Info is for the initiator of a=20
> > request to know what happened to a request, it is also of=20
> great use to=20
> > the recipient (End of Section 2 overview paragraph).
> >
> > Section 1.2:
> > How is History-Info any different than Via w.r.t. security?
> >
> >
> >
> > Nits:
> > Section 2.5, first paragraph, last sentence:
> > this would like be a local
> > this would likely be a local
> >                ^^
> >
> > Appendix B, message F8, extra space before last=20
> angle-bracket in the=20
> > History-Info line.
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>=20
>=20
>=20


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Subject: RE: [Sip] GRUU, Contacts, and privacy
Date: Tue, 11 Nov 2003 05:54:04 +0200
Message-ID: <2038BCC78B1AD641891A0D1AE133DBB7017973AF@esebe019.ntc.nokia.com>
Thread-Topic: [Sip] GRUU, Contacts, and privacy
Thread-Index: AcOlZbZiztwxCjDXRrihtIlyWg0sZgCoZ4Ig
To: <fluffy@cisco.com>, <sip@ietf.org>, <jdrosen@dynamicsoft.com>
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How about reg event package? You get NOTIFY with new GRUU.

/Hisham

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
> Cullen Jennings
> Sent: Friday, November 07, 2003 9:29 PM
> To: sip@ietf.org; Jonathan Rosenberg
> Cc: Paul Kyzivat
> Subject: [Sip] GRUU, Contacts, and privacy
>=20
>=20
>    =20
> When making anonymous calls, the contact should change on=20
> every call. For
> example of if you don't do this ...  Bill makes an anonymous call from
> Monika's phone to Hillary. Hillary, takes the call, is=20
> suppositious about
> the location and makes note of the contact. Now Hillary calls=20
> Monika and
> notes the contact. If the two contacts are the same, the=20
> privacy of the
> original call is lost and Hillary knows the original call was=20
> from Monika's
> phone.=20
>=20
> Unfortunately, it looks like the grid approach still reveals this
> information - the grid on the two calls might be different=20
> but the core GRUU
> is still the same.=20
>=20
> I have a hard time imagining a simple solution that allowed the UA to
> generate the GRUUs and still have fast lookup for the proxy.=20
> It seems like a
> way is needed for the UA to get many GRUU that it can use and=20
> that these
> need to be anonymous and uncorelatable by third parties.
>=20
> One possible solution for this is to send another Register=20
> after each call
> to get a new GRUU that can be used. This would drive the text=20
> to change so
> that re-registrations must result in a new GRUU - the current=20
> encryption
> mechanism with salt would meet this. Presumable, the old GRUU's would
> continue to work and be good as long as the contact that=20
> generated it was
> still valid.=20
>=20
> Cullen
>=20
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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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: Mon, 10 Nov 2003 20:19:47 -0600
From: Alan Johnston <alan.johnston@mci.com>
Subject: RE: [Sip] GRUU, Contacts, and privacy
In-reply-to: <E1AIJyv-000063-00@ietf-mx>
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At 07:41 PM 11/7/2003 -0800, Sean Olson wrote:
>I would like to offer the suggestion that while privacy is an interesting
>and useful property, it is not a necessity for many applications that might
>otherwise use GRUU. I would go even further and state that is runs counter
>to some applications that might want to use a globally routable and unique
>URI. Part of the complexity of the grid approach is this requirement for
>obfuscation/privacy. If you remove that requirement, it is entirely possible
>to come up with a simpler solution.

I agree 100%.  A private Contact URI is just one small piece of the privacy 
puzzle.  I think this privacy requirement should be removed.

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com


>-----Original Message-----
>From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Cullen
>Jennings
>Sent: Friday, November 07, 2003 11:29 AM
>To: sip@ietf.org; Jonathan Rosenberg
>Cc: Paul Kyzivat
>Subject: [Sip] GRUU, Contacts, and privacy
>
>
>When making anonymous calls, the contact should change on every call. For
>example of if you don't do this ...  Bill makes an anonymous call from
>Monika's phone to Hillary. Hillary, takes the call, is suppositious about
>the location and makes note of the contact. Now Hillary calls Monika and
>notes the contact. If the two contacts are the same, the privacy of the
>original call is lost and Hillary knows the original call was from Monika's
>phone.
>
>Unfortunately, it looks like the grid approach still reveals this
>information - the grid on the two calls might be different but the core GRUU
>is still the same.
>
>I have a hard time imagining a simple solution that allowed the UA to
>generate the GRUUs and still have fast lookup for the proxy. It seems like a
>way is needed for the UA to get many GRUU that it can use and that these
>need to be anonymous and uncorelatable by third parties.
>
>One possible solution for this is to send another Register after each call
>to get a new GRUU that can be used. This would drive the text to change so
>that re-registrations must result in a new GRUU - the current encryption
>mechanism with salt would meet this. Presumable, the old GRUU's would
>continue to work and be good as long as the contact that generated it was
>still valid.
>
>Cullen
>
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use
>sip-implementors@cs.columbia.edu for questions on current sip Use
>sipping@ietf.org for new developments on the application of sip
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


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



From exim@www1.ietf.org  Wed Nov 12 15:02:16 2003
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Subject: [Sip] PUBLISH message rates
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To get the conversation started on the list, I'll summarize
what I heard as salient points being made at the microphone.

 - Event packages define SHOULD or MUST strength requirements
   that NOTIFYs cannot be sent faster than a hard limit.
   Aki's proposal can be summarized as: PUBLISH will inherit
   these hard limits as maximum rates of publication.

 - It was pointed out that recipients of PUBLISH requests can
   push back against excessive traffic with 5xx class responses
   that have "Retry-After" headers.

 - A point was made that NOTIFY will often be subject to filtering,
   in which case there is no one-to-one correlation between the
   amount of information sent to an event server in PUBLISH and
   the amount of information sent by an event server in NOTIFYs.

To be honest, after giving the topic a bit more thought, I'm
somewhat on the fence.

/a

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To amplify the third point: Since multiple entities may publish for the 
same presentity, for example, the NOTIFY rate may be as high as the sum 
of the PUBLISH rates. Thus, restricting PUBLISH cannot be relied upon to 
limit the NOTIFY rate, since the different PUBLISH sources presumably 
have no idea what others have been up to.

Thus, requiring some max. PUBLISH rate is not likely to be effective, 
but may be harmful in some (future) scenarios.

Adam Roach wrote:

> To get the conversation started on the list, I'll summarize
> what I heard as salient points being made at the microphone.
> 
>  - Event packages define SHOULD or MUST strength requirements
>    that NOTIFYs cannot be sent faster than a hard limit.
>    Aki's proposal can be summarized as: PUBLISH will inherit
>    these hard limits as maximum rates of publication.
> 
>  - It was pointed out that recipients of PUBLISH requests can
>    push back against excessive traffic with 5xx class responses
>    that have "Retry-After" headers.
> 
>  - A point was made that NOTIFY will often be subject to filtering,
>    in which case there is no one-to-one correlation between the
>    amount of information sent to an event server in PUBLISH and
>    the amount of information sent by an event server in NOTIFYs.
> 
> To be honest, after giving the topic a bit more thought, I'm
> somewhat on the fence.
> 
> /a
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From exim@www1.ietf.org  Wed Nov 12 16:14:48 2003
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From: "Ashir Ahmed" <a.ahmed@ntt.com>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>,
        "Adam Roach" <adam@dynamicsoft.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] PUBLISH message rates
Date: Thu, 13 Nov 2003 06:13:29 +0900
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Is there any ways that the PUA know about the watcher's request?
Why do I care?
I have a mobile phone ( a PUA). I don't want to pay money for PUBLISHing
unnecessary information. Besides the frequency of  PUBLISH, there is another
concern about the volume of  PUBLISH. So, if the PUA have the information
about the watcher's request (WHAT and when (the frequency) is being
subscribed), the PUA can respond accordingly.

ashir


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Henning
Schulzrinne
Sent: Thursday, November 13, 2003 5:25 AM
To: Adam Roach
Cc: 'sip@ietf.org'
Subject: Re: [Sip] PUBLISH message rates


To amplify the third point: Since multiple entities may publish for the
same presentity, for example, the NOTIFY rate may be as high as the sum
of the PUBLISH rates. Thus, restricting PUBLISH cannot be relied upon to
limit the NOTIFY rate, since the different PUBLISH sources presumably
have no idea what others have been up to.

Thus, requiring some max. PUBLISH rate is not likely to be effective,
but may be harmful in some (future) scenarios.

Adam Roach wrote:

> To get the conversation started on the list, I'll summarize
> what I heard as salient points being made at the microphone.
>
>  - Event packages define SHOULD or MUST strength requirements
>    that NOTIFYs cannot be sent faster than a hard limit.
>    Aki's proposal can be summarized as: PUBLISH will inherit
>    these hard limits as maximum rates of publication.
>
>  - It was pointed out that recipients of PUBLISH requests can
>    push back against excessive traffic with 5xx class responses
>    that have "Retry-After" headers.
>
>  - A point was made that NOTIFY will often be subject to filtering,
>    in which case there is no one-to-one correlation between the
>    amount of information sent to an event server in PUBLISH and
>    the amount of information sent by an event server in NOTIFYs.
>
> To be honest, after giving the topic a bit more thought, I'm
> somewhat on the fence.
>
> /a
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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] PUBLISH message rates
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Ashir Ahmed wrote:

> Is there any ways that the PUA know about the watcher's request?
> Why do I care?
> I have a mobile phone ( a PUA). I don't want to pay money for PUBLISHing
> unnecessary information. Besides the frequency of  PUBLISH, there is another
> concern about the volume of  PUBLISH. So, if the PUA have the information
> about the watcher's request (WHAT and when (the frequency) is being
> subscribed), the PUA can respond accordingly.

Unless you have a very special situation in mind, the number of watchers 
is typically large. You would have to compute the max. desired rate 
across all watchers and in addition take into account the filters they 
may have installed. Since you're paying, you should just update the 
presence information when you feel like it. After all, you're providing 
the service (typically) for free.

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



From exim@www1.ietf.org  Wed Nov 12 18:00:34 2003
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	Wed, 12 Nov 2003 14:53:39 -0000
From: "Christian Stredicke" <stredicke@snom.de>
To: <sip@ietf.org>
Date: Wed, 12 Nov 2003 16:59:45 -0600
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Hi There!

The discussion today about moving to TCP transport layer and the
consequences on NAT raises the question if we should define a header
that asks the proxy explicitly to keep the TCP connection open. As far
as I see this has not been defined yet. 

I see two possibilities:

1) Implicit: The proxy detects that client cannot be reached by the
address which it is using (using the received or rport parameter in the
top via). 

2) Explicit: The client explicitly requests this. This could be done
with the famous Connection header, maybe as a parameter in the top via.

In any case the behavior of the UAC as well as the proxy needs to be
(re)defined, I think the current TCP specification does not allow a
permanent connection.

Any thoughts on this?

CS
--
Christian Stredicke
tel:+49.30.39833.401 (ENUM & PSTN)
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From exim@www1.ietf.org  Wed Nov 12 22:30:34 2003
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To: Adam Roach <adam@dynamicsoft.com>
CC: "'sip@ietf.org'" <sip@ietf.org>
Subject: Re: [Sip] PUBLISH message rates
References: <9BF66EBF6BEFD942915B4D4D45C051F3E865FD@dyn-tx-exch-001.dynamicsoft.com>
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Are we assuming that any event package can automatically be used with 
publish? Or do we expect explicit documentation about how to use publish 
with a package? Maybe future event packages should document use of 
publish as part of the package documentation.

If we do expect package specific documentation for the use of PUBLISH, 
then it wouuld make sense to ask for it to specify the behavior just 
like we expect package documentation to do it. One option would be to 
state that the rules are inherited from the NOTIFY documentation. But 
for some packages, other approaches may make sense.

Adam Roach wrote:

> To get the conversation started on the list, I'll summarize
> what I heard as salient points being made at the microphone.
> 
>  - Event packages define SHOULD or MUST strength requirements
>    that NOTIFYs cannot be sent faster than a hard limit.
>    Aki's proposal can be summarized as: PUBLISH will inherit
>    these hard limits as maximum rates of publication.
> 
>  - It was pointed out that recipients of PUBLISH requests can
>    push back against excessive traffic with 5xx class responses
>    that have "Retry-After" headers.
> 
>  - A point was made that NOTIFY will often be subject to filtering,
>    in which case there is no one-to-one correlation between the
>    amount of information sent to an event server in PUBLISH and
>    the amount of information sent by an event server in NOTIFYs.
> 
> To be honest, after giving the topic a bit more thought, I'm
> somewhat on the fence.
> 
> /a
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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



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Date: Wed, 12 Nov 2003 22:14:18 -0600
From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Wireless Research and Development Group
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To: Christian Stredicke <stredicke@snom.de>
CC: sip@ietf.org, rajnishjain@xl.com
Subject: Re: [Sip] Connection: Keep-Alive
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Christian Stredicke wrote:

> Hi There!
> 
> The discussion today about moving to TCP transport layer and the
> consequences on NAT raises the question if we should define a header
> that asks the proxy explicitly to keep the TCP connection open. As far
> as I see this has not been defined yet. 

I've discussed this in a I-D I presented at the Atlanta IETF.
See
http://www.ietf.org/internet-drafts/draft-jain-sipping-persistent-conn-reqs-00.txt
on some ways to do so.  That I-D builds on Rohan's connection-reuse
draft and fills in a couple of gaps in it.  Namely the longevity
aspects of the connection and the neediness of a client in
requesting a persistent connection.

> I see two possibilities:
> 
> 1) Implicit: The proxy detects that client cannot be reached by the
> address which it is using (using the received or rport parameter in the
> top via). 
> 
> 2) Explicit: The client explicitly requests this. This could be done
> with the famous Connection header, maybe as a parameter in the top via.

Section 8 of that I-D describes some ways to explicitly signal
the intent.

> In any case the behavior of the UAC as well as the proxy needs to be
> (re)defined, I think the current TCP specification does not allow a
> permanent connection.
> 
> Any thoughts on this?

At the Atlanta IETF, it was decided that we will open a bug
against rfc3261 to provide more information on TCP connection
handling.  But as I said on the floor then, I believe that we
need to go beyond that; a BCP outlining TCP connection handling
was suggested as an additional venue.  I have not had the time
to do that so far.

Given the current and growing interest in this topic,
maybe we can get together some more heads interested in this
to hammer out a BCP.  We can use the above I-D as a starting
point.  The need for persistent connections will become
pressing as more implementations support TLS.  Despite TLS
session resumption, persistent TLS connections between two
high-volume proxies can be extremely beneficial.

Question to SIP/SIPPING chairs: Is the time ripe for a design
team on this?

Cheers,

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


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Vijay K. Gurbani wrote:


>> In any case the behavior of the UAC as well as the proxy needs to be
>> (re)defined, I think the current TCP specification does not allow a
>> permanent connection.
>>
>> Any thoughts on this?
> 
> 
> At the Atlanta IETF, it was decided that we will open a bug
> against rfc3261 to provide more information on TCP connection
> handling. 

Well, in particular, it was concluded that signaling "please keep this 
connection open" was not needed at all. Rather, the server should keep 
the connection open as long as it could, and only close it if it ran 
out of resources. Since there was no benefit to closing it prematurely 
(extra costs of setting it up again), and since signaling the desire 
for persistence could not be used to force the server to keep it open 
if the server ran out of resources, the signaling was not needed. 
Indeed, the observation was that many implementations in any case were 
doing this; it was just unclear in the spec.

-Jonathan R.

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


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To: Henning Schulzrinne <hgs@cs.columbia.edu>
CC: Adam Roach <adam@dynamicsoft.com>, "'sip@ietf.org'" <sip@ietf.org>
Subject: Re: [Sip] PUBLISH message rates
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Henning Schulzrinne wrote:

> To amplify the third point: Since multiple entities may publish for the 
> same presentity, for example, the NOTIFY rate may be as high as the sum 
> of the PUBLISH rates. Thus, restricting PUBLISH cannot be relied upon to 
> limit the NOTIFY rate, since the different PUBLISH sources presumably 
> have no idea what others have been up to.

I didnt think the intent was to use a limit on the PUBLISH rate to 
enforce the limit on the NOTIFY rate.

I think the point was that, if we believed there was value in limiting 
the NOTIFY rate (and we do, according to rfc3265), there is probably 
value in limiting the PUBLISH rate too.

I dont feel that strongly about this, since I think its a non-issue 
practically speaking until we have packages that really push limits. 
Before we get there, its mostly an academic exercise.

In that case, I am inclined to go for consistency, and have PUBLISH 
limited in rate as NOTIFYs are within a package.

-Jonathan R.

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


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Subject: RE: [Sip] Connection: Keep-Alive
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Actually, many implementations are not doing this unfortunately. A BCP is
not a bad idea. 

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Jonathan
Rosenberg
Sent: Wednesday, November 12, 2003 11:12 PM
To: Vijay K. Gurbani
Cc: Christian Stredicke; sip@ietf.org; rajnishjain@xl.com
Subject: Re: [Sip] Connection: Keep-Alive



Vijay K. Gurbani wrote:


>> In any case the behavior of the UAC as well as the proxy needs to be 
>> (re)defined, I think the current TCP specification does not allow a 
>> permanent connection.
>>
>> Any thoughts on this?
> 
> 
> At the Atlanta IETF, it was decided that we will open a bug against 
> rfc3261 to provide more information on TCP connection handling.

Well, in particular, it was concluded that signaling "please keep this
connection open" was not needed at all. Rather, the server should keep the
connection open as long as it could, and only close it if it ran out of
resources. Since there was no benefit to closing it prematurely (extra costs
of setting it up again), and since signaling the desire for persistence
could not be used to force the server to keep it open if the server ran out
of resources, the signaling was not needed. 
Indeed, the observation was that many implementations in any case were doing
this; it was just unclear in the spec.

-Jonathan R.

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


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


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Date: Thu, 13 Nov 2003 02:42:50 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: Re: [Sip] Connection: Keep-Alive
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Sorry for not being clear - I wasnt saying we DONT want a document. At 
the meeting today, I advocated (and I think many others agreed) for 
creation of a document. In particular, I think it makes sense to have 
a proposed standard document that describes all of the issues with 
getting tcp to work. This would presumably be an expansion of the 
existing connect-reuse draft. It needs to be PS since there is 
normative behavior associated with many parts of it.

-Jonathan R.

Sean Olson wrote:

> Actually, many implementations are not doing this unfortunately. A BCP is
> not a bad idea. 
> 
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Jonathan
> Rosenberg
> Sent: Wednesday, November 12, 2003 11:12 PM
> To: Vijay K. Gurbani
> Cc: Christian Stredicke; sip@ietf.org; rajnishjain@xl.com
> Subject: Re: [Sip] Connection: Keep-Alive
> 
> 
> 
> Vijay K. Gurbani wrote:
> 
> 
> 
>>>In any case the behavior of the UAC as well as the proxy needs to be 
>>>(re)defined, I think the current TCP specification does not allow a 
>>>permanent connection.
>>>
>>>Any thoughts on this?
>>
>>
>>At the Atlanta IETF, it was decided that we will open a bug against 
>>rfc3261 to provide more information on TCP connection handling.
> 
> 
> Well, in particular, it was concluded that signaling "please keep this
> connection open" was not needed at all. Rather, the server should keep the
> connection open as long as it could, and only close it if it ran out of
> resources. Since there was no benefit to closing it prematurely (extra costs
> of setting it up again), and since signaling the desire for persistence
> could not be used to force the server to keep it open if the server ran out
> of resources, the signaling was not needed. 
> Indeed, the observation was that many implementations in any case were doing
> this; it was just unclear in the spec.
> 
> -Jonathan R.
> 

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


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Subject: RE: [Sip] Connection: Keep-Alive
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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'Eric Burger'" <eburger@snowshore.com>, Rohan Mahy <rohan@cisco.com>
Cc: "SIP List (E-mail)" <sip@ietf.org>
Subject: RE: [Sip] Comments on history-info
Date: Mon, 10 Nov 2003 22:52:43 -0600
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Eric,

Thanks for your comments and reviewing the draft. I can add an example in
the next version.

Mary.

-----Original Message-----
From: Eric Burger [mailto:eburger@snowshore.com]
Sent: Monday, November 10, 2003 5:28 PM
To: Rohan Mahy
Cc: SIP List (E-mail)
Subject: RE: [Sip] Comments on history-info


Whoops!  I should have looked closer at the ABNF.

Maybe we should include an example that has multiple lines.  Otherwise, we
KNOW people will look at the examples, think everything has to go on a
single line, and then barf when a fully conformant UA sends a response with
multiple lines.

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: Mon, November 10, 2003 12:32 PM
> To: Eric Burger
> Cc: SIP List (E-mail)
> Subject: Re: [Sip] Comments on history-info
> 
> 
> On Thursday, November 6, 2003, at 01:44 AM, Eric Burger wrote:
> > Big question:
> > Why don't we use multiple History-Info headers, instead of a 
> > (potentially VERY long) single header?
> 
> Hey Eric,
> 
> These are semantically equivalent and implementations are supposed to 
> support receiving BOTH syntaxes.
> 
> Just like:
> 
> Supported: 100rel, timer, replaces
> 
> is equivalent to
> 
> Supported: 100rel
> Supported: timer
> Supported: replaces
> 
> AND
> 
> Via: SIP/2.0/TCP sip1.example.com, SIP/2.0/TCP sip2.example.com, 
> SIP/2.0/TCP sip3.example.com
> 
> is equivalent to
> 
> Via: SIP/2.0/TCP sip1.example.com
> Via: SIP/2.0/TCP sip2.example.com
> Via: SIP/2.0/TCP sip3.example.com
> 
> (branches omitted for clarity)
> 
> Bottom Line: The draft is using the correct BNF.
> 
> thanks,
> -rohan
> 
> > I would argue for multiple headers on two grounds.
> >
> > First, the idea of a proxy modifying history information 
> doesn't sound 
> > smart.  If it screws up, it is hard to figure out who screwed up.
> >
> > Second, the index parameter frees us from worrying about 
> header order, 
> > which is a VERY good thing.
> >
> >
> >
> > Overview:
> > While the motivation for History-Info is for the initiator of a 
> > request to know what happened to a request, it is also of 
> great use to 
> > the recipient (End of Section 2 overview paragraph).
> >
> > Section 1.2:
> > How is History-Info any different than Via w.r.t. security?
> >
> >
> >
> > Nits:
> > Section 2.5, first paragraph, last sentence:
> > this would like be a local
> > this would likely be a local
> >                ^^
> >
> > Appendix B, message F8, extra space before last 
> angle-bracket in the 
> > History-Info line.
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> 
> 


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From exim@www1.ietf.org  Thu Nov 13 10:24:39 2003
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Subject: Re: [Sip] Connection: Keep-Alive
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Jonathan Rosenberg wrote:
> Sorry for not being clear - I wasnt saying we DONT want a document. At 
> the meeting today, I advocated (and I think many others agreed) for 
> creation of a document. In particular, I think it makes sense to have a 
> proposed standard document that describes all of the issues with getting 
> tcp to work. This would presumably be an expansion of the existing 
> connect-reuse draft. It needs to be PS since there is normative behavior 
> associated with many parts of it.

Jonathan:

Thanks.  A document will go a long way.  My experience has been
that many existing implementations do not keep connections open
in any deterministic manner.

As the I-D I presented in Atlanta indicates, I still think
there is some amount of advantages to actually negotiating
a persistent connection instead of simply relying on server
implementations to keep the connection open and implement
a LRU algorithm to do connection garbage collection.  Clients
behind a NAT may want to signal to the server to keep the
connection alive.

But in the absence of a negotiation, I am happy to at least
have some work go into specifying connection management
and reuse techniques.

Will the co-chairs designate a design team, or should we
"go and do it".  The Atlanta I-D and discussion was a start
to "go and do it", but it'll be good to have more folks
involved with some recognition from the chairs.

Cheers,

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


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We should look at the motivation for the rate limitation. It can't be 
transport - that's a separate problem. There is some justification for 
limiting NOTIFYs since these are directed to devices that have a loose 
relationship to the PA. Also, these devices may have limited 
capabilities and may not be ready to receive 100 notifications/second.

However, in most cases of interest, the PA and the publisher are closely 
related. In addition, since the end system has complete control over the 
rate of sending PUBLISH, the 'protect low-end end systems' arguments 
doesn't apply.

I don't see why a spec should proscribe how often my UA and my PA can 
talk to each other. I'm not aware of any other Internet application 
protocol that makes such recommendations. We don't have a spec that says 
that you have to limit your SMTP rate to your outbound MTA to a certain 
value.

I'm philosophically opposed to "pulled out of thin air" recommendations 
that are ineffective and have no engineering basis.

Jonathan Rosenberg wrote:
> 
> 
> Henning Schulzrinne wrote:
> 
>> To amplify the third point: Since multiple entities may publish for 
>> the same presentity, for example, the NOTIFY rate may be as high as 
>> the sum of the PUBLISH rates. Thus, restricting PUBLISH cannot be 
>> relied upon to limit the NOTIFY rate, since the different PUBLISH 
>> sources presumably have no idea what others have been up to.
> 
> 
> I didnt think the intent was to use a limit on the PUBLISH rate to 
> enforce the limit on the NOTIFY rate.

But why else tie the two rates in any way?

> 
> I think the point was that, if we believed there was value in limiting 
> the NOTIFY rate (and we do, according to rfc3265), there is probably 
> value in limiting the PUBLISH rate too.
> 
> I dont feel that strongly about this, since I think its a non-issue 
> practically speaking until we have packages that really push limits. 
> Before we get there, its mostly an academic exercise.
> 
> In that case, I am inclined to go for consistency, and have PUBLISH 
> limited in rate as NOTIFYs are within a package.
> 
> -Jonathan R.
> 

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CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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Subject: Re: [Sip] PUBLISH message rates
References: <9BF66EBF6BEFD942915B4D4D45C051F3E865FD@dyn-tx-exch-001.dynamicsoft.com> <3FB2972C.7010608@cs.columbia.edu> <3FB33044.1000201@dynamicsoft.com> <3FB3AAF0.4070701@cs.columbia.edu>
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ext Henning Schulzrinne wrote:
> We should look at the motivation for the rate limitation. It can't be 
> transport - that's a separate problem. There is some justification for 
> limiting NOTIFYs since these are directed to devices that have a loose 
> relationship to the PA. Also, these devices may have limited 
> capabilities and may not be ready to receive 100 notifications/second.

Even this isn't solved using the default rate limit in notifications, 
because it will only apply to a single notifier. Typically, an endpoint 
has many active subscriptions, so the aggregate rate may still be 
considerable. The event-throttle work attempts to address this.

> However, in most cases of interest, the PA and the publisher are closely 
> related. In addition, since the end system has complete control over the 
> rate of sending PUBLISH, the 'protect low-end end systems' arguments 
> doesn't apply.

Right, it would mainly be about protecting the SIP routing fabric and 
the composer.

> I don't see why a spec should proscribe how often my UA and my PA can 
> talk to each other. I'm not aware of any other Internet application 
> protocol that makes such recommendations. We don't have a spec that says 
> that you have to limit your SMTP rate to your outbound MTA to a certain 
> value.
> 
> I'm philosophically opposed to "pulled out of thin air" recommendations 
> that are ineffective and have no engineering basis.

I guess these recommendations would anyway be "hand-waiving" with or 
without a hard rate limit recommendations. It's not like the rate could 
ever really be enforced in a meaningful way.

I think we can agree that documenting this issue is a good thing. Also, 
I would say explaining how a composer can use the SIP level push-back 
should be in the draft. Whether to have a concrete rate limit or not, 
I'm ok either way. I can suggest text that has a SHOULD strength 
recommendation that publishing EPAs pay attention to the rate at which 
they send updates.

Cheers,
Aki

> Jonathan Rosenberg wrote:
> 
>>
>>
>> Henning Schulzrinne wrote:
>>
>>> To amplify the third point: Since multiple entities may publish for 
>>> the same presentity, for example, the NOTIFY rate may be as high as 
>>> the sum of the PUBLISH rates. Thus, restricting PUBLISH cannot be 
>>> relied upon to limit the NOTIFY rate, since the different PUBLISH 
>>> sources presumably have no idea what others have been up to.
>>
>>
>>
>> I didnt think the intent was to use a limit on the PUBLISH rate to 
>> enforce the limit on the NOTIFY rate.
> 
> 
> But why else tie the two rates in any way?
> 
>>
>> I think the point was that, if we believed there was value in limiting 
>> the NOTIFY rate (and we do, according to rfc3265), there is probably 
>> value in limiting the PUBLISH rate too.
>>
>> I dont feel that strongly about this, since I think its a non-issue 
>> practically speaking until we have packages that really push limits. 
>> Before we get there, its mostly an academic exercise.
>>
>> In that case, I am inclined to go for consistency, and have PUBLISH 
>> limited in rate as NOTIFYs are within a package.
>>
>> -Jonathan R.
>>
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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References: <9BF66EBF6BEFD942915B4D4D45C051F3E865FD@dyn-tx-exch-001.dynamicsoft.com> <3FB2972C.7010608@cs.columbia.edu> <3FB33044.1000201@dynamicsoft.com> <3FB3AAF0.4070701@cs.columbia.edu> <3FB3B4CE.3080001@nokia.com>
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> Even this isn't solved using the default rate limit in notifications, 
> because it will only apply to a single notifier. Typically, an endpoint 
> has many active subscriptions, so the aggregate rate may still be 
> considerable. The event-throttle work attempts to address this.

True (and somewhat beyond the topic of this discussion). One could 
defend the rate limit in the NOTIFY case in arguing that at least the 
poor resource-constrained UA knows that it is subscribing to N events 
and thus can upperbound the event rate by summing them up. The PA can't, 
since it has generally no way of knowing how many entities will publish 
for a single event.


> 
>> However, in most cases of interest, the PA and the publisher are 
>> closely related. In addition, since the end system has complete 
>> control over the rate of sending PUBLISH, the 'protect low-end end 
>> systems' arguments doesn't apply.
> 
> 
> Right, it would mainly be about protecting the SIP routing fabric and 
> the composer.

Since it doesn't in any way limit the aggregate rate, this is a very 
loose way to protect anything. As noted, we don't limit the individual 
rate of sending email in the Internet by protocol spec to protect the 
aggregate MTA population. (Obviously, individual ISPs do sometimes apply 
rate limits for email submission to keep spammers out of their network, 
but that's a local policy decision, not a protocol decision.)



> I guess these recommendations would anyway be "hand-waiving" with or 
> without a hard rate limit recommendations. It's not like the rate could 
> ever really be enforced in a meaningful way.

We have Congress to pass meaningless declarations of good intent, so no 
need for the IETF to get into that business :-)

> 
> I think we can agree that documenting this issue is a good thing. Also, 
> I would say explaining how a composer can use the SIP level push-back 
> should be in the draft. Whether to have a concrete rate limit or not, 
> I'm ok either way. I can suggest text that has a SHOULD strength 
> recommendation that publishing EPAs pay attention to the rate at which 
> they send updates.

My basic test for any protocol spec normative statement is whether I can 
tell if somebody is violating the spec by observing external behavior. 
I'm unclear how I can measure 'paying attention' - I have a hard enough 
time with telling whether my students are ...



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From exim@www1.ietf.org  Thu Nov 13 14:09:29 2003
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Date: Thu, 13 Nov 2003 08:26:31 -0800
Subject: Re: [Sip] GRUU, Contacts, and privacy
From: Cullen Jennings <fluffy@cisco.com>
To: Adam Roach <adam@dynamicsoft.com>,
        "'Alan Johnston'" <alan.johnston@mci.com>, <seancolson@yahoo.com>,
        <sip@ietf.org>, Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Ok, I can live with that.

Cullen


> Alan Johnston [mailto:alan.johnston@mci.com] writes:
> 
>> I agree 100%.  A private Contact URI is just one small piece
>> of the privacy puzzle.  I think this privacy requirement should
>> be removed.


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ext Chris Boulton wrote:
> Ben, <<see comments>>
>=20
> -----Original Message----- From: Ben Campbell
> [mailto:bcampbell@dynamicsoft.com] Sent: Thu 13/11/2003 03:28 To:
> Adam Roach Cc: 'sip@ietf.org' Subject: Re: [Sip] PUBLISH message
> rates =20
>=20
> Are we assuming that any event package can automatically be used with
>  publish? Or do we expect explicit documentation about how to use
> publish with a package? Maybe future event packages should document
> use of publish as part of the package documentation.
>=20
> << I think that we have to assume that every event pacakage COULD use
> PUBLISH in some
>=20
> shape or form.  Even if initially the use does not seem obvious.>> =20

> If we do expect package specific documentation for the use of
> PUBLISH, then it wouuld make sense to ask for it to specify the
> behavior just like we expect package documentation to do it. One
> option would be to state that the rules are inherited from the NOTIFY
> documentation. But for some packages, other approaches may make
> sense.
>=20
> <<  I may be missing something BUT I don't see how this helps,
> especially in the case of
>=20
> large (and potentially  expanding) numbers of PA all Publishing at
> the same time.>> =20

It doesn't. That is a separate problem, and one way to address this=20
would be the SIP level push-back from the composer.

I'm inclined to agree with Henning on this. We could inherit the event=20
package specific limits, but in PUBLISH, they are even more useless. At=20
least in NOTIFYs, given the default rate of 1/x, the subscriber can=20
calculate the absolute aggregate rate of its y subscriptions: y * 1/x.

The composer doesn't have that possibility.

Cheers,
Aki

Adam Roach wrote:  > To get the conversation
> started on the list, I'll summarize > what I heard as salient points
> being made at the microphone. > >  - Event packages define SHOULD or
> MUST strength requirements >    that NOTIFYs cannot be sent faster
> than a hard limit. >    Aki's proposal can be summarized as: PUBLISH
> will inherit >    these hard limits as maximum rates of publication.=20
> > >  - It was pointed out that recipients of PUBLISH requests can >
> push back against excessive traffic with 5xx class responses >
> that have "Retry-After" headers. > >  - A point was made that NOTIFY
> will often be subject to filtering, >    in which case there is no
> one-to-one correlation between the >    amount of information sent to
> an event server in PUBLISH and >    the amount of information sent by
> an event server in NOTIFYs. > > To be honest, after giving the topic
> a bit more thought, I'm > somewhat on the fence. > > /a > >
> _______________________________________________ > Sip mailing list
> https://www1.ietf.org/mailman/listinfo/sip > This list is for NEW
> development of the core SIP Protocol > Use
> sip-implementors@cs.columbia.edu for questions on current sip > Use
> sipping@ietf.org for new developments on the application of sip   =20
> _______________________________________________ Sip mailing list
> https://www1.ietf.org/mailman/listinfo/sip This list is for NEW
> development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip=20
>=20
> J*fj)b?
> b??m????=0C0?'?~???f??f??X??)=DF=A3?"?8b?X??+=1F??DY=D7=AFzZ)????ay?+y"=
=0F>?-??%R=C7=AC????W?z{h??,r?n???y=DB=9F???z?b?{(?=CB=AB???*T??"????'?~?=
?~??{=07^??h?g???'?=17???bq?b?z=1Fsip=3D


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Date: Thu, 13 Nov 2003 15:11:12 -0800
Subject: Re: [Sip] Certificates handling in SIP/TLS
From: Cullen Jennings <fluffy@cisco.com>
To: Johan Bilien <jobi@via.ecp.fr>, <sip@ietf.org>
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On 11/10/03 8:20 AM, "Johan Bilien" <jobi@via.ecp.fr> wrote:

> On Sun, Nov 09, 2003, Cullen Jennings wrote:
>>> . When the mutual authentication isn't used, should the TLS connection
>>> between the proxy and the UA be kept alive after the first
>>> registration ? 
>> 
>> Yes - and it it goes down, the UA should re-register. If you are behind many
>> types of firewalls or gasp worse yet a NAT, you will have to do this even if
>> you are not using TLS.
> 
> So the client should check continuously that the connection is up ? And
> if the connection goes down, and the client does not see it for some
> reason, the server will not be able to forward the next INVITE messages.
> In some cases, it is not very convenient to have to maintain a
> connection alive, for example in case of mobile IP.

Yes I agree. 

> 
> On the other hand, having the UA being able to handle TLS-server
> connections would allow to re-establish a closed connection in any case.
> Using TLS session resumption, re-establishing the connection is very
> fast. The need for a UA certificate has other motivations (see below).
> 

This is good as long as you have stable address that is used to reach the
device that can be bound into the credentials of the device. This is
typically true in wireless systems. However, with IP phones or soft phones
often get their address via DHCP and do not have a stable IP address or DNS
domain name and this is unfortunately what is used to address them. I agree
that session resumption can greatly speed the set up.

> 
>>> . When using mutual authentication, to which identity should the UA
>>> certificate's CN refer ? To the FQDN of the UA, or to the SIP URI ?
>> 
>> It's really the SubjectAltname not the CN that should be used in SIP. This
>> should refer to the same FQDN as the proxy would be trying to connect to -
>> this basically means it needs to assert the hostname portion of of whatever
>> it put in it's contact. This is not really possible if you got your
>> hostname/IP via DHCP like nearly every device that registers does. I view
>> mutual TLS as very useful between proxies and to things like gateways or
>> voice mail servers but not really very practical to most devices where user
>> register. 
> 
> In the case of a UA, having the certificate point to the FQDM doesn't
> make much sense to me. On the other hand, having a certificate pointing
> to the SIP URI of the user would allow several interesting things:

I assume you mean the SIP AOR like sip:alice@example.com so you can do end
to end or middle to end security. I agree this is a good idea but I think
this what was being address by S/MIME stuff.

> 
> . Replace the HTTP Digest client-to-proxy authentication with a TLS
> mutual authentication.

Sure - see e2m stuff.

> 
> . Allow a user to switch from one device to another without having to
> use different certificates.
>

This does introduce the tricky problem of making sure both devices have the
same private key but yes.
 
> . Use the same certificate for the signature of a Diffie-Hellman key
> exchange, in a scheme such a MIKEY [1], and maybe for instant
> messaging encryption and signature ...
> 
> In this case, the association user-hostname would still be provided in a
> secured way through the contact header, in the TLS-encrypted REGISTER.
> 
> [1] http://www.ietf.org/internet-drafts/draft-ietf-msec-mikey-07.txt
> 
> Best regards,


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From exim@www1.ietf.org  Thu Nov 13 20:05:35 2003
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From: "Christian Stredicke" <stredicke@snom.de>
To: "Rohan Mahy" <rohan@cisco.com>
Cc: <sip@ietf.org>
Date: Thu, 13 Nov 2003 19:04:45 -0600
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Subject: [Sip] Dialog-State Use Cases
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Hi Rohan,

looks like we have just been bombed back into the requirements phase of
the dialog (or whatever the name is) package. 

I see the following use-cases:

1. STATUS information

We need to see if a dialog is "ringing", "talking", "on hold", or has
just been "terminated". Also it would be nice to have the to- and from,
maybe referred-by and the start of the "talking" (so that we can
calculate the duration). This information is nice for LED and
SWITCHBOARD applications. 


2. Call pickup, join, and other ACTIONS

The dialog state subscriber needs to know what URL to dial in order to
pick up a call or on how to join a call. Other actions might also be
indicated.


3. The question of HANDSET-LIFTING

We need to know if handsets have been lifted up (that's a simple
requirement from the switchboard people which laugh at us if we say
"impossible"; it is also needed for AUTOMATIC CALL BACK). We could model
this as dialog where "to" is not set yet.


That's all on my wish list! No need for CSeq or other stuff end-users
are not interested in!

Please don't forget SIP did not replace the good old PBX yet and there
is a reason for this.


CS


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To: Christian Stredicke <stredicke@snom.de>
CC: Rohan Mahy <rohan@cisco.com>, sip@ietf.org
Subject: Re: [Sip] Dialog-State Use Cases
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I think a lot of confusion could be eliminated by renaming the document 
and the event package it defines. While it has something to do with 
dialogs, it isn't fundamentally about dialogs. For instance, it has 
*nothing* to do with dialogs containing only subscriptions.

It is largely but not entirely related to dialogs that contain an 
invitation. But it is concerned with some things that can't be learned 
from just that.

I suggest something like "call-info".

More inline...

	Paul

Christian Stredicke wrote:
> 
> 3. The question of HANDSET-LIFTING
> 
> We need to know if handsets have been lifted up (that's a simple
> requirement from the switchboard people which laugh at us if we say
> "impossible"; it is also needed for AUTOMATIC CALL BACK). We could model
> this as dialog where "to" is not set yet.

It really need have nothing to do with lifting a handset. Lots of 
devices don't even have a handset - at least not one you lift.

I think it is about being in a state of preparing to make a call. But it 
is more than that, because not all devices would want to report this 
when preparing to make a call. It also has something to do with being 
unwilling or unable to receive a call while in this state, and/or 
wanting to seize a shared line resource.


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Subject: RE: [Sip] PUBLISH message rates
Date: Fri, 14 Nov 2003 05:50:04 +0200
Message-ID: <2038BCC78B1AD641891A0D1AE133DBB7017973D0@esebe019.ntc.nokia.com>
Thread-Topic: [Sip] PUBLISH message rates
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Cc: <jdrosen@dynamicsoft.com>, <adam@dynamicsoft.com>, <sip@ietf.org>
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I agree with Henning. If there is ever 1 subscriber and 1 publisher, =
then this rate limiting makes sense. But in many cases, especially =
presence, there are n publishers.

If there are 5 publishers for a package that limits the rate to 5 per =
second. There will be 20 NOTIFYs and 100 PUBLISHs sent in one minute . =
How does that help?

The only way to do this right is to do what RTCP does with report =
interval, and that is to adjust the rate of PUBLISH using some formula =
that takes into account the number of publishers at a given time. The =
more publishers that join, the less the rate at which PUBLISH is sent =
at. The compositors need to communicate that calculated rate to the =
publishers somehow. Do we want to make it that complicated? If not, then =
I suggest adding text the says if the compositor is overloaded, it can =
send a 5xx response with retry-after.

Regards,
Hisham

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
> Henning Schulzrinne
> Sent: Thursday, November 13, 2003 7:01 PM
> To: Niemi Aki (NMP/Helsinki)
> Cc: Jonathan Rosenberg; Adam Roach; 'sip@ietf.org'
> Subject: Re: [Sip] PUBLISH message rates
>=20
>=20
> > Even this isn't solved using the default rate limit in=20
> notifications,=20
> > because it will only apply to a single notifier. Typically,=20
> an endpoint=20
> > has many active subscriptions, so the aggregate rate may still be=20
> > considerable. The event-throttle work attempts to address this.
>=20
> True (and somewhat beyond the topic of this discussion). One could=20
> defend the rate limit in the NOTIFY case in arguing that at least the=20
> poor resource-constrained UA knows that it is subscribing to N events=20
> and thus can upperbound the event rate by summing them up.=20
> The PA can't,=20
> since it has generally no way of knowing how many entities=20
> will publish=20
> for a single event.
>=20
>=20
> >=20
> >> However, in most cases of interest, the PA and the publisher are=20
> >> closely related. In addition, since the end system has complete=20
> >> control over the rate of sending PUBLISH, the 'protect low-end end=20
> >> systems' arguments doesn't apply.
> >=20
> >=20
> > Right, it would mainly be about protecting the SIP routing=20
> fabric and=20
> > the composer.
>=20
> Since it doesn't in any way limit the aggregate rate, this is a very=20
> loose way to protect anything. As noted, we don't limit the=20
> individual=20
> rate of sending email in the Internet by protocol spec to protect the=20
> aggregate MTA population. (Obviously, individual ISPs do=20
> sometimes apply=20
> rate limits for email submission to keep spammers out of=20
> their network,=20
> but that's a local policy decision, not a protocol decision.)
>=20
>=20
>=20
> > I guess these recommendations would anyway be=20
> "hand-waiving" with or=20
> > without a hard rate limit recommendations. It's not like=20
> the rate could=20
> > ever really be enforced in a meaningful way.
>=20
> We have Congress to pass meaningless declarations of good=20
> intent, so no=20
> need for the IETF to get into that business :-)
>=20
> >=20
> > I think we can agree that documenting this issue is a good=20
> thing. Also,=20
> > I would say explaining how a composer can use the SIP level=20
> push-back=20
> > should be in the draft. Whether to have a concrete rate=20
> limit or not,=20
> > I'm ok either way. I can suggest text that has a SHOULD strength=20
> > recommendation that publishing EPAs pay attention to the=20
> rate at which=20
> > they send updates.
>=20
> My basic test for any protocol spec normative statement is=20
> whether I can=20
> tell if somebody is violating the spec by observing external=20
> behavior.=20
> I'm unclear how I can measure 'paying attention' - I have a=20
> hard enough=20
> time with telling whether my students are ...
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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Date: Thu, 13 Nov 2003 21:51:56 -0600
From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sip] Draft SIP Minutes for IETF 58
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Please review the follwing and correct where needed.

Master copy posted at:
http://www.softarmor.com/sipwg/meets/ietf58/notes/minutes-sipwg-ietf58.ht=
ml

Text follows:
----------------------------------

Draft Minutes, SIP WG, IETF 58
Official Scribe, Ben Campbell
Supplementary Scribe, Vijay Gurbani
Minutes Edited by Dean Willis


Topic: Call to Order and Agenda Bash
---------------------------------------

No issues raised.

Topic: Status, Chairs
---------------------

Slides presented.

New milestones proposed. No schedule objections raised, but it was=20
proposed that the app-interaction framework be published as a proposed=20
standard instead of BCP. Concerns about MIB status raised. Noted that=20
SIP should certainly plan to operate at least through the current=20
work-elevation plan from SIPPING, which extends through December 2004.
Topic: Congestion Safety,  Dean Willis
---------------------------------------
Reading list:
    draft-ietf-sip-congest-safe-02

There appears to be little interest in implementing this draft.  Some=20
suggestions for simplifying it were made. It was also suggested that we=20
need to change RFC 3263to prefer TCP over UDP, explain congestion issues=20
and avoidance in a BCP, and complete the connection reuse work in order=20
to encourage TCP deployment. Many endpoints now implement only UDP. We=20
also need to differentiate fragmentation and congestion. It was noted=20
that this problem does not just apply to MESSAGE, but can be frequently=20
expected with NOTIFY, and with INVITES using S/MIME protection. DCCP is=20
only a partial solution. Several people volunteered to work on this=20
proposed BCP.

It was also suggested that the current draft could be simplified to=20
eliminate new response codes 515 and 516, providing a minimal "NO UDP"=20
assurance.


Topic: RFC 3312 Updates, Gonzalo Camarillo
Reading list:
     draft-camarillo-sip-rfc3312-update-00.txt

Slides presented.

Suggested that we adopt as WG item. No objections were raised.

Conclusion: adopt draft as wg item, negotiate milestone.



Topic: Publish, Aki Niemi
Reading List:
     draft-ietf-sip-publish-01

Slides presented.

Issue: Should etags be updated on every publish? Consensus: Usage should=20
be made consistent with HTTP. If it cannot be, we should call these=20
something else.

Issue: Publication rate. Proposed that rate from event package be used.=20
Noted that this is probably an irrelevant and meaningless number, as=20
there may be multiple publishers with no knowledge of each other. Much=20
debate ensued, with no resolution  Further discussion deferred to the=20
mailing list.



Topic:  Request History, Mary Barnes
Reading List:
     draft-ietf-sip-history-info-01
     draft-jennings-sip-voicemail-uri-00.txt
     RFC 3087

Slides presented.

Issue: r-uri captured anytime request is forwarded. Proposal: Only add=20
reason for history entries added due to retargeting.

Issue: Privacy. Proposal: information not included when there are=20
privacy considerations. (when uas sets session or header level privacy) .

Next steps: complete flow detail. Do we want to fold reqs into doc as=20
front matter? Request more mailing list feedback. Dependency on mid-end=20
security draft.
Presentation of applicability to voicemail.

Issue: Optionality. Suggested that we apply stronger language to suggest=20
applications gracefully degrade in absence of history.

Issue: Implementability: Concerns raised by Robert Sparks, who will=20
detail on mailing list.

Issue: Security. Note that this in-general seems to depend on the=20
middle-to-end and end-to-middle security work.

Conclusion: Work should continue, but more thinking about optionality is=20
needed. RJS and Mary to talk offline.



1400 Non-Invite Transactions,   Robert Sparks
Reading list:
     draft-sparks-sip-noninvite-01.txt

Slides presented.

Noted that this problem is not related to the transport protocol, but is=20
specific to the SIP-layer transaction reliability mechanism for all=20
transactions other than INVITE.

Several alternatives were presented:

A) a series of tweaks
B) Eliminate Timer F, allowing non-invite transactions to pend.
C) USe path-timing estimation techniques to improve UAs knowledge of timi=
ng.
D) Revise 3263 caching language to reduce severity of impact.
E) Ignore the problem

Suggested that SIP be revised to use 3-way handshake like INVITE on all=20
transactions. This might be made backward-compatible through use of an=20
option tag.

Polls indicated  no consensus on any action, and that the working group=20
didn't understand the nature of problem or disagreed as to its severity.=20
Despite the objections of Morton Thiokol engineers, the shuttle was=20
launched in cold-weather conditions under which it had never been=20
tested, and the O-rings in the boosters failed.

Conclusion: None. Further discussion deferred to the mailing list.


Topic: GRUU, Jonathan Rosenberg
Reading list:
     draft-rosenberg-sip-gruu-00.txt
     draft-rosenberg-sipping-gruu-reqs-01.txt

Slides presented.

Issue: GRUU generation for stateless proxy behavior. Consensus that we=20
need a sample algorithm for understanding.

Issues: Liftime of a GRUU. Can a gruu change during a registration?=20
Conclusion: No, gruu is good for lifetime of registrations, refreshes=20
get same gruu.
Do we need a guarantee of difference between registrations? Probably no,=20
discuss on list.

Issue: Dialog reuse=97no longer a need with gruu. Should gruu spec=20
deprecate dialog resuse if both peers support gruu? (Impacts refer,=20
other subscribe usage).
Conclusion: Do not address in gruu spec, address in other drafts, put=20
pressure on future work to avoid dialog resuse.

Issue: Does Gruu interferes with e2e signaling. UA can try to generate a=20
local gruu, could use ice-like mechaninsm to decide if it is reachable.=20
Propose to add to draft, but never try to put local address in contact=20
header without using the mechanism.  Chair suggestion: Put statement=20
about not using local gruu in gruu draft, put =93iceing=94 in separate=20
draft. Author agreed.

Issue:  MUST NOT use locally generated gruu is too strong. Clarified=20
that definition of local gruu is one with a different domain part than=20
AoR, MUST does not apply to the examples given.


Topic: Join,  Rohan Mahy
Reading List:
     draft-ietf-sip-join-02.txt

Slides presented.

Noted that no comments in wglc. Room shows interest, but no one has=20
implemented.
We want more list discussion before sending to IESG.



Topic: REFER Semantics, Rohan Mahy
Reading List:
     draft-olson-sipping-refer-extensions
     draft-mahy-sip-remote-cc

Slides presented.

Basic premise is that the semantics of  the various uses of REFER are=20
under-specified. Much discussion ensued.


Issue: Is this equivalent to specifying fixed services? If we have to=20
specify all the services/features, then the refer approach has failed.

Issue: How do you put refer requests in a context? Dialog reuse?=20
Separate GRUUs? Explicit refer-to header parameters?  Noted that it may=20
require per-scheme semantics in addition to context.

Issue: Need more than context=97how do you know what an endpoint will do=20
with a particular URI scheme?

Issue: Not useful for authorization, because referee cannot decide if=20
issuing the request could be bad. Another motivation that is relevant is=20
to determine how to render the UI for the action.

Sugested: this does not mean fixing refer, it means adding something new=20
to it.

Discussion on proceeding: How do we proceed? Refer for other than=20
transfer unlikely to work on today=92s UAs. Rohan suggests defining=20
semantics for baskets of functionality. Use option tags to make sure=20
they are supported. Propose explicit dialog parameters for refer-to.=20
Provide guidance for remote call control vs. remote UI invocation.

No conclusion, further discussion deferred to list. Interested parties=20
are to contact Rohan and work on it.


Final To-Do:
-------------

App Interaction Framework -- change from BCP to PS.
Add milestone for RFC 3312 update as PS.
Discuss publication rate on mailing list.


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: Re: [Sip] PUBLISH message rates
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hisham.khartabil@nokia.com wrote:

> The only way to do this right is to do what RTCP does with report
> interval, and that is to adjust the rate of PUBLISH using some
> formula that takes into account the number of publishers at a given
> time. The more publishers that join, the less the rate at which
> PUBLISH is sent at. The compositors need to communicate that
> calculated rate to the publishers somehow. Do we want to make it that
> complicated? If not, then I suggest adding text the says if the
> compositor is overloaded, it can send a 5xx response with
> retry-after.

And that probably assumes some reasonably uniform generation pattern for 
the events. The whole point of events is that they are unpredictable, so 
that rate smoothing would have to be applied over longish time 
intervals, further complicating the whole system. Unlike for filters 
that are subscriber-specific, the publishers would also have to guess 
what notifications are droppable - which they can only do if they know 
what kind of things the subscribers care about most, even assuming that 
they have uniform tastes.

I can see that for some continuous-change events, this is at least 
plausible. For geospatial notifications that have change thresholds, I 
can easily compute the minimum displacement threshold across subscribers 
and adjust the publication rate accordingly. This would not be a simple 
rate limit, as the rate would depend on my movement pattern (slow if 
circling the block, fast if driving at highway speeds). Thus, you'd 
almost want to feed back some kind of 'distilled' subscription to the 
publisher - I would find that far more useful than some hard 
'messages/second' limit.




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



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From: "Christian Stredicke" <stredicke@snom.de>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
Cc: "'Rohan Mahy'" <rohan@cisco.com>, <sip@ietf.org>
Subject: AW: [Sip] Dialog-State Use Cases
Date: Fri, 14 Nov 2003 00:35:07 -0600
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> I suggest something like "call-info".
> 
Absolutely!

> > 3. The question of HANDSET-LIFTING
> >
> > We need to know if handsets have been lifted up (that's a simple
> > requirement from the switchboard people which laugh at us if we say
> > "impossible"; it is also needed for AUTOMATIC CALL BACK). We could
model
> > this as dialog where "to" is not set yet.
> 
> It really need have nothing to do with lifting a handset. Lots of
> devices don't even have a handset - at least not one you lift.
> 
> I think it is about being in a state of preparing to make a call. But
it
> is more than that, because not all devices would want to report this
> when preparing to make a call. It also has something to do with being
> unwilling or unable to receive a call while in this state, and/or
> wanting to seize a shared line resource.

Call it "virtual handset lifting"... I think we are in agreement that we
need something to indicate inability to receive a call while in this
state.

Christian


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From: "Christian Stredicke" <stredicke@snom.de>
To: <hisham.khartabil@nokia.com>, <hgs@cs.columbia.edu>, <aki.niemi@nokia.com>
Cc: <jdrosen@dynamicsoft.com>, <adam@dynamicsoft.com>, <sip@ietf.org>
Subject: AW: [Sip] PUBLISH message rates
Date: Fri, 14 Nov 2003 06:07:35 -0600
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> If there are 5 publishers for a package that limits the rate to 5 per
> second. There will be 20 NOTIFYs and 100 PUBLISHs sent in one minute .
How
> does that help?

Limiting the rate is about statistics. There is definitely a correlation
between the rate limitation and the actual rate. In other words: it has
an effect. Ok you can't make any guarantees on the actual performance,
but it helps to reduce the network load.

In the above example, we are talking about say 1000 messages without
limitation and 100 with limitation. I would call that a success.

Anyway I would not be too religious about this. I think practically it's
not really relevant as implementers should keep an eye on rate anyway
even without explicit rate negotiation. 

Christian


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Subject: Re: [Sip] PUBLISH message rates
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> hisham.khartabil@nokia.com wrote:
> 
>> The only way to do this right is to do what RTCP does with report
>> interval, and that is to adjust the rate of PUBLISH using some
>> formula that takes into account the number of publishers at a given
>> time. The more publishers that join, the less the rate at which
>> PUBLISH is sent at. The compositors need to communicate that
>> calculated rate to the publishers somehow. Do we want to make it that
>> complicated? If not, then I suggest adding text the says if the
>> compositor is overloaded, it can send a 5xx response with
>> retry-after.

Of course use of 5xx with retry-after wastes the effort the compositor 
has already expended in processing the publish up to that point.

It might be interesting to make a change permitting use of the 
Retry-After header with a 2xx response. Then the compositor could accept 
  a publish and at the same time ensure that the next one doesn't come 
too soon. This wouldn't be useful in all cases, but maybe in some.

	Paul


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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: <sip@ietf.org>
Cc: <rohan@cisco.com>,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: [Sip] Chair Rambling: Thoughts on Conflicting Pressures in SIP Work Planning
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Over the last week, I've had several fascinating conversations with =
people
about the way that SIP work gets planned and done. This has led me to a
realization of an underlying conflict that needs to be carefully =
managed.

Observation 1: Over time, the set of actively contributing people has
declined. New people find it hard to penetrate the "Inner Circle", and =
take
their energy, enthusiasm, and talent elsewhere -- or even worse, =
nowhere.

Observation 2: We have a lot of diffferent topics under consideration --
more than the set of actively contributing people can execute, or even
assimilate. It was noted several times in the SIP meeting that "This is =
a
great idea, and very interesting, but we just don't have the bandwidth =
to do
it."

The traditional way to deal with #1 is to open up to new ideas.

The traditional way to deal with #2 is to focus narrowly on priorities
(usually meaning "stuff we're already doing") and refuse to consider any =
new
topics. Another approach is to invest effort in getting more people =
actively
involved and contributing, which can be difficult "late in the game".

What we've been trying to do is steer a happy medium between the two --
giving consideration to new ideas that MIGHT become priority items,
encouraging new participation, and managing to priorities. SIPPING, for
example, is a mechanism for helping SIP evaluate and set incoming
priorities, while also entertaining newer ideas. It has certainly =
helped.
But there is always room for improvement.

As you relax and bask in that happy post-meeting glow (which I hope =
lasts no
more than about 72 hours, we have WORK to do!), I invite you all to =
consider
this matter, and share any insights either with the list as a whole or =
one
or all of the chairs as you feel is most appropriate.

Thanks, and I wish all of you who were at IETF 58 a safe trip home.

--
Dean


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Christian,

Thanks for your quick turn around.

-rohan

On Thursday, November 13, 2003, at 07:04 PM, Christian Stredicke wrote:

> Hi Rohan,
>
> looks like we have just been bombed back into the requirements phase of
> the dialog (or whatever the name is) package.
>
> I see the following use-cases:
>
> 1. STATUS information
>
> We need to see if a dialog is "ringing", "talking", "on hold", or has
> just been "terminated". Also it would be nice to have the to- and from,
> maybe referred-by and the start of the "talking" (so that we can
> calculate the duration). This information is nice for LED and
> SWITCHBOARD applications.
>
>
> 2. Call pickup, join, and other ACTIONS
>
> The dialog state subscriber needs to know what URL to dial in order to
> pick up a call or on how to join a call. Other actions might also be
> indicated.
>
>
> 3. The question of HANDSET-LIFTING
>
> We need to know if handsets have been lifted up (that's a simple
> requirement from the switchboard people which laugh at us if we say
> "impossible"; it is also needed for AUTOMATIC CALL BACK). We could 
> model
> this as dialog where "to" is not set yet.
>
>
> That's all on my wish list! No need for CSeq or other stuff end-users
> are not interested in!
>
> Please don't forget SIP did not replace the good old PBX yet and there
> is a reason for this.
>
>
> CS
>


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Date: Sat, 15 Nov 2003 16:28:08 -0800
Subject: Re: [Sip] GRUU, Contacts, and privacy
From: Cullen Jennings <fluffy@cisco.com>
To: <sip@ietf.org>
CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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I'm convinced :-) I agree we should not bother to deal with the privacy
stuff in GRUU. 

Cullen

On 11/12/03 11:39 PM, "Jonathan Rosenberg" <jdrosen@dynamicsoft.com> wrote:

> I agree, and I believe this was the consensus at the meeting as well.
> Interestingly, the current requirements document:
> http://www.ietf.org/internet-drafts/draft-rosenberg-sipping-gruu-reqs-01.txt
> 
> doesnt actually have any privacy requirements at all. So, there is
> nothing to remove :)
> 
> -Jonathan R.
> 
> Adam Roach wrote:
> 
>> Alan Johnston [mailto:alan.johnston@mci.com] writes:
>> 
>>> I agree 100%.  A private Contact URI is just one small piece
>>> of the privacy puzzle.  I think this privacy requirement should
>>> be removed.
>> 
>> I further agree.
>> 
>> /a
>> 


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Subject: Re: [Sip] Dialog-State Use Cases
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On Thursday, November 13, 2003, at 07:25 PM, Paul Kyzivat wrote:

> I think a lot of confusion could be eliminated by renaming the 
> document and the event package it defines. While it has something to 
> do with dialogs, it isn't fundamentally about dialogs. For instance, 
> it has *nothing* to do with dialogs containing only subscriptions.
>
> It is largely but not entirely related to dialogs that contain an 
> invitation. But it is concerned with some things that can't be learned 
> from just that.
>
> I suggest something like "call-info".

That was the original title, but for some reason we changed that.

Event package for SIP session and session-participation information ?

thanks,
-rohan


> More inline...
>
> 	Paul
>
> Christian Stredicke wrote:
>> 3. The question of HANDSET-LIFTING
>> We need to know if handsets have been lifted up (that's a simple
>> requirement from the switchboard people which laugh at us if we say
>> "impossible"; it is also needed for AUTOMATIC CALL BACK). We could 
>> model
>> this as dialog where "to" is not set yet.
>
> It really need have nothing to do with lifting a handset. Lots of 
> devices don't even have a handset - at least not one you lift.
>
> I think it is about being in a state of preparing to make a call. But 
> it is more than that, because not all devices would want to report 
> this when preparing to make a call. It also has something to do with 
> being unwilling or unable to receive a call while in this state, 
> and/or wanting to seize a shared line resource.


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To: "Dean Willis" <dwillis@dynamicsoft.com>
Cc: sip@ietf.org, sip-admin@ietf.org
Subject: Re: [Sip] Chair Rambling: Thoughts on Conflicting Pressures in SIP Work
 Planning
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Hi Dean, all, 

Nice summary of conflicting pressures. 

However, another serious pressure which has severely affected most all of 
us, especially over the past 2 - 3 years, is overlooked -- economics. 
These have been nasty economic times, to say the least.   And IETF is 
basically a volunteer organization.   It is certainly difficult to 
contribute to the greater good when the short term bottom line threatens 
basic personal security.  Therefore another contributing factor to the 
difficulty to progress the movement / mayhem we call SIP has been good ol' 
fashioned economic reality. 

With these points in mind I think it is paramount to make sure any new 
direction is forced to show a connection to the real world.   What is the 
value to the end user?  What is the value to those that would deploy it? 
What is the business case that will make this sustainable?  Therefore, 
what is in it for me, the would-be contributor of my precious personal 
time? 

Don't get me wrong!!!  While I have not been able to spend the time 
recently, I still want to do so.  And I still believe strongly in the 
cause.  However, any sort of process that would allow proof of the value 
to me and my employer would be a huge help in being able to commit the 
time required to really do the job.  While I actually think SIP/SIPPING 
does a pretty good job of showing value, and of driving related 
requirements, an explicit and visible evaluation process does seem to be 
lacking. 

Of course, I have NO IDEA how to put such a process in place.  I just 
think it would be highly valuable, both to gaining willing / committed 
contributors to the process, and to focusing efforts on overall success. 

My $0.01, 
Peter Blatherwick 







"Dean Willis" <dwillis@dynamicsoft.com>
Sent by: sip-admin@ietf.org
14.11.03 11:55

 
        To:     <sip@ietf.org>
        cc:     <rohan@cisco.com>, "'Gonzalo Camarillo'" 
<Gonzalo.Camarillo@lmf.ericsson.se>
        Subject:        [Sip] Chair Rambling: Thoughts on Conflicting Pressures in SIP Work 
Planning



Over the last week, I've had several fascinating conversations with people
about the way that SIP work gets planned and done. This has led me to a
realization of an underlying conflict that needs to be carefully managed.

Observation 1: Over time, the set of actively contributing people has
declined. New people find it hard to penetrate the "Inner Circle", and 
take
their energy, enthusiasm, and talent elsewhere -- or even worse, nowhere.

Observation 2: We have a lot of diffferent topics under consideration --
more than the set of actively contributing people can execute, or even
assimilate. It was noted several times in the SIP meeting that "This is a
great idea, and very interesting, but we just don't have the bandwidth to 
do
it."

The traditional way to deal with #1 is to open up to new ideas.

The traditional way to deal with #2 is to focus narrowly on priorities
(usually meaning "stuff we're already doing") and refuse to consider any 
new
topics. Another approach is to invest effort in getting more people 
actively
involved and contributing, which can be difficult "late in the game".

What we've been trying to do is steer a happy medium between the two --
giving consideration to new ideas that MIGHT become priority items,
encouraging new participation, and managing to priorities. SIPPING, for
example, is a mechanism for helping SIP evaluate and set incoming
priorities, while also entertaining newer ideas. It has certainly helped.
But there is always room for improvement.

As you relax and bask in that happy post-meeting glow (which I hope lasts 
no
more than about 72 hours, we have WORK to do!), I invite you all to 
consider
this matter, and share any insights either with the list as a whole or one
or all of the chairs as you feel is most appropriate.

Thanks, and I wish all of you who were at IETF 58 a safe trip home.

--
Dean


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<br><font size=2 face="sans-serif">Hi Dean, all, </font>
<br>
<br><font size=2 face="sans-serif">Nice summary of conflicting pressures. &nbsp;</font>
<br>
<br><font size=2 face="sans-serif">However, another serious pressure which has severely affected most all of us, especially over the past 2 - 3 years, is overlooked -- economics. &nbsp;These have been nasty economic times, to say the least. &nbsp; And IETF is basically a volunteer organization. &nbsp; It is certainly difficult to contribute to the greater good when the short term bottom line threatens basic personal security. &nbsp;Therefore another contributing factor to the difficulty to progress the movement / mayhem we call SIP has been good ol' fashioned economic reality. &nbsp;</font>
<br>
<br><font size=2 face="sans-serif">With these points in mind I think it is paramount to make sure any new direction is forced to show a connection to the real world. &nbsp; What is the value to the end user? &nbsp;What is the value to those that would deploy it? &nbsp;What is the business case that will make this sustainable? &nbsp;Therefore, what is in it for me, the would-be contributor of my precious personal time? &nbsp;</font>
<br>
<br><font size=2 face="sans-serif">Don't get me wrong!!! &nbsp;While I have not been able to spend the time recently, I still want to do so. &nbsp;And I still believe strongly in the cause. &nbsp;However, any sort of process that would allow proof of the value to me and my employer would be a huge help in being able to commit the time required to really do the job. &nbsp;While I actually think SIP/SIPPING does a pretty good job of showing value, and of driving related requirements, an explicit and visible evaluation process does seem to be lacking. &nbsp;</font>
<br>
<br><font size=2 face="sans-serif">Of course, I have NO IDEA how to put such a process in place. &nbsp;I just think it would be highly valuable, both to gaining willing / committed contributors to the process, and to focusing efforts on overall success. &nbsp;</font>
<br>
<br><font size=2 face="sans-serif">My $0.01, </font>
<br><font size=2 face="sans-serif">Peter Blatherwick </font>
<br>
<br>
<br>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td><font size=1 face="sans-serif"><b>&quot;Dean Willis&quot; &lt;dwillis@dynamicsoft.com&gt;</b></font>
<br><font size=1 face="sans-serif">Sent by: sip-admin@ietf.org</font>
<p><font size=1 face="sans-serif">14.11.03 11:55</font>
<br>
<td><font size=1 face="Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;&lt;sip@ietf.org&gt;</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;&lt;rohan@cisco.com&gt;, &quot;'Gonzalo Camarillo'&quot; &lt;Gonzalo.Camarillo@lmf.ericsson.se&gt;</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;[Sip] Chair Rambling: Thoughts on Conflicting Pressures in SIP Work Planning</font></table>
<br>
<br>
<br><font size=2 face="Courier New"><br>
Over the last week, I've had several fascinating conversations with people<br>
about the way that SIP work gets planned and done. This has led me to a<br>
realization of an underlying conflict that needs to be carefully managed.<br>
<br>
Observation 1: Over time, the set of actively contributing people has<br>
declined. New people find it hard to penetrate the &quot;Inner Circle&quot;, and take<br>
their energy, enthusiasm, and talent elsewhere -- or even worse, nowhere.<br>
<br>
Observation 2: We have a lot of diffferent topics under consideration --<br>
more than the set of actively contributing people can execute, or even<br>
assimilate. It was noted several times in the SIP meeting that &quot;This is a<br>
great idea, and very interesting, but we just don't have the bandwidth to do<br>
it.&quot;<br>
<br>
The traditional way to deal with #1 is to open up to new ideas.<br>
<br>
The traditional way to deal with #2 is to focus narrowly on priorities<br>
(usually meaning &quot;stuff we're already doing&quot;) and refuse to consider any new<br>
topics. Another approach is to invest effort in getting more people actively<br>
involved and contributing, which can be difficult &quot;late in the game&quot;.<br>
<br>
What we've been trying to do is steer a happy medium between the two --<br>
giving consideration to new ideas that MIGHT become priority items,<br>
encouraging new participation, and managing to priorities. SIPPING, for<br>
example, is a mechanism for helping SIP evaluate and set incoming<br>
priorities, while also entertaining newer ideas. It has certainly helped.<br>
But there is always room for improvement.<br>
<br>
As you relax and bask in that happy post-meeting glow (which I hope lasts no<br>
more than about 72 hours, we have WORK to do!), I invite you all to consider<br>
this matter, and share any insights either with the list as a whole or one<br>
or all of the chairs as you feel is most appropriate.<br>
<br>
Thanks, and I wish all of you who were at IETF 58 a safe trip home.<br>
<br>
--<br>
Dean<br>
<br>
<br>
_______________________________________________<br>
Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
This list is for NEW development of the core SIP Protocol<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
Use sipping@ietf.org for new developments on the application of sip<br>
</font>
<br>
<br>
--=_alternative 001423D085256DE1_=--

_______________________________________________
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From exim@www1.ietf.org  Mon Nov 17 08:33:33 2003
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From: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] Connection: Keep-Alive
Date: Mon, 17 Nov 2003 13:31:19 -0000
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All,

Can I just flag up that in one respect, UDP can have advantages over TCP =
(or
any other reliable transport).  Once the path from sender to receiver
machines becomes congested, TCP builds up a queue of pending SIP requests.
Note that this "queue" consists of messages queued both at the receiver (=
if
receiver is unable to handle received requests at a rate equal or in exce=
ss
of arrival rates) and the sender (if the network itself or the connection=
 to
the receiver is congested).  Quite where messages are queued can be
immaterial.

Ultimately, the time taken for a request to move from the bottom of this
queue to the top can exceed the 32s (64 * T1) request timeout imposed by =
RFC
3261 and policed by the sender.  Once this occurs, _all_ SIP requests sen=
t
to the receiver machine then fail until congestion subsides and the "bott=
om
to top" time falls below 32s (64 x T1).

By comparison, a UDP stack drops frames when congested so many SIP reques=
ts
will be dropped and fail but some will be get lucky and get processed by =
the
receiver.  As a result, under severe congestion, UDP can have _better_
request success rates that TCP.

As is currently being discussed, congestion can have many causes and solv=
ing
it is very tricky, if not impossible across the entire network.  I just
wanted to flag that using a reliable transport can introduce some problem=
s
as well as solving others.

Finally, a question for the group to consider.  If we are using a reliabl=
e
transport, does it even make sense to enforce the 32s (64 x T1)  timeout?
It seems that if we have a reliable transport, we should accept that the
request _will_ reach the next hop and that a response _will_ come back
(unless the connection fails - which we should be able to detect) and
therefore the 32s timeout might be unneccessary.

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


-----Original Message-----
From: Chris Boulton [mailto:cboulton@ubiquity.net]
Sent: 13 November 2003 12:52
To: SIP (E-mail)
Subject: RE: [Sip] Connection: Keep-Alive


I also agree that a document is important.  The consensus in the room
yesterday was am emphasis on TCP usage due to the known limitations of UD=
P.
Some form of BCP might go a long way to promoting it's currently limited
usage.
=20
Chris.
=20

	-----Original Message-----=20
	From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]=20
	Sent: Thu 13/11/2003 07:42=20
	To: seancolson@yahoo.com=20
	Cc: 'Vijay K. Gurbani'; 'Christian Stredicke'; sip@ietf.org;
rajnishjain@xl.com=20
	Subject: Re: [Sip] Connection: Keep-Alive
=09
=09

	Sorry for not being clear - I wasnt saying we DONT want a document.
At
	the meeting today, I advocated (and I think many others agreed) for
	creation of a document. In particular, I think it makes sense to
have
	a proposed standard document that describes all of the issues with
	getting tcp to work. This would presumably be an expansion of the
	existing connect-reuse draft. It needs to be PS since there is
	normative behavior associated with many parts of it.
=09
	-Jonathan R.
=09
	Sean Olson wrote:
=09
	> Actually, many implementations are not doing this unfortunately. A
BCP is
	> not a bad idea.
	>
	> -----Original Message-----
	> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
Jonathan
	> Rosenberg
	> Sent: Wednesday, November 12, 2003 11:12 PM
	> To: Vijay K. Gurbani
	> Cc: Christian Stredicke; sip@ietf.org; rajnishjain@xl.com
	> Subject: Re: [Sip] Connection: Keep-Alive
	>
	>
	>
	> Vijay K. Gurbani wrote:
	>
	>
	>
	>>>In any case the behavior of the UAC as well as the proxy needs to
be
	>>>(re)defined, I think the current TCP specification does not allow
a
	>>>permanent connection.
	>>>
	>>>Any thoughts on this?
	>>
	>>
	>>At the Atlanta IETF, it was decided that we will open a bug
against
	>>rfc3261 to provide more information on TCP connection handling.
	>
	>
	> Well, in particular, it was concluded that signaling "please keep
this
	> connection open" was not needed at all. Rather, the server should
keep the
	> connection open as long as it could, and only close it if it ran
out of
	> resources. Since there was no benefit to closing it prematurely
(extra costs
	> of setting it up again), and since signaling the desire for
persistence
	> could not be used to force the server to keep it open if the
server ran out
	> of resources, the signaling was not needed.
	> Indeed, the observation was that many implementations in any case
were doing
	> this; it was just unclear in the spec.
	>
	> -Jonathan R.
	>
=09
	--
	Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
	Chief Technology Officer                    Parsippany, NJ
07054-2711
	dynamicsoft
	jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
	http://www.jdrosen.net                      PHONE: (973) 952-5000
	http://www.dynamicsoft.com
=09
=09
	_______________________________________________
	Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
	This list is for NEW development of the core SIP Protocol
	Use sip-implementors@cs.columbia.edu for questions on current sip
	Use sipping@ietf.org for new developments on the application of sip
=09

J*fj)b	bm=EB=81=B6?=0C0=EC=87=9B'~ffX)=EE=96=8A"8bX=E9=AC=B6+=1FDY=D7=AFz=
Z)	ay=ED=AF=81+y"=0F>-%R=E0=A7=9BWz{h,r=E4=89=89ny=DB=9Fzb=E6=A4=A2{(=CB=AB
*T"=E7=A6=98'~~=E6=8A=8A{=07^hgg'=EB=81=B6=17bqbz=1F

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Use sip-implementors@cs.columbia.edu for questions on current sip
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From: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
To: "'sip@ietf.org'" <sip@ietf.org>
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Subject: [Sip] 3-way handshake bad!
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All,

For context, see IETF 58 minutes.  There was a suggestion that the
non-INVITE avalanche problem be solved by using a 3-way handshake.  I just
want to flag that this idea can have unfortunate consequences during
congestion.

Firstly, consider a server under load.  The 3-way handshake requires the
following.

server processes request - #1
server sends response - #2
server processes request/handshake - #3.

By adding the 3rd step, we have _doubled_ the number of requests to be
handled by the server increasing the load and increasing the chances it
becomes congested.

As per my earlier e-mail, for TCP, the chances of the initial request
completing can be either 1.0 or 0.0 depending on the congestion.  Increasing
the congestion at the server, increases the chance of requests failing.
For UDP, there is a 1/x (x >= 1) chance of the request being received and
processed by the server.  Similarly, there is a 1/x chance of the 3rd
request/handshake reaching the server.  This means the chances of the 3-way
handshake completing are
1/(x^2).  So adding the third step greatly affects the chances of the 3-way
handshake completing.

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


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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: Mon, 17 Nov 2003 10:18:36 -0500
From: "David R. Oran" <oran@cisco.com>
To: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>,
        "'sip@ietf.org'" <sip@ietf.org>
Subject: Re: [Sip] 3-way handshake bad!
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--On Monday, November 17, 2003 3:01 PM +0000 "Paul D.Smith"=20
<Paul.D.Smith@dataconnection.com> wrote:

> All,
>
> For context, see IETF 58 minutes.  There was a suggestion that the
> non-INVITE avalanche problem be solved by using a 3-way handshake.  I =
just
> want to flag that this idea can have unfortunate consequences during
> congestion.
>
I made that suggestion. I disagree that it has unfortunate congestion=20
consequences. See below. Note however, that it may be unproductive to start =

an extended dialog on this subject since such a change to SIP may be too=20
radical at this stage in the protocol's evolution. We need to weigh this=20
against the efficacy-versus-radicality tradeoffs of other possible ways to=20
ameliorate the problems.

> Firstly, consider a server under load.  The 3-way handshake requires the
> following.
>
> server processes request - #1
> server sends response - #2
> server processes request/handshake - #3.
>
> By adding the 3rd step, we have _doubled_ the number of requests to be
> handled by the server increasing the load and increasing the chances it
> becomes congested.
>
Not really. The processing of an ACK transaction is not nearly as expensive =

as processing of the actual request. All it requires is transaction=20
matching and state cleanup (if any). This had better be blindingly fast in=20
any reasonable SIP server or you have a lot of worse things to worry about, =

including DoS attacks from legitimate authenticated/authorized sources.

(Aside: I actually have a patent disclosure filed on how to do DoS=20
prevention by SIP servers where the relative costs of transaction matching, =

authentication, authorization, and state-creation are taken into account).

> As per my earlier e-mail, for TCP, the chances of the initial request
> completing can be either 1.0 or 0.0 depending on the congestion.
For a single signaling hop this is more-or-less true, but assumes a number=20
of things, including whether the server is programmed to back-pressure the=20
TCP connection (causing HOL blocking) or is maintaining internal request=20
queues from which requests might get dropped on overflow. For multi-hop=20
signaling, TCP is of little help especially when multiplexing is happening.

> Increasing the congestion at the server, increases the chance of requests
> failing. For UDP, there is a 1/x (x >=3D 1) chance of the request being
> received and processed by the server.  Similarly, there is a 1/x chance
> of the 3rd request/handshake reaching the server.
This is true only if all requests have equal costs, and if ACKs have equal=20
costs to requests. This is demonstrably not the case.

> This means the chances
> of the 3-way handshake completing are
> 1/(x^2).  So adding the third step greatly affects the chances of the
> 3-way handshake completing.
>
I believe this logic is flawed.

Cheers, Dave.

> Paul D.Smith
> Network Protocols Group
> Data Connection Ltd (DCL)
> Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com
> Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
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Date: Mon, 17 Nov 2003 10:26:02 -0500
From: "David R. Oran" <oran@cisco.com>
To: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>,
        "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] Connection: Keep-Alive
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Unlike Paul's earlier message on 3-way handshaking, with which I disagreed, =

this post makes a very important point with which I whole-heartedly agree.

When the signaling is single-hop, it may in fact make sense to disable the=20
request timers. Unfortunately, when the signaling is multi-hop this doesn't =

work and it isn't clear what to do.

One of the objections to 3-way handshakes (aside from the one Paul made in=20
his earlier post) is the additional state in proxies. If we in fact do=20
enough surgery to SIP to introduce 3-way handshakes on non-invite=20
transactions, perhaps we can also at the same time introduce a header hint=20
on non-invite transactions which indicate to proxies that it is safe to=20
operate statelessly on such transactions if the proxy so chooses.

Dave.

--On Monday, November 17, 2003 1:31 PM +0000 "Paul D.Smith"=20
<Paul.D.Smith@dataconnection.com> wrote:

> All,
>
> Can I just flag up that in one respect, UDP can have advantages over TCP
> (or any other reliable transport).  Once the path from sender to receiver
> machines becomes congested, TCP builds up a queue of pending SIP =
requests.
> Note that this "queue" consists of messages queued both at the receiver
> (if receiver is unable to handle received requests at a rate equal or in
> excess of arrival rates) and the sender (if the network itself or the
> connection to the receiver is congested).  Quite where messages are
> queued can be immaterial.
>
> Ultimately, the time taken for a request to move from the bottom of this
> queue to the top can exceed the 32s (64 * T1) request timeout imposed by
> RFC 3261 and policed by the sender.  Once this occurs, _all_ SIP requests
> sent to the receiver machine then fail until congestion subsides and the
> "bottom to top" time falls below 32s (64 x T1).
>
> By comparison, a UDP stack drops frames when congested so many SIP
> requests will be dropped and fail but some will be get lucky and get
> processed by the receiver.  As a result, under severe congestion, UDP can
> have _better_ request success rates that TCP.
>
> As is currently being discussed, congestion can have many causes and
> solving it is very tricky, if not impossible across the entire network.
> I just wanted to flag that using a reliable transport can introduce some
> problems as well as solving others.
>
> Finally, a question for the group to consider.  If we are using a =
reliable
> transport, does it even make sense to enforce the 32s (64 x T1)  timeout?
> It seems that if we have a reliable transport, we should accept that the
> request _will_ reach the next hop and that a response _will_ come back
> (unless the connection fails - which we should be able to detect) and
> therefore the 32s timeout might be unneccessary.
>
> Paul D.Smith
> Network Protocols Group
> Data Connection Ltd (DCL)
> Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com
> Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
>
>
> -----Original Message-----
> From: Chris Boulton [mailto:cboulton@ubiquity.net]
> Sent: 13 November 2003 12:52
> To: SIP (E-mail)
> Subject: RE: [Sip] Connection: Keep-Alive
>
>
> I also agree that a document is important.  The consensus in the room
> yesterday was am emphasis on TCP usage due to the known limitations of
> UDP. Some form of BCP might go a long way to promoting it's currently
> limited usage.
>
> Chris.
>
>
> 	-----Original Message-----
> 	From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> 	Sent: Thu 13/11/2003 07:42
> 	To: seancolson@yahoo.com
> 	Cc: 'Vijay K. Gurbani'; 'Christian Stredicke'; sip@ietf.org;
> rajnishjain@xl.com
> 	Subject: Re: [Sip] Connection: Keep-Alive
> 	
> 	
>
> 	Sorry for not being clear - I wasn't saying we DONT want a document.
> At
> 	the meeting today, I advocated (and I think many others agreed) for
> 	creation of a document. In particular, I think it makes sense to
> have
> 	a proposed standard document that describes all of the issues with
> 	getting tcp to work. This would presumably be an expansion of the
> 	existing connect-reuse draft. It needs to be PS since there is
> 	normative behavior associated with many parts of it.
> 	
> 	-Jonathan R.
> 	
> 	Sean Olson wrote:
> 	
> 	> Actually, many implementations are not doing this unfortunately. A
> BCP is
> 	> not a bad idea.
> 	>
> 	> -----Original Message-----
> 	> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
> Jonathan
> 	> Rosenberg
> 	> Sent: Wednesday, November 12, 2003 11:12 PM
> 	> To: Vijay K. Gurbani
> 	> Cc: Christian Stredicke; sip@ietf.org; rajnishjain@xl.com
> 	> Subject: Re: [Sip] Connection: Keep-Alive
> 	>
> 	>
> 	>
> 	> Vijay K. Gurbani wrote:
> 	>
> 	>
> 	>
> 	>>>In any case the behavior of the UAC as well as the proxy needs to
> be
> 	>>>(re)defined, I think the current TCP specification does not allow
> a
> 	>>>permanent connection.
> 	>>>
> 	>>>Any thoughts on this?
> 	>>
> 	>>
> 	>>At the Atlanta IETF, it was decided that we will open a bug
> against
> 	>>rfc3261 to provide more information on TCP connection handling.
> 	>
> 	>
> 	> Well, in particular, it was concluded that signaling "please keep
> this
> 	> connection open" was not needed at all. Rather, the server should
> keep the
> 	> connection open as long as it could, and only close it if it ran
> out of
> 	> resources. Since there was no benefit to closing it prematurely
> (extra costs
> 	> of setting it up again), and since signaling the desire for
> persistence
> 	> could not be used to force the server to keep it open if the
> server ran out
> 	> of resources, the signaling was not needed.
> 	> Indeed, the observation was that many implementations in any case
> were doing
> 	> this; it was just unclear in the spec.
> 	>
> 	> -Jonathan R.
> 	>
> 	
> 	--
> 	Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> 	Chief Technology Officer                    Parsippany, NJ
> 07054-2711
> 	dynamicsoft
> 	jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> 	http://www.jdrosen.net                      PHONE: (973) 952-5000
> 	http://www.dynamicsoft.com
> 	
> 	
> 	_______________________________________________
> 	Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> 	This list is for NEW development of the core SIP Protocol
> 	Use sip-implementors@cs.columbia.edu for questions on current sip
> 	Use sipping@ietf.org for new developments on the application of sip
> 	
>
> J*fj)b	bm??=0C0?'~ffX)?"8bX?+=1FDY?zZ)	ay?+y"=0F>-%R?Wz{h,r?ny?zb?{(?
> *T"?'~~?{=07^hgg'?=17bqbz=1F
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
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Has any progress been taken on the draft by Frank Miller on Carrying TCAP in SIP messages?  Are there other drafts that address SIP access to SS7 database information through TCAP or another method?
 
Thanks,
 
Kevin 

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<DIV><SPAN class=086330721-17112003><FONT face=Arial size=2>Has any progress 
been taken on the draft by Frank Miller on Carrying TCAP in SIP messages?&nbsp; 
Are there other drafts that address SIP access to SS7 database information 
through TCAP or another method?</FONT></SPAN></DIV>
<DIV><SPAN class=086330721-17112003><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=086330721-17112003><FONT face=Arial 
size=2>Thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=086330721-17112003><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV align=left><FONT face="Monotype Corsiva" size=4>Kevin 
</FONT></DIV></BODY></HTML>

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To: "'McCandless, Kevin'" <KMcCandless@verisign.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Sip support for TCAP queries
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=20

You mean other than us implementing it?  No, I don't think so.  I think =
its
considered Rosemary's Baby by most.

=20

FM

=20

=20

Frank W. Miller, Ph.D.

Chief Technical Officer

sentitO Networks, Inc.

fmiller@sentito.com

=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
McCandless, Kevin
Sent: Monday, November 17, 2003 4:11 PM
To: sip@ietf.org
Subject: [Sip] Sip support for TCAP queries

=20

Has any progress been taken on the draft by Frank Miller on Carrying =
TCAP in
SIP messages?  Are there other drafts that address SIP access to SS7
database information through TCAP or another method?

=20

Thanks,

=20

Kevin=20


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I don&#8217;t think so.&nbsp; I think its considered Rosemary&#8217;s =
Baby by most&#8230;</span></font></p>

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

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

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

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

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DDE
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Frank W. Miller, =
Ph.D.</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>sentitO Networks, =
Inc.</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
 =
10.0pt;font-family:Arial;color:navy'>fmiller@sentito.com</span></font></p=
>

</div>

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

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'>-----Original =
Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> sip-admin@ietf.org
[mailto:sip-admin@ietf.org] <b><span style=3D'font-weight:bold'>On =
Behalf Of </span></b>McCandless,
Kevin<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, November =
17, 2003
4:11 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] Sip =
support for
TCAP queries</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Has any progress been taken =
on the
draft by Frank Miller on Carrying TCAP in SIP messages?&nbsp; Are there =
other
drafts that address SIP access to SS7 database information through TCAP =
or
another method?</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Thanks,</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D4 =
face=3D"Monotype Corsiva"><span
style=3D'font-size:13.5pt;font-family:"Monotype Corsiva"'>Kevin =
</span></font></p>

</div>

</body>

</html>

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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From exim@www1.ietf.org  Tue Nov 18 06:34:27 2003
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	Tue, 18 Nov 2003 11:32:17 GMT
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X-Originating-Email: [rajarshi_chakraborty@hotmail.com]
From: "Rajarshi Chakraborty" <rajarshi_chakraborty@hotmail.com>
To: sip@ietf.org
Cc: agupta@cse.iitkgp.ernet.in
Date: Tue, 18 Nov 2003 11:32:17 +0000
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Subject: [Sip] URGENT: Refering to obsolete drafts
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Hi,

I want to put an Internet Draft's name in the reference of my research 
paper. This Internet Draft happens to be 
obsolete(draft-schulzrinne-sip-register-01); it expired on Oct'01 and no 
more versions of it were released. Could someone suggest me how to address 
this obsolete draft in the reference section?

I used the following:

Schulzrinne, H., "SIP Registration", 
<draft-schulzrinne-sip-register-01.txt>, work in progress, October 2001

Is the above correct? Is it proper to use "work in progress" for this 
obsolete draft? Any kind of immediate response and/or suggestion will be 
highly appreciated. I am new in this.

regards,
Rajarshi

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

_________________________________________________________________
Garfield on your mobile. Download now. http://server1.msn.co.in/sp03/gprs/ 
How cool can life get?


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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 support for TCAP queries
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Thread-Topic: [Sip] Sip support for TCAP queries
Thread-Index: AcOtYXkLHJlfBjGrSiGFCFE0fdXjAAAhUBqg
From: "Chong, Koan S, ALABS" <kschong@att.com>
To: "Frank W. Miller" <fmiller@sentito.com>,
        "McCandless, Kevin" <KMcCandless@verisign.com>
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------_=_NextPart_001_01C3ADF0.D33915A5
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=20
A couple specific questions:
=20
What is the recommendation on how a SIP network element can access the
callers' name databases (CNAM in US) in PSTN after the SIP element has
determined that the called parties have subscribed to Caller ID?
=20
How about interworking with CLASS Auto-Recall/Callback (CCBS in ITU?)
procedures?
=20
Assume TCAP over SIGTRAN is a bit long long term....
=20
Koan
Koan S.Chong
AT&T
1-732-420-4557
=20

-----Original Message-----
From: Frank W. Miller [mailto:fmiller@sentito.com]
Sent: Monday, November 17, 2003 6:18 PM
To: 'McCandless, Kevin'
Cc: sip@ietf.org
Subject: RE: [Sip] Sip support for TCAP queries



=20

You mean other than us implementing it?  No, I don't think so.  I think
its considered Rosemary's Baby by most...

=20

FM

=20

=20

Frank W. Miller, Ph.D.

Chief Technical Officer

sentitO Networks, Inc.

fmiller@sentito.com

=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
McCandless, Kevin
Sent: Monday, November 17, 2003 4:11 PM
To: sip@ietf.org
Subject: [Sip] Sip support for TCAP queries

=20

Has any progress been taken on the draft by Frank Miller on Carrying
TCAP in SIP messages?  Are there other drafts that address SIP access to
SS7 database information through TCAP or another method?

=20

Thanks,

=20

Kevin=20


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</HEAD>
<BODY lang=3DEN-US vLink=3Dpurple link=3Dblue>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D249561415-18112003><FONT face=3DArial color=3D#0000ff =
size=3D2>A=20
couple&nbsp;specific questions:</FONT></SPAN></DIV>
<DIV><SPAN class=3D249561415-18112003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D249561415-18112003>What=20
is the recommendation on how a SIP network element can access the =
callers' name=20
databases (CNAM in US) in PSTN after the SIP element has determined that =
the=20
called parties have subscribed to Caller ID?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D249561415-18112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D249561415-18112003>How=20
about interworking with CLASS Auto-Recall/Callback (CCBS in ITU?)=20
procedures?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D249561415-18112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D249561415-18112003>Assume=20
TCAP over SIGTRAN&nbsp;is a bit long long term....</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D249561415-18112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D249561415-18112003>Koan</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D249561415-18112003>Koan=20
S.Chong</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D249561415-18112003>AT&amp;T</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D249561415-18112003>1-732-420-4557</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Frank W. Miller=20
  [mailto:fmiller@sentito.com]<BR><B>Sent:</B> Monday, November 17, 2003 =
6:18=20
  PM<BR><B>To:</B> 'McCandless, Kevin'<BR><B>Cc:</B>=20
  sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Sip support for TCAP=20
  queries<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"></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">You mean =
other than=20
  us implementing it?&nbsp; No, I don&#8217;t think so.&nbsp; I think =
its considered=20
  Rosemary&#8217;s Baby by most&#8230;</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">FM</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 class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN =
lang=3DDE=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Frank W. =
Miller,=20
  Ph.D.</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">Chief =
Technical=20
  Officer</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">sentitO =
Networks,=20
  Inc.</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">fmiller@sentito.com</SPAN></FONT></P></DIV>
  <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>=20
  sip-admin@ietf.org [mailto:sip-admin@ietf.org] <B><SPAN=20
  style=3D"FONT-WEIGHT: bold">On Behalf Of </SPAN></B>McCandless,=20
  Kevin<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Monday, =
November=20
  17, 2003 4:11 PM<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">To:</SPAN></B>=20
  sip@ietf.org<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">Subject:</SPAN></B> [Sip]=20
  Sip support for TCAP queries</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">Has any progress been =
taken on the=20
  draft by Frank Miller on Carrying TCAP in SIP messages?&nbsp; Are =
there other=20
  drafts that address SIP access to SS7 database information through =
TCAP or=20
  another method?</SPAN></FONT></P></DIV>
  <DIV>
  <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">Thanks,</SPAN></FONT></P></DIV>
  <DIV>
  <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=3D"Monotype Corsiva"=20
  size=3D4><SPAN style=3D"FONT-SIZE: 13.5pt; FONT-FAMILY: 'Monotype =
Corsiva'">Kevin=20
  </SPAN></FONT></P></DIV></BLOCKQUOTE></BODY></HTML>

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From: "Frank W. Miller" <fmiller@sentito.com>
To: "'Chong, Koan S, ALABS'" <kschong@att.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Sip support for TCAP queries
Date: Tue, 18 Nov 2003 11:50:15 -0500
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=20

The main idea of this draft was to provide a capability for SIP =
endpoints to
do TCAP queries without having to implement the binary TCAP protocol.  =
The
motivation is that there are a lot of TCAP queries that take place (800,
LNP, CNAME) that are really simple and that major carriers may not want =
to
go and reimplement in an ENUM database in the short term.  These =
carriers
have spent a lot of money on their SCPs and converting the to ENUM is
non-trivial.  Now, the SCPs could be front-ended and conversion can be
automated but this draft provides an alternative.  So, to answer the =
first
question, the SIP endpoint formats the correct query in the XML format
specified and sends if using an INFO to a Signaling Gateway (SG).  The =
SG
then converts the query to binary TCAP and dips the SCP.  The XML format =
can
be templated in the implementation so the effort is quite minimal on the =
SIP
endpoint.  To answer the second question, the draft talks about a couple =
of
examples but the specification encompasses the entire TCAP message set =
so
you can format the appropriate dip for AR/C.  One caveat is that the =
draft
is Telcordia based.  It would need to be extended to take into account =
other
variants.  On the third question, <personal view> I'm a believer in
translation over tunneling, tunneling just puts off work and mixes =
messaging
from different signaling networks unnecessarily</personal view>.

=20

FM

=20

=20

=20

Frank W. Miller, Ph.D.

Chief Technical Officer

sentitO Networks, Inc.

fmiller@sentito.com

=20

-----Original Message-----
From: Chong, Koan S, ALABS [mailto:kschong@att.com]=20
Sent: Tuesday, November 18, 2003 11:27 AM
To: Frank W. Miller; McCandless, Kevin
Cc: sip@ietf.org
Subject: RE: [Sip] Sip support for TCAP queries

=20

=20

A couple specific questions:

=20

What is the recommendation on how a SIP network element can access the
callers' name databases (CNAM in US) in PSTN after the SIP element has
determined that the called parties have subscribed to Caller ID?

=20

How about interworking with CLASS Auto-Recall/Callback (CCBS in ITU?)
procedures?

=20

Assume TCAP over SIGTRAN is a bit long long term....

=20

Koan

Koan S.Chong

AT&T

1-732-420-4557

=20

-----Original Message-----
From: Frank W. Miller [mailto:fmiller@sentito.com]
Sent: Monday, November 17, 2003 6:18 PM
To: 'McCandless, Kevin'
Cc: sip@ietf.org
Subject: RE: [Sip] Sip support for TCAP queries

=20

You mean other than us implementing it?  No, I don't think so.  I think =
its
considered Rosemary's Baby by most.

=20

FM

=20

=20

Frank W. Miller, Ph.D.

Chief Technical Officer

sentitO Networks, Inc.

fmiller@sentito.com

=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
McCandless, Kevin
Sent: Monday, November 17, 2003 4:11 PM
To: sip@ietf.org
Subject: [Sip] Sip support for TCAP queries

=20

Has any progress been taken on the draft by Frank Miller on Carrying =
TCAP in
SIP messages?  Are there other drafts that address SIP access to SS7
database information through TCAP or another method?

=20

Thanks,

=20

Kevin=20


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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>The main idea of this draft was to =
provide
a capability for SIP endpoints to do TCAP queries without having to =
implement the
binary TCAP protocol. &nbsp;The motivation is that there are a lot of =
TCAP
queries that take place (800, LNP, CNAME) that are really simple and =
that major
carriers may not want to go and reimplement in an ENUM database in the =
short
term. &nbsp;These carriers have spent a lot of money on their SCPs and
converting the to ENUM is non-trivial.&nbsp; Now, the SCPs could be =
front-ended
and conversion can be automated but this draft provides an =
alternative.&nbsp;
So, to answer the first question, the SIP endpoint formats the correct =
query in
the XML format specified and sends if using an INFO to a Signaling =
Gateway
(SG). &nbsp;The SG then converts the query to binary TCAP and dips the
SCP.&nbsp; The XML format can be templated in the implementation so the =
effort
is quite minimal on the SIP endpoint.&nbsp; To answer the second =
question, the
draft talks about a couple of examples but the specification encompasses =
the
entire TCAP message set so you can format the appropriate dip for =
AR/C.&nbsp; One
caveat is that the draft is Telcordia based.&nbsp; It would need to be =
extended
to take into account other variants.&nbsp; On the third question, =
&lt;personal
view&gt; I&#8217;m a believer in translation over tunneling, tunneling =
just
puts off work and mixes messaging from different signaling networks =
unnecessarily&lt;/personal
view&gt;.</span></font></p>

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

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

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

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

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

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DDE
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Frank W. Miller, =
Ph.D.</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>sentitO Networks, =
Inc.</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
 =
10.0pt;font-family:Arial;color:navy'>fmiller@sentito.com</span></font></p=
>

</div>

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

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'>-----Original =
Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> Chong, Koan S, =
ALABS
[mailto:kschong@att.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, November =
18, 2003
11:27 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Frank W. Miller; =
McCandless,
Kevin<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [Sip] Sip =
support for
TCAP queries</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>A =
couple&nbsp;specific
questions:</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>What is the
recommendation on how a SIP network element can access the callers' name
databases (CNAM in US) in PSTN after the SIP element has determined that =
the called
parties have subscribed to Caller ID?</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>How about =
interworking
with CLASS Auto-Recall/Callback (CCBS in ITU?) =
procedures?</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>Assume TCAP over
SIGTRAN&nbsp;is a bit long long term....</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>Koan</span></font=
></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>Koan =
S.Chong</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>AT&amp;T</span></=
font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>1-732-420-4557</s=
pan></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<blockquote style=3D'border:none;border-left:solid blue =
1.5pt;padding:0in 0in 0in 3.0pt;
margin-left:3.0pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'>=


<p class=3DMsoNormal =
style=3D'margin-right:0in;margin-bottom:12.0pt;margin-left:
.5in'><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>-----Original
Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> Frank W. Miller
[mailto:fmiller@sentito.com]<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, November =
17, 2003
6:18 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> 'McCandless, =
Kevin'<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [Sip] Sip =
support for
TCAP queries</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dnavy face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>You mean other =
than us
implementing it?&nbsp; No, I don&#8217;t think so.&nbsp; I think its =
considered
Rosemary&#8217;s Baby by most&#8230;</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dnavy face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>FM</span></font><=
/p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dnavy face=3DArial><span
lang=3DDE style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Frank =
W. Miller,
Ph.D.</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dnavy face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Chief Technical =
Officer</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dnavy face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>sentitO =
Networks, Inc.</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dnavy face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>fmiller@sentito.c=
om</span></font></p>

</div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:1.0in'><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'>-----Original =
Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> sip-admin@ietf.org
[mailto:sip-admin@ietf.org] <b><span style=3D'font-weight:bold'>On =
Behalf Of </span></b>McCandless,
Kevin<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, November =
17, 2003
4:11 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] Sip =
support for
TCAP queries</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:1.0in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<div>

<p class=3DMsoNormal style=3D'margin-left:1.0in'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Has any progress been taken =
on the
draft by Frank Miller on Carrying TCAP in SIP messages?&nbsp; Are there =
other drafts
that address SIP access to SS7 database information through TCAP or =
another
method?</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:1.0in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:1.0in'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Thanks,</span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:1.0in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<p class=3DMsoNormal style=3D'margin-left:1.0in'><font size=3D4
face=3D"Monotype Corsiva"><span =
style=3D'font-size:13.5pt;font-family:"Monotype Corsiva"'>Kevin
</span></font></p>

</blockquote>

</div>

</body>

</html>

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From: "Tolga Asveren" <asveren@ulticom.com>
To: "Frank W. Miller" <fmiller@sentito.com>,
        "'Chong, Koan S, ALABS'" <kschong@att.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Sip support for TCAP queries
Date: Tue, 18 Nov 2003 12:05:45 -0500
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Message
  -----Original Message-----
  From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Frank W.
Miller
  Sent: Tuesday, November 18, 2003 11:50 AM
  To: 'Chong, Koan S, ALABS'
  Cc: sip@ietf.org
  Subject: RE: [Sip] Sip support for TCAP queries




  The main idea of this draft was to provide a capability for SIP endpoints
to do TCAP queries without having to implement the binary TCAP protocol.
The motivation is that there are a lot of TCAP queries that take place (800,
LNP, CNAME) that are really simple and that major carriers may not want to
go and reimplement in an ENUM database in the short term.  These carriers
have spent a lot of money on their SCPs and converting the to ENUM is
non-trivial.  Now, the SCPs could be front-ended and conversion can be
automated but this draft provides an alternative.  So, to answer the first
question, the SIP endpoint formats the correct query in the XML format
specified and sends if using an INFO to a Signaling Gateway (SG).  The SG
then converts the query to binary TCAP and dips the SCP.  The XML format can
be templated in the implementation so the effort is quite minimal on the SIP
endpoint.  To answer the second question, the draft talks about a couple of
examples but the specification encompasses the entire TCAP message set so
you can format the appropriate dip for AR/C.  One caveat is that the draft
is Telcordia based.  It would need to be extended to take into account other
variants.  On the third question, <personal view> I'm a believer in
translation over tunneling, tunneling just puts off work and mixes messaging
from different signaling networks unnecessarily</personal view>.

  [TOLGA]-Although I believe this comment would fit more to SIGTRAN WG-
SIGTRAN is not tunneling, it just uses SCTP as transport, like SIP uses
TCAP/UDP/SCTP. If you consider SS7 messages tunneled with SIGTRAN, the same
should apply also for SIP messages. So, I don't see any disadvantage of
using it to do a TCAP query to a database. And I would think,  SIP is also
not the best way to do it, considering that it is a protocol about
"sessions". If one wants to reuse existing equipment -which talks TCAP-,
traditional SS7 or SIGTRAN are IMO the best two choices.



  FM







  Frank W. Miller, Ph.D.

  Chief Technical Officer

  sentitO Networks, Inc.

  fmiller@sentito.com



  -----Original Message-----
  From: Chong, Koan S, ALABS [mailto:kschong@att.com]
  Sent: Tuesday, November 18, 2003 11:27 AM
  To: Frank W. Miller; McCandless, Kevin
  Cc: sip@ietf.org
  Subject: RE: [Sip] Sip support for TCAP queries





  A couple specific questions:



  What is the recommendation on how a SIP network element can access the
callers' name databases (CNAM in US) in PSTN after the SIP element has
determined that the called parties have subscribed to Caller ID?



  How about interworking with CLASS Auto-Recall/Callback (CCBS in ITU?)
procedures?



  Assume TCAP over SIGTRAN is a bit long long term....



  Koan

  Koan S.Chong

  AT&T

  1-732-420-4557



    -----Original Message-----
    From: Frank W. Miller [mailto:fmiller@sentito.com]
    Sent: Monday, November 17, 2003 6:18 PM
    To: 'McCandless, Kevin'
    Cc: sip@ietf.org
    Subject: RE: [Sip] Sip support for TCAP queries



    You mean other than us implementing it?  No, I don't think so.  I think
its considered Rosemary's Baby by most.



    FM





    Frank W. Miller, Ph.D.

    Chief Technical Officer

    sentitO Networks, Inc.

    fmiller@sentito.com



    -----Original Message-----
    From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
McCandless, Kevin
    Sent: Monday, November 17, 2003 4:11 PM
    To: sip@ietf.org
    Subject: [Sip] Sip support for TCAP queries



    Has any progress been taken on the draft by Frank Miller on Carrying
TCAP in SIP messages?  Are there other drafts that address SIP access to SS7
database information through TCAP or another method?



    Thanks,



    Kevin

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<BODY lang=3DEN-US vLink=3Dpurple link=3Dblue>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> sip-admin@ietf.org =

  [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Frank W. =
Miller<BR><B>Sent:</B>=20
  Tuesday, November 18, 2003 11:50 AM<BR><B>To:</B> 'Chong, Koan S,=20
  ALABS'<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> RE: [Sip] Sip =
support for=20
  TCAP queries<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"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">The main =
idea of this=20
  draft was to provide a capability for SIP endpoints to do TCAP queries =
without=20
  having to implement the binary TCAP protocol. &nbsp;The motivation is =
that=20
  there are a lot of TCAP queries that take place (800, LNP, CNAME) that =
are=20
  really simple and that major carriers may not want to go and =
reimplement in an=20
  ENUM database in the short term. &nbsp;These carriers have spent a lot =
of=20
  money on their SCPs and converting the to ENUM is non-trivial.&nbsp; =
Now, the=20
  SCPs could be front-ended and conversion can be automated but this =
draft=20
  provides an alternative.&nbsp; So, to answer the first question, the =
SIP=20
  endpoint formats the correct query in the XML format specified and =
sends if=20
  using an INFO to a Signaling Gateway (SG). &nbsp;The SG then converts =
the=20
  query to binary TCAP and dips the SCP.&nbsp; The XML format can be =
templated=20
  in the implementation so the effort is quite minimal on the SIP=20
  endpoint.&nbsp; To answer the second question, the draft talks about a =
couple=20
  of examples but the specification encompasses the entire TCAP message =
set so=20
  you can format the appropriate dip for AR/C.&nbsp; One caveat is that =
the=20
  draft is Telcordia based.&nbsp; It would need to be extended to take =
into=20
  account other variants.&nbsp; On the third question, &lt;personal =
view&gt; I&#8217;m=20
  a believer in translation over tunneling, tunneling just puts off work =
and=20
  mixes messaging from different signaling networks =
unnecessarily&lt;/personal=20
  view&gt;.<SPAN class=3D245425916-18112003><FONT=20
  color=3D#0000ff>&nbsp;</FONT></SPAN></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><SPAN=20
  class=3D245425916-18112003><FONT color=3D#0000ff>[TOLGA]-Although I =
believe=20
  this&nbsp;comment would fit more to SIGTRAN WG- SIGTRAN is not =
tunneling, it=20
  just uses SCTP&nbsp;as transport, like SIP uses TCAP/UDP/SCTP. If you =
consider=20
  SS7 messages tunneled with SIGTRAN, the same should apply also for SIP =

  messages. So, I don't see any disadvantage&nbsp;of&nbsp;using it =
to&nbsp;do a=20
  TCAP query to a database. And I would think,&nbsp;</FONT>&nbsp;<FONT=20
  color=3D#0000ff>SIP is also not the best way to do it, considering =
that it is a=20
  protocol about "sessions". If one wants to reuse existing equipment =
-which=20
  talks TCAP-, traditional SS7 or SIGTRAN are IMO the best two=20
  choices.</FONT></SPAN></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">FM</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"></SPAN></FONT>&nbsp;</P>
  <DIV>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN =
lang=3DDE=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Frank W. =
Miller,=20
  Ph.D.</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">Chief =
Technical=20
  Officer</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">sentitO =
Networks,=20
  Inc.</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">fmiller@sentito.com</SPAN></FONT></P></DIV>
  <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> =
Chong,=20
  Koan S, ALABS [mailto:kschong@att.com] <BR><B><SPAN=20
  style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Tuesday, November 18, =
2003 11:27=20
  AM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Frank W. =
Miller;=20
  McCandless, Kevin<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">Cc:</SPAN></B>=20
  sip@ietf.org<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">Subject:</SPAN></B> RE:=20
  [Sip] Sip support for TCAP queries</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=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 =
color=3Dblue=20
  size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">A=20
  couple&nbsp;specific questions:</SPAN></FONT></P></DIV>
  <DIV>
  <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 =
color=3Dblue=20
  size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">What is=20
  the recommendation on how a SIP network element can access the =
callers' name=20
  databases (CNAM in US) in PSTN after the SIP element has determined =
that the=20
  called parties have subscribed to Caller ID?</SPAN></FONT></P></DIV>
  <DIV>
  <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 =
color=3Dblue=20
  size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">How=20
  about interworking with CLASS Auto-Recall/Callback (CCBS in ITU?)=20
  procedures?</SPAN></FONT></P></DIV>
  <DIV>
  <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 =
color=3Dblue=20
  size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">Assume=20
  TCAP over SIGTRAN&nbsp;is a bit long long =
term....</SPAN></FONT></P></DIV>
  <DIV>
  <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 =
color=3Dblue=20
  size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">Koan</SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
color=3Dblue=20
  size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">Koan=20
  S.Chong</SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
color=3Dblue=20
  size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">AT&amp;T</SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
color=3Dblue=20
  size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">1-732-420-4557</SPAN></FONT></P></DIV>
  <DIV>
  <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>
  <BLOCKQUOTE=20
  style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: =
medium none; PADDING-LEFT: 3pt; PADDING-BOTTOM: 0in; MARGIN: 5pt 0in 5pt =
3pt; BORDER-LEFT: blue 1.5pt solid; PADDING-TOP: 0in; BORDER-BOTTOM: =
medium none">
    <P class=3DMsoNormal=20
    style=3D"MARGIN-BOTTOM: 12pt; MARGIN-LEFT: 0.5in; MARGIN-RIGHT: =
0in"><FONT=20
    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> Frank W.=20
    Miller [mailto:fmiller@sentito.com]<BR><B><SPAN=20
    style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Monday, November 17, =
2003 6:18=20
    PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> =
'McCandless,=20
    Kevin'<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Cc:</SPAN></B>=20
    sip@ietf.org<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">Subject:</SPAN></B> RE:=20
    [Sip] Sip support for TCAP queries</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>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
color=3Dnavy=20
    size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">You=20
    mean other than us implementing it?&nbsp; No, I don&#8217;t think =
so.&nbsp; I=20
    think its considered Rosemary&#8217;s Baby by =
most&#8230;</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>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
color=3Dnavy=20
    size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">FM</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>
    <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 =
color=3Dnavy=20
    size=3D2><SPAN lang=3DDE=20
    style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Frank W. =
Miller,=20
    Ph.D.</SPAN></FONT></P>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
color=3Dnavy=20
    size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">Chief=20
    Technical Officer</SPAN></FONT></P>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
color=3Dnavy=20
    size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">sentitO =
Networks,=20
    Inc.</SPAN></FONT></P>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
color=3Dnavy=20
    size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">fmiller@sentito.com</SPAN></FONT></P></DIV>
    <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>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 1in"><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>=20
    sip-admin@ietf.org [mailto:sip-admin@ietf.org] <B><SPAN=20
    style=3D"FONT-WEIGHT: bold">On Behalf Of </SPAN></B>McCandless,=20
    Kevin<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> =
Monday,=20
    November 17, 2003 4:11 PM<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] Sip support =
for TCAP=20
    queries</SPAN></FONT></P>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 1in"><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: 1in"><FONT face=3DArial =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Has any progress been =
taken on=20
    the draft by Frank Miller on Carrying TCAP in SIP messages?&nbsp; =
Are there=20
    other drafts that address SIP access to SS7 database information =
through=20
    TCAP or another method?</SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 1in"><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: 1in"><FONT face=3DArial =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Thanks,</SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal style=3D"MARGIN-LEFT: 1in"><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: 1in"><FONT =
face=3D"Monotype Corsiva"=20
    size=3D4><SPAN=20
    style=3D"FONT-SIZE: 13.5pt; FONT-FAMILY: 'Monotype Corsiva'">Kevin=20
    </SPAN></FONT></P></BLOCKQUOTE></DIV></BLOCKQUOTE></BODY></HTML>

------=_NextPart_000_00AD_01C3ADCC.4BDACCF0--


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From exim@www1.ietf.org  Tue Nov 18 23:55:15 2003
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Subject: [Sip] Are the models in draft-schulzrinne-sip-register obsolete?
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Hi,

We have implemented mobility using SIP and that too based on a couple of 
registration models from the Internet Draft - 
draft-schulzrinne-sip-register-01. I figured no more versions of this draft 
have been released. This draft expired on October 2001. I would like to know 
if the models proposed in this draft have been incorporated in some RFC or 
if have they been completely rejected. If there is any such RFC, which one 
is it? Thanks in advance.

regards,
Rajarshi


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

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From exim@www1.ietf.org  Thu Nov 20 11:56:33 2003
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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'Ben Campbell'" <bcampbell@dynamicsoft.com>
Cc: Paul Kyzivat <pkyzivat@cisco.com>, Cullen Jennings <fluffy@cisco.com>,
        Adam Roach <adam@dynamicsoft.com>, Markus.Isomaki@nokia.com,
        Dean Willis
	 <dean.willis@softarmor.com>,
        Jonathan Rosenberg
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        "'sip@ietf.org'" <sip@ietf.org>
Date: Thu, 20 Nov 2003 10:54:47 -0600
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Subject: [Sip] RE: What does 3087 say (Was Re: [Simple] ad-hoc list	subscription
 s)
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Ben,

I think rather than being taken private, this discussion needs to be
discussed in SIP (or SIPPING) since it is relevant to the History-Info draft
and I think it's this topic that has been one of the primary concerns folks
have had with this work.  Although, I want to clearly state that VM is not
at all the only use case we have for this and isn't the primary motivation
at this point (and it's unfortunate that it was the initial use case that we
had when we first approached this topic).

Mary. 

-----Original Message-----
From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
Sent: Thursday, November 20, 2003 10:31 AM
To: Jonathan Rosenberg
Cc: Paul Kyzivat; Cullen Jennings; Adam Roach; Markus.Isomaki@nokia.com;
Dean Willis; simple@ietf.org
Subject: Re: What does 3087 say (Was Re: [Simple] ad-hoc list
subscriptions)


As I commented in a previous note, this seems to be vearing a bit off 
topic for SIMPLE. I am trimming the simple list from this response. If 
anyone things this discussion is really relevant to the list at large, 
feel free to put it back.

More inline:

Jonathan Rosenberg wrote:

> 
> 
> Paul Kyzivat wrote:
> 
>>
>>
>> Ben Campbell wrote:
>>
>>>
>>> VM is sort of in a gray area, as I think there is room for all sorts 
>>> of interesting, non-conventional approaches. One _really_ big problem 
>>> facing carriers these days is difficulty in deploying new services. I 
>>> think this will be the death of some of the "conventional" telecom 
>>> carriers in the long run. 3087 is all about trying to make it easier 
>>> to create new (hopefully innovative) services. This was something 
>>> that my employer (at the time of the original writing) was really 
>>> worrying about.
>>
>>
>>
>> This makes a lot of sense, and I like the approach. But I can see how 
>> a supplier of VM systems would want to hedge its bets. This approach 
>> depends on the user having a configurable proxy that can select among 
>> all the VM alternatives. Much of the intelligence has been moved from 
>> the VM to the proxy.
> 
> 
> Well, this is perhaps the essence of the long standing dispute on 
> history vs. URI-invocation as the way to talk to voicemail systems.
> 
>>
>> That is great if the VM provider is also the proxy provider. It is 
>> also probably ok if all the possible proxy providers that the VM is 
>> likely to find itself deployed with provide that functionality. But 
>> today those aren't necessarily good assumptions. To hedge the bets, 
>> the VM provider will probably want to have an alternative way to 
>> discriminate the different services on its own.
>>
>> So the tricky part is getting all the pieces to a stage where they can 
>> smoothly interoperate with one another.
> 
> 
> The other issue is that, even if you dont standardize the syntax, and 
> leave that up to configuration, there needs to be common understanding 
> of the semantics. Going back to the VM case, if invocation of a 
> voicemail drop requires the VM server to get the subscriber ID, the 
> reason for the voicemail connection, and the desired prompt, there needs 
> to be sufficient logic in the proxy to extract that information from the 
> request and make sure it finds its way into the r-uri.
> 
> As such, I think the 3087 model only works for cases where the service 
> invocation is parameterless; i.e., its a pure proxy function, without 
> the need for the proxy to know how to pull information out of messages 
> or other context, and shove it into an r-uri.

The scenario you describe is _exactly_ what 3087 was intended to cover. 
In this particular case, the VM system associates a mailbox and menu 
tree entry point with a key, in the form of a specific, arbitrarily 
named URI. The only reason it needs to be encoded in the URI is for 
human-readability.

I agree with your comment for things like ad-hoc list exploders, in that 
the list membership cannot be known in advance, so it must be encoded in 
someway.

Also, to reiterate, there were 2 concepts in 3261. The first was to use 
the r-URI for this sort of things in the first place. The second was the 
arbitrariness of said URIs. Exploders can fit into the first concept, 
but not the second. So even here, 3087 partially applies.



> 
> -Jonathan R.
> 



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Date: Thu, 20 Nov 2003 09:53:03 -0800
Subject: Re: [Sip] RE: What does 3087 say (Was Re: [Simple] ad-hoc list
	subscription s)
From: Cullen Jennings <fluffy@cisco.com>
To: Mary Barnes <mary.barnes@nortelnetworks.com>,
        "'Ben Campbell'" <bcampbell@dynamicsoft.com>
CC: Paul Kyzivat <pkyzivat@cisco.com>, Adam Roach <adam@dynamicsoft.com>,
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I think Jonathan really nailed this exactly right in the comment below.

> Jonathan Rosenberg wrote:
> 
>> As such, I think the 3087 model only works for cases where the service
>> invocation is parameterless; i.e., its a pure proxy function, without
>> the need for the proxy to know how to pull information out of messages
>> or other context, and shove it into an r-uri.


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From exim@www1.ietf.org  Thu Nov 20 15:31:30 2003
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From: Internet-Drafts@ietf.org
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Subject: [Sip] I-D ACTION:draft-ietf-sip-sctp-04.txt
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--NextPart

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

	Title		: The Stream Control Transmission Protocol as 
			  a Transport for  for the Session Initiation Protocol
	Author(s)	: J. Rosenberg, H. Schulzrinne, G. Camarillo
	Filename	: draft-ietf-sip-sctp-04.txt
	Pages		: 8
	Date		: 2003-11-20
	
This document specifies a mechanism for usage of SCTP (the Stream
Control Transmission Protocol) as the transport between SIP entities.
SCTP is a new protocol which provides several features that may prove
beneficial for transport between SIP entities which exchange a large
amount of messages, including gateways and proxies. As SIP is
transport independent, support of SCTP is a relatively
straightforward process, nearly identical to support for TCP.

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



From exim@www1.ietf.org  Fri Nov 21 03:19:31 2003
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Subject: RE: [Sip] 3-way handshake bad!
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Thread-Topic: [Sip] 3-way handshake bad!
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From: "Pugaczewski, Jack" <Jack.Pugaczewski@qwest.com>
To: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>, <sip@ietf.org>
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Question:

Why is a 3-way handshake bad ?  ISUP has used it successully for call
setup and call teardown:

Call Setup:    Setup -->; Call Proceeding <--; Connect <---
Call Teardown: Disconnect -->; Release <--; Release Complete -->



-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Paul
D.Smith
Sent: Monday, November 17, 2003 9:01 AM
To: 'sip@ietf.org'
Subject: [Sip] 3-way handshake bad!


All,

For context, see IETF 58 minutes.  There was a suggestion that the
non-INVITE avalanche problem be solved by using a 3-way handshake.  I
just want to flag that this idea can have unfortunate consequences
during congestion.

Firstly, consider a server under load.  The 3-way handshake requires the
following.

server processes request - #1
server sends response - #2
server processes request/handshake - #3.

By adding the 3rd step, we have _doubled_ the number of requests to be
handled by the server increasing the load and increasing the chances it
becomes congested.

As per my earlier e-mail, for TCP, the chances of the initial request
completing can be either 1.0 or 0.0 depending on the congestion.
Increasing the congestion at the server, increases the chance of
requests failing. For UDP, there is a 1/x (x >=3D 1) chance of the =
request
being received and processed by the server.  Similarly, there is a 1/x
chance of the 3rd request/handshake reaching the server.  This means the
chances of the 3-way handshake completing are 1/(x^2).  So adding the
third step greatly affects the chances of the 3-way handshake
completing.

Paul D.Smith
Network Protocols Group=20
Data Connection Ltd (DCL)
Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com=20
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.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 exim@www1.ietf.org  Fri Nov 21 04:11:33 2003
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sip] Registries
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Folks,

The WGLCs for the registries drafts ended some time ago. I am about to 
release new versions of the drafts that contain all the comments 
received during the last calls.

One last thing that Bill Mashall pointed out and I want to clarify. The 
SIP header fields that have to do with authorizations carry credentials 
or challenges. The ABNF looks like this:

Authorization     =  "Authorization" HCOLON credentials
Proxy-Authorization  =  "Proxy-Authorization" HCOLON credentials

WWW-Authenticate  =  "WWW-Authenticate" HCOLON challenge
Proxy-Authenticate  =  "Proxy-Authenticate" HCOLON challenge

The definitions for challenge and credentials are:

challenge           =  ("Digest" LWS digest-cln *(COMMA digest-cln))
                        / other-challenge

credentials       =  ("Digest" LWS digest-response)
                      / other-response


Digest defines one format for credentials and challenge that contains a 
bunch of parameters (username, realm, nonce, etc...) Since these 
parameters are digest specific, and not Authorization header specific, I 
chose not to add them to the registry of header field parameters. They 
are Digest parameters, not header field parameters.

Gonzalo






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



From exim@www1.ietf.org  Fri Nov 21 04:30:34 2003
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Date: Fri, 21 Nov 2003 11:24:15 +0200
From: Paraskevopoulos Pavlos <ppar@intracom.gr>
Organization: INTRACOM
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Subject: [Sip] Sip Transfer and Cisco Gateway
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Hi,

I don't know if this is the appropriate list for my case.

Assume a connection between a sip endpoint and a POTS peer, AB 
connection. The access to the POTS is through a cisco gateway. 

A (pots) ----------- GW (3620)-------------- B sipEndpoint.

|
|  (Transfer the AB connection to AC)
|

C sipEndpoint

Whenever we try to transfer the AB connection to AC, the GW responses 
with just a 202 Accepted Message.

Enabling debugging in the cisco router we get the following messages.

cause i = 0x82E0981C Mandatory information Element Missing.
Call State i=0x0A

The version of the router is:

Cisco Internetwork Operating System Software
IOS (tm) 3600 Software (C3620-IS-M), Version 12.2(13)T,  RELEASE 
SOFTWARE (fc1)

Compiled Sat 16-Nov-02 09:30 by ccai
Image text-base: 0x6000891C, data-base: 0x61848000

System image file is "flash:c3620-is-mz.122-13.T.bin"

The dial peers are correctly configured.
The transfer with Cisco 7960 and ATA188 has no problem.
I wonder if someone could help. What is the mandatory element missing?

Thanks in advance.

The messages flow is following:

   Request line: REFER sip:002106674342@146.124.120.215:5060 SIP/2.0
   Message Header
       Max-Forwards: 69
       Via: SIP/2.0/UDP 146.124.120.212;branch=z9hG4bKf08e.ecd768c1.0
       Via: SIP/2.0/UDP 146.124.120.94:5060
       From: <sip:4348@telesip1>;tag=58912
       To: <sip:002106674342@telesip1>;tag=240DD5C8-2B8
       Call-ID: 17952@146.124.120.94
       CSeq: 3 REFER
       Refer-To: <sip:8001@telesip1>
       Referred-By: <sip:4348@telesip1>
       Contact: <sip:4348@telesip1:5060>
       User-Agent: netPhone-2.9
       Content-Length: 0

   Status line: SIP/2.0 202 Accepted
   Message Header
       Via: SIP/2.0/UDP 
146.124.120.212;branch=z9hG4bKf08e.ecd768c1.0,SIP/2.0/UDP 
146.124.120.94:5060
       From: <sip:4348@telesip1>;tag=58912
       To: <sip:002106674342@telesip1>;tag=240DD5C8-2B8
       Date: Mon, 08 Mar 1993 00:01:36 GMT
       Call-ID: 17952@146.124.120.94
       Server: Cisco-SIPGateway/IOS-12.x
       CSeq: 3 REFER
       Content-Length: 0







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



From exim@www1.ietf.org  Fri Nov 21 11:07:34 2003
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Date: Thu, 20 Nov 2003 21:10:03 -0800
From: Cullen Jennings <fluffy@cisco.com>
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Subject: [Sip] SCTP and TLS
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How do you indicate that you want to use SCTP and TLS (assuming a TLS
profile for SCTP ...)



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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] SCTP and TLS
Date: Fri, 21 Nov 2003 12:47:55 -0800
Message-ID: <6677B3346233B94EBB11C06093510120024A6536@vtg-um-e2k1.sj21ad.cisco.com>
Thread-Topic: [Sip] SCTP and TLS
Thread-Index: AcOwSXgFBpZ8E5PNTfmFzxGxbkiwHAAJ1Ojg
From: "Amit Bhadoria" <bhadoria@cisco.com>
To: "Cullen Jennings" <fluffy@cisco.com>, <sip@ietf.org>
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Isn't it "sips:user@domain;transport=3Dsctp"?

--
Amit Bhadoria
Software Engineer
http://www.cisco.com=20

-----Original Message-----
From: Cullen Jennings [mailto:fluffy@cisco.com]=20
Sent: Thursday, November 20, 2003 9:10 PM
To: sip@ietf.org
Subject: [Sip] SCTP and TLS



How do you indicate that you want to use SCTP and TLS (assuming a TLS
profile for SCTP ...)



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

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



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Date: Fri, 21 Nov 2003 16:53:31 -0500
From: Manoj Bhatia <manojb@cisco.com>
Organization: Cisco Systems, Inc. , RTP, NC, USA
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Subject: Re: [Sip] Sip Transfer and Cisco Gateway
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You should be sending this to Techincal Assistance Center (TAC)
at Cisco.
You may also send at cs-sip@cisco.com for help on
Cisco related SIP issues .

If you send me the 'sh run' config on the gw, I can help you further.


thanks
-Manoj


Paraskevopoulos Pavlos wrote:

> Hi,
>
> I don't know if this is the appropriate list for my case.
>
> Assume a connection between a sip endpoint and a POTS peer, AB
> connection. The access to the POTS is through a cisco gateway.
>
> A (pots) ----------- GW (3620)-------------- B sipEndpoint.
>
> |
> |  (Transfer the AB connection to AC)
> |
>
> C sipEndpoint
>
> Whenever we try to transfer the AB connection to AC, the GW responses
> with just a 202 Accepted Message.
>
> Enabling debugging in the cisco router we get the following messages.
>
> cause i = 0x82E0981C Mandatory information Element Missing.
> Call State i=0x0A
>
> The version of the router is:
>
> Cisco Internetwork Operating System Software
> IOS (tm) 3600 Software (C3620-IS-M), Version 12.2(13)T,  RELEASE
> SOFTWARE (fc1)
>
> Compiled Sat 16-Nov-02 09:30 by ccai
> Image text-base: 0x6000891C, data-base: 0x61848000
>
> System image file is "flash:c3620-is-mz.122-13.T.bin"
>
> The dial peers are correctly configured.
> The transfer with Cisco 7960 and ATA188 has no problem.
> I wonder if someone could help. What is the mandatory element missing?
>
> Thanks in advance.
>
> The messages flow is following:
>
>    Request line: REFER sip:002106674342@146.124.120.215:5060 SIP/2.0
>    Message Header
>        Max-Forwards: 69
>        Via: SIP/2.0/UDP 146.124.120.212;branch=z9hG4bKf08e.ecd768c1.0
>        Via: SIP/2.0/UDP 146.124.120.94:5060
>        From: <sip:4348@telesip1>;tag=58912
>        To: <sip:002106674342@telesip1>;tag=240DD5C8-2B8
>        Call-ID: 17952@146.124.120.94
>        CSeq: 3 REFER
>        Refer-To: <sip:8001@telesip1>
>        Referred-By: <sip:4348@telesip1>
>        Contact: <sip:4348@telesip1:5060>
>        User-Agent: netPhone-2.9
>        Content-Length: 0
>
>    Status line: SIP/2.0 202 Accepted
>    Message Header
>        Via: SIP/2.0/UDP
> 146.124.120.212;branch=z9hG4bKf08e.ecd768c1.0,SIP/2.0/UDP
> 146.124.120.94:5060
>        From: <sip:4348@telesip1>;tag=58912
>        To: <sip:002106674342@telesip1>;tag=240DD5C8-2B8
>        Date: Mon, 08 Mar 1993 00:01:36 GMT
>        Call-ID: 17952@146.124.120.94
>        Server: Cisco-SIPGateway/IOS-12.x
>        CSeq: 3 REFER
>        Content-Length: 0
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

--
Manoj Bhatia                           |        |         |
manojb@cisco.com                       |       :|:       :|:
7025 Kit Creek Road, RTP, NC - 27709   |     :|||||:   :|||||:
Ph: 919-392-3873  Fax:919-392-6801     |  C i s c o S y s t e m s



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
&nbsp;
<br>You should be sending this to Techincal Assistance Center (TAC)
<br>at Cisco.
<br>You may also send at cs-sip@cisco.com for help on
<br>Cisco related SIP issues .
<p>If you send me the 'sh run' config on the gw, I can help you further.
<br>&nbsp;
<p>thanks
<br>-Manoj
<br>&nbsp;
<p>Paraskevopoulos Pavlos wrote:
<blockquote TYPE=CITE>Hi,
<p>I don't know if this is the appropriate list for my case.
<p>Assume a connection between a sip endpoint and a POTS peer, AB
<br>connection. The access to the POTS is through a cisco gateway.
<p>A (pots) ----------- GW (3620)-------------- B sipEndpoint.
<p>|
<br>|&nbsp; (Transfer the AB connection to AC)
<br>|
<p>C sipEndpoint
<p>Whenever we try to transfer the AB connection to AC, the GW responses
<br>with just a 202 Accepted Message.
<p>Enabling debugging in the cisco router we get the following messages.
<p>cause i = 0x82E0981C Mandatory information Element Missing.
<br>Call State i=0x0A
<p>The version of the router is:
<p>Cisco Internetwork Operating System Software
<br>IOS (tm) 3600 Software (C3620-IS-M), Version 12.2(13)T,&nbsp; RELEASE
<br>SOFTWARE (fc1)
<p>Compiled Sat 16-Nov-02 09:30 by ccai
<br>Image text-base: 0x6000891C, data-base: 0x61848000
<p>System image file is "flash:c3620-is-mz.122-13.T.bin"
<p>The dial peers are correctly configured.
<br>The transfer with Cisco 7960 and ATA188 has no problem.
<br>I wonder if someone could help. What is the mandatory element missing?
<p>Thanks in advance.
<p>The messages flow is following:
<p>&nbsp;&nbsp; Request line: REFER sip:002106674342@146.124.120.215:5060
SIP/2.0
<br>&nbsp;&nbsp; Message Header
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Max-Forwards: 69
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Via: SIP/2.0/UDP 146.124.120.212;branch=z9hG4bKf08e.ecd768c1.0
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Via: SIP/2.0/UDP 146.124.120.94:5060
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; From: &lt;sip:4348@telesip1>;tag=58912
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; To: &lt;sip:002106674342@telesip1>;tag=240DD5C8-2B8
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Call-ID: 17952@146.124.120.94
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CSeq: 3 REFER
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Refer-To: &lt;sip:8001@telesip1>
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Referred-By: &lt;sip:4348@telesip1>
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Contact: &lt;sip:4348@telesip1:5060>
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; User-Agent: netPhone-2.9
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Content-Length: 0
<p>&nbsp;&nbsp; Status line: SIP/2.0 202 Accepted
<br>&nbsp;&nbsp; Message Header
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Via: SIP/2.0/UDP
<br>146.124.120.212;branch=z9hG4bKf08e.ecd768c1.0,SIP/2.0/UDP
<br>146.124.120.94:5060
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; From: &lt;sip:4348@telesip1>;tag=58912
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; To: &lt;sip:002106674342@telesip1>;tag=240DD5C8-2B8
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date: Mon, 08 Mar 1993 00:01:36
GMT
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Call-ID: 17952@146.124.120.94
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Server: Cisco-SIPGateway/IOS-12.x
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CSeq: 3 REFER
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Content-Length: 0
<p>_______________________________________________
<br>Sip mailing list&nbsp; <a href="https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org/mailman/listinfo/sip</a>
<br>This list is for NEW development of the core SIP Protocol
<br>Use sip-implementors@cs.columbia.edu for questions on current sip
<br>Use sipping@ietf.org for new developments on the application of sip</blockquote>

<pre>--&nbsp;
Manoj Bhatia&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |
manojb@cisco.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :|:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :|:
7025 Kit Creek Road, RTP, NC - 27709&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp; :|||||:&nbsp;&nbsp; :|||||:
Ph: 919-392-3873&nbsp; Fax:919-392-6801&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp; C i s c o S y s t e m s</pre>
&nbsp;</html>

--------------463F61FE043BA1870FE38B16--


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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] SCTP and TLS
From: Cullen Jennings <fluffy@cisco.com>
To: Amit Bhadoria <bhadoria@cisco.com>, <sip@ietf.org>
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This will work in some cases where you want to insist that all hops use TLS
but if you don't care about that you need something like

transport=sctp+tls



On 11/21/03 12:47 PM, "Amit Bhadoria" <bhadoria@cisco.com> wrote:

> Isn't it "sips:user@domain;transport=sctp"?
> 
> --
> Amit Bhadoria
> Software Engineer
> http://www.cisco.com
> 
> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Thursday, November 20, 2003 9:10 PM
> To: sip@ietf.org
> Subject: [Sip] SCTP and TLS
> 
> 
> 
> How do you indicate that you want to use SCTP and TLS (assuming a TLS
> profile for SCTP ...)
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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: "Vijay K. Gurbani" <vkg@lucent.com>
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Cullen Jennings wrote:

> This will work in some cases where you want to insist that all hops 
> use TLS but if you don't care about that you need something like
> 
> transport=sctp+tls

Cullen:

Where are you heading with this...?  Do you forsee cases
where the SIP message travels encrypted over portions of the
network and in clear over others?  I would presume that once
a user specifically chose a sips URI, all hops in between need
to be secure.  No?

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


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From: Cullen Jennings <fluffy@cisco.com>
To: Dean Willis <dean.willis@softarmor.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Paul Kyzivat <pkyzivat@cisco.com>
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Subject: [Sip] Re: What does 3087 say (Was Re: [Simple] ad-hoc list
 subscriptions)
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The example you gave below is exactly what I am saying is parameterless.
Effectively there are no parameters - the URL is always
vlmner@vmhost;boxid=172.198,GR=27. I understand you CPL script could decided
to use different parameterless URLs.

If you needed to have parameters (ie something that algorithmically changed)
this would not work.

Cullen


On 11/21/03 3:02 PM, "Dean Willis" <dean.willis@softarmor.com> wrote:

>> As such, I think the 3087 model only works for cases where
>> the service 
>> invocation is parameterless; i.e., its a pure proxy function, without
>> the need for the proxy to know how to pull information out of
>> messages 
>> or other context, and shove it into an r-uri.
> 
> Not sur we agree here. The Proxy doesn't have to understand ANY of this.
> Whoever sets up the forwarding entries on the proxy has to understand it.
> 
> Case in point:
> 
> If you call +1 972 473 5455, this ends up as an INVITE to
> sip:dwillis@dynamicsoft.com.
> 
> This SIP URL is served by a proxy -- in fact, it's a really stupid antique
> proxy that hasn't been updated in years. All it knows how to do is apply a
> primitive version of CPL to the call . . .
> 
> So this week, my CPL says "Fork to all registrations".
> 
> As a result, all my phones ring -- and if none of them answer, after ten
> seconds, the Asterisk voice mail server that also registered to
> dwillis@dynamicsoft.com asnwers, and the caller gets my voice mail.
> 
> Two weeks ago, my cpl said: "Fork to all registrations, and if not answered
> in 8 seconds, cancel all forks and forward to phone number 2142821376",
> which diverted it to my cell phone, which if not answered diverts to voice
> mail.
> 
> Let's say I go to VoiceMailsRUs and rent a voice mail box. They tell me the
> CFBNA URI for my new box is:
> 
> sip:vlmner@voicemailsrus.np;boxid=172.98
> 
> Cool. My proxy doesn't have to understand this. All I have to do is paste it
> into my cpl, and we're good to go.
> 
> Now, let's say I wanted a different greeting for when my one and only SIP
> phone is busy. I go back to VoiceMailsRUs, and the tell me to use the
> optional code GR with a value of 27 to get the nifty new "Hi, this is Austin
> Powers. The swinging cat you have called is too groovy to chat right now.
> Behave, and leave a message" message.
> 
> So, I go change my CPL to forward calls, on "busy" at all forks, to
> 
> sip:vlmner@voicemailsrus.np;boxid=172.98,GR=27
> 
> And once again, despite complete ignorance of anything above the level of
> SIP response codes in my proxy, I get the service I wanted.
> 
> THERE IS ABSOLUTELY NO NEED TO MAKE VOICEMAILSRUS CONFORM TO ANY EXTERNAL
> SEMANTIC MAP.
> 
> The flip side of the rationale of 3087 is to ask "And how does VoiceMailsRUs
> feel about this?" After all they're going to buy some VM boxes from Bob, and
> some from Alice, and they want both brands to work with thir provisioning
> system, which uses a naming convention made up a demented gnome who retains
> his position by controlling incriminating photos from the 1999 company party
> . . .
> 
> So VoiceMailsRUs wants their provisioning system to be able to hook up to a
> new VM box and say to it "Make me a new box named
> vlmner@vmhost;boxid=172.198,GR=27, and provision it with the greeting
> APSWCYCISTGTCRNBALM.WAV".
> 
> VoiceMailrsRUs does NOT want to have to change their provisioning system to
> understand the behavior-specific mailbox generation semantics of Alice, or
> of Bob, or Charlie, or any other smug vendor who thinks they can replace the
> entire installed base at VoiceMailsRUs.
> 
> And that's where 3087 came from.
> 
> --
> Dean
> 
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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] SCTP and TLS
From: Cullen Jennings <fluffy@cisco.com>
To: Vijay Gurbani <vkg@lucent.com>
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On 11/21/03 2:59 PM, "Vijay K. Gurbani" <vkg@lucent.com> wrote:

> Cullen Jennings wrote:
> 
>> This will work in some cases where you want to insist that all hops
>> use TLS but if you don't care about that you need something like
>> 
>> transport=sctp+tls
> 
> Cullen:
> 
> Where are you heading with this...?  Do you forsee cases
> where the SIP message travels encrypted over portions of the
> network and in clear over others?  I would presume that once
> a user specifically chose a sips URI, all hops in between need
> to be secure.  No?
> 
> - vijay

Well I very well might phone you with a sip URL because I am not sure you
support TLS, however, my preference is TLS and I use TLS over TCP from my UA
to my proxy. My proxy does SCTP with TLS to your proxy. And your proxy does
UDP over IPSEC to you. A sips would have failed but a sip works and still
allows TLS to be used for much of the network.

The whole topic is not a big deal, we just need something with the semantic
meeting of transport=sctp+tls in the draft. I don't care what the sysntax
is. It could be transport=sctps.


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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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To: Cullen Jennings <fluffy@cisco.com>
CC: Dean Willis <dean.willis@softarmor.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Paul Kyzivat <pkyzivat@cisco.com>, Adam Roach <adam@dynamicsoft.com>,
        Markus.Isomaki@nokia.com, sip@ietf.org
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Subject: [Sip] Re: What does 3087 say (Was Re: [Simple] ad-hoc list	subscriptions)
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Cullen Jennings wrote:

> The example you gave below is exactly what I am saying is parameterless.
> Effectively there are no parameters - the URL is always
> vlmner@vmhost;boxid=172.198,GR=27. I understand you CPL script could decided
> to use different parameterless URLs.
> 
> If you needed to have parameters (ie something that algorithmically changed)
> this would not work.

OK, just to see if you mean what I think you mean--you speak of 
situations (exploders, for example) where some input cannot be know in 
advance, right? This makes sense for ad-hoc exploder lists, since they 
by nature cannot be known in advance. VM is usually _not_ a good 
example, since the number of initial states probably can be known in 
advance.

> 
> Cullen
> 
> 
> On 11/21/03 3:02 PM, "Dean Willis" <dean.willis@softarmor.com> wrote:
> 
> 
>>>As such, I think the 3087 model only works for cases where
>>>the service 
>>>invocation is parameterless; i.e., its a pure proxy function, without
>>>the need for the proxy to know how to pull information out of
>>>messages 
>>>or other context, and shove it into an r-uri.
>>
>>Not sur we agree here. The Proxy doesn't have to understand ANY of this.
>>Whoever sets up the forwarding entries on the proxy has to understand it.
>>
>>Case in point:
>>
>>If you call +1 972 473 5455, this ends up as an INVITE to
>>sip:dwillis@dynamicsoft.com.
>>
>>This SIP URL is served by a proxy -- in fact, it's a really stupid antique
>>proxy that hasn't been updated in years. All it knows how to do is apply a
>>primitive version of CPL to the call . . .
>>
>>So this week, my CPL says "Fork to all registrations".
>>
>>As a result, all my phones ring -- and if none of them answer, after ten
>>seconds, the Asterisk voice mail server that also registered to
>>dwillis@dynamicsoft.com asnwers, and the caller gets my voice mail.
>>
>>Two weeks ago, my cpl said: "Fork to all registrations, and if not answered
>>in 8 seconds, cancel all forks and forward to phone number 2142821376",
>>which diverted it to my cell phone, which if not answered diverts to voice
>>mail.
>>
>>Let's say I go to VoiceMailsRUs and rent a voice mail box. They tell me the
>>CFBNA URI for my new box is:
>>
>>sip:vlmner@voicemailsrus.np;boxid=172.98
>>
>>Cool. My proxy doesn't have to understand this. All I have to do is paste it
>>into my cpl, and we're good to go.
>>
>>Now, let's say I wanted a different greeting for when my one and only SIP
>>phone is busy. I go back to VoiceMailsRUs, and the tell me to use the
>>optional code GR with a value of 27 to get the nifty new "Hi, this is Austin
>>Powers. The swinging cat you have called is too groovy to chat right now.
>>Behave, and leave a message" message.
>>
>>So, I go change my CPL to forward calls, on "busy" at all forks, to
>>
>>sip:vlmner@voicemailsrus.np;boxid=172.98,GR=27
>>
>>And once again, despite complete ignorance of anything above the level of
>>SIP response codes in my proxy, I get the service I wanted.
>>
>>THERE IS ABSOLUTELY NO NEED TO MAKE VOICEMAILSRUS CONFORM TO ANY EXTERNAL
>>SEMANTIC MAP.
>>
>>The flip side of the rationale of 3087 is to ask "And how does VoiceMailsRUs
>>feel about this?" After all they're going to buy some VM boxes from Bob, and
>>some from Alice, and they want both brands to work with thir provisioning
>>system, which uses a naming convention made up a demented gnome who retains
>>his position by controlling incriminating photos from the 1999 company party
>>. . .
>>
>>So VoiceMailsRUs wants their provisioning system to be able to hook up to a
>>new VM box and say to it "Make me a new box named
>>vlmner@vmhost;boxid=172.198,GR=27, and provision it with the greeting
>>APSWCYCISTGTCRNBALM.WAV".
>>
>>VoiceMailrsRUs does NOT want to have to change their provisioning system to
>>understand the behavior-specific mailbox generation semantics of Alice, or
>>of Bob, or Charlie, or any other smug vendor who thinks they can replace the
>>entire installed base at VoiceMailsRUs.
>>
>>And that's where 3087 came from.
>>
>>--
>>Dean
>>
>>
>>
>>_______________________________________________
>>Simple mailing list
>>Simple@ietf.org
>>https://www1.ietf.org/mailman/listinfo/simple
>>



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Ben Campbell wrote:

> Cullen Jennings wrote:
> 
>> The example you gave below is exactly what I am saying is parameterless.
>> Effectively there are no parameters - the URL is always
>> vlmner@vmhost;boxid=172.198,GR=27. I understand you CPL script could 
>> decided
>> to use different parameterless URLs.
>>
>> If you needed to have parameters (ie something that algorithmically 
>> changed)
>> this would not work.
> 
> 
> OK, just to see if you mean what I think you mean--you speak of 
> situations (exploders, for example) where some input cannot be know in 
> advance, right? This makes sense for ad-hoc exploder lists, since they 
> by nature cannot be known in advance. VM is usually _not_ a good 
> example, since the number of initial states probably can be known in 
> advance.

I was thinking of different cases, where the information is known at 
the time a message is received, but is, as Cullen put it, 
algorithmically derived from some aspect of the request.

As an example, consider a message transcoding service that takes SIP 
MESSAGE requests, and transcodes the content from one type to another 
type, and then forwards the request to the recipient. In such a case, 
the service can be viewed like a method invocation that takes two 
parameters - the recipient and the mime type to convert to:

doTranscode(SipURI recipient, MIMEType targetMimeType);

so, if a proxy receives a request like:

MESSAGE sip:dwillis@dynamicsoft.com SIP/2.0
From: sip:fluffy@cisco.com
To: sip:dwillis@dynamicsoft.com
Content-Type: image/jpeg
.....

the proxy would forward it like this:

MESSAGE 
sip:xcoder-service@xcoders.com;recpt=dwillis@phone.dynamicsoft.com;target=image/tiff

In this case, the proxy would need to algorithmically derive the recpt 
parameter (its equal to the r-uri) and the target (its based on mime 
types that are listed as supported from the calle caps in the 
registration from the UA).

This service requires the proxy to understand its semantics, and to 
know how to compute the two parameters used in its invocation.

-Jonathan R.


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


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From: "Vijay K. Gurbani" <vkg@lucent.com>
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Cullen Jennings wrote:
> Well I very well might phone you with a sip URL because I am not sure
> you support TLS, however, my preference is TLS and I use TLS over TCP
> from my UA to my proxy. My proxy does SCTP with TLS to your proxy.
> And your proxy does UDP over IPSEC to you. A sips would have failed
> but a sip works and still allows TLS to be used for much of the
> network.
> 
> The whole topic is not a big deal, we just need something with the
> semantic meeting of transport=sctp+tls in the draft. I don't care
> what the sysntax is. It could be transport=sctps.

OK, I see.

However, couple of observations.  I seem to recall that
"transport=tls" was deprecated in favor of the sips URI which
implies it.  Second, from your description, is seems to me that
the caller is indicating a preference that says "even though the
R-URI is a sip URI, use TLS over any transport {SCTP, TCP}
wherever possible".

Logically, this could be qualified as a caller pref
since it gives guidance to intermediaries on how to handle
the request.  But since draft-ietf-sip-callerpref-10 is in
last call, I don't think it has any chance of being included
in it.

I am unsure where is the best place to indicate these
semantics: in the transport parameter of the R-URI or a new
header.  I lean towards the latter.

- vijay


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Subject: [Sip] Last Call: 'Session Timers in the Session Initiation Protocol (SIP)
 to Proposed Standard
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The IESG has received a request from the Session Initiation Protocol WG to 
consider the following document:

- 'Session Timers in the Session Initiation Protocol (SIP) '
   <draft-ietf-sip-session-timer-12.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2003-12-08.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-session-timer-12.txt


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To: "'Vijay K. Gurbani'" <vkg@lucent.com>,
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Cc: sip@ietf.org
Subject: RE: [Sip] SCTP and TLS
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Vijay K. Gurbani [mailto:vkg@lucent.com] wrote:

> I seem to recall that "transport=tls" was deprecated in favor
> of the sips URI which implies it.

Yes, this is precisely the correct answer to Cullen's question.
Section 26.2.2 of RFC 3261 addresses this:

  Note that in the SIPS URI scheme, transport is independent of TLS,
  and thus "sips:alice@atlanta.com;transport=tcp" and
  "sips:alice@atlanta.com;transport=sctp" are both valid (although
  note that UDP is not a valid transport for SIPS).  The use of
  "transport=tls" has consequently been deprecated, partly because
  it was specific to a single hop of the request.  This is a change
  since RFC 2543.

Personally, I couldn't be more pleased, since calling tls a transport
always puts my teeth on edge.

However, this leaves the issue of Via headers.

I can say:

  Via: SIP/2.0/SCTP pc33.example.com;branch=z9hG4bK776asdhds

Or:

  Via: SIP/2.0/TLS pc33.example.com;branch=z9hG4bK776asdhds

But there is no standardized syntax that allows me to say
both at the same time.

/a

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Subject: [Sip] Re: What does 3087 say (Was Re: [Simple] ad-hoc list	subscriptions)
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Jonathan Rosenberg wrote:

> 
> 
> Ben Campbell wrote:
> 
>> Cullen Jennings wrote:
>>
>>> The example you gave below is exactly what I am saying is parameterless.
>>> Effectively there are no parameters - the URL is always
>>> vlmner@vmhost;boxid=172.198,GR=27. I understand you CPL script could 
>>> decided
>>> to use different parameterless URLs.
>>>
>>> If you needed to have parameters (ie something that algorithmically 
>>> changed)
>>> this would not work.
>>
>>
>>
>> OK, just to see if you mean what I think you mean--you speak of 
>> situations (exploders, for example) where some input cannot be know in 
>> advance, right? This makes sense for ad-hoc exploder lists, since they 
>> by nature cannot be known in advance. VM is usually _not_ a good 
>> example, since the number of initial states probably can be known in 
>> advance.
> 
> 
> I was thinking of different cases, where the information is known at the 
> time a message is received, but is, as Cullen put it, algorithmically 
> derived from some aspect of the request.
> 
> As an example, consider a message transcoding service that takes SIP 
> MESSAGE requests, and transcodes the content from one type to another 
> type, and then forwards the request to the recipient. In such a case, 
> the service can be viewed like a method invocation that takes two 
> parameters - the recipient and the mime type to convert to:
> 
> doTranscode(SipURI recipient, MIMEType targetMimeType);
> 
> so, if a proxy receives a request like:
> 
> MESSAGE sip:dwillis@dynamicsoft.com SIP/2.0
> From: sip:fluffy@cisco.com
> To: sip:dwillis@dynamicsoft.com
> Content-Type: image/jpeg
> .....
> 
> the proxy would forward it like this:
> 
> MESSAGE 
> sip:xcoder-service@xcoders.com;recpt=dwillis@phone.dynamicsoft.com;target=image/tiff 
> 
> 
> In this case, the proxy would need to algorithmically derive the recpt 
> parameter (its equal to the r-uri) and the target (its based on mime 
> types that are listed as supported from the calle caps in the 
> registration from the UA).
> 
> This service requires the proxy to understand its semantics, and to know 
> how to compute the two parameters used in its invocation.

I think we are talking about the same thing. You cannot know in advance 
who the final recipient of a message requiring transcoding will be, just 
like you cannot know in advance what the membership of an ad-hoc list 
will be.

On the other hand, one probably _can_ know the target mime-types in 
advance, so one could pre-define a separate URI for each. Of course, 
this gets cumbersome if the number of target types is large.

> 
> -Jonathan R.
> 
> 



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Subject: Re: [Sip] SCTP and TLS
From: Cullen Jennings <fluffy@cisco.com>
To: Adam Roach <adam@dynamicsoft.com>, Vijay Gurbani <vkg@lucent.com>,
        <sip@ietf.org>
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sips means every hop needs TLS, transport=tls is about one hop needing TLS.

transport=tls was deprecated in a SIPS uri (because it is redundant there)
but I don't believe it was deprecated in a SIP uri. I was taking about the
case of a sip uri.  Imagine A sends with a sip uri over UDP to B with a
preloaded route to C and A wishes to indicated that a TLS protected protocol
should be used from B to C. However C can send the message to D any way it
wants so we don't want to upgrade to a sips.

This is not an unreasonable scenario - A and D could be clients in an
enterprise network that is protected in some way and B and C could be
enterprise proxies that do TLS protection across the internet.

There is also the Via problem...

I see a few possibilities for this ... (yes and I agree with Adam on the
confounding of transport type and security type into one tag was not the
most elegant move but yet that is where we are)

We need to be able to indicate in the VIA what the protocol is and also if
TLS was used or not.

Solution A
- define SCTPS to be SCTP with TLS and UDPS to be UDP with TLS and let the
via look like

Via: SIP/2.0/SCTPS pc33.example.com;branch=z9hG4bK776asdhds

Solution B
- change the SIP in the via to be either SIP or SIPS and treat this to be
sort of like the SIP or SIPS in a NAPTR record

Via: SIPS/2.0/SCTP pc33.example.com;branch=z9hG4bK776asdhds

Solution C
- add a security tag that defines if security TLS is used or not
Via: SIP/2.0/SCTP pc33.example.com;branch=z9hG4bK776asdhds;security=TLS

There are backwards compatibility issues with all of these.

Next we need to be able to represent that we want security for a single hop
in a URI ( could be a request URI or route or others)

Solution 1
make the transport tag be one of UDP, TCP, SCTP, DCCP and add a security tag
that could be TLS
sip:fluffy@securityforever.org;transport=SCTP;security=TLS

Solution 2
define some more transports of udp+tls sctp+tls, dccp, dccp+tls. Note that
tls alone means tls over tcp.
sip:fluffy@securityforever.org;transport=SCTP+TLS
Also a URL with transport=+TLS would mean that any transport could be used
but it must include TLS.

Given we have two security protocols (nothing and tls) and a handful of
transports (UDP, TCP, SCTP, DCCP) - Solution A and 2 could work without an
explosion of labels. Solution 1 is tricky to make sure no one ever sends
transport=tcp;security=tls to someone expecting transport=tls. Solution B
will no doubt break someone who gets a "Via: SIPS/2.0/TCP" on a TLS
connection. Solution C would be the right things if we went with Solution 2
instead of Solution 1.

Sorry this is such a disorganized email. Thoughts?

Thanks, Cullen



On 11/24/03 8:50 AM, "Adam Roach" <adam@dynamicsoft.com> wrote:

> Vijay K. Gurbani [mailto:vkg@lucent.com] wrote:
> 
>> I seem to recall that "transport=tls" was deprecated in favor
>> of the sips URI which implies it.
> 
> Yes, this is precisely the correct answer to Cullen's question.
> Section 26.2.2 of RFC 3261 addresses this:
> 
> Note that in the SIPS URI scheme, transport is independent of TLS,
> and thus "sips:alice@atlanta.com;transport=tcp" and
> "sips:alice@atlanta.com;transport=sctp" are both valid (although
> note that UDP is not a valid transport for SIPS).  The use of
> "transport=tls" has consequently been deprecated, partly because
> it was specific to a single hop of the request.  This is a change
> since RFC 2543.
> 
> Personally, I couldn't be more pleased, since calling tls a transport
> always puts my teeth on edge.
> 
> However, this leaves the issue of Via headers.
> 
> I can say:
> 
> Via: SIP/2.0/SCTP pc33.example.com;branch=z9hG4bK776asdhds
> 
> Or:
> 
> Via: SIP/2.0/TLS pc33.example.com;branch=z9hG4bK776asdhds
> 
> But there is no standardized syntax that allows me to say
> both at the same time.
> 
> /a
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 


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From: Adam Roach <adam@dynamicsoft.com>
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        Adam Roach
	 <adam@dynamicsoft.com>, Vijay Gurbani <vkg@lucent.com>,
        sip@ietf.org
Subject: RE: [Sip] SCTP and TLS
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Cullen Jennings [mailto:fluffy@cisco.com] wrote:

> transport=tls was deprecated in a SIPS uri (because it is 
> redundant there)
> but I don't believe it was deprecated in a SIP uri.

I think the intention was to deprecate it everywhere, not just
in sips: URIs. To emphasize the final sentence of the passage
I quoted (regarding deprecation of the "transport=tls"
indication):

  "This is a change since RFC 2543."

Since the "sips:" scheme is not defined in 2543, it seems a
bit of a stretch to call this deprecation a change unless it
applies at least to "sip:" URIs.

If you find 3261 less than coherent on this topic, there
is a perfectly reasonable explanation, best summarized as:

  "We apologise for the fault in the hop-by-hop security model.
   Those responsible have been sacked, and hop-by-hop security
   has been completed in an entirely different style at great
   expense at the last minute."

/a

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Subject: RE: [Sip] SCTP and TLS
Date: Tue, 25 Nov 2003 13:14:14 +0200
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> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
> Cullen Jennings
> Sent: 25.November.2003 04:17
> To: Adam Roach; Vijay Gurbani; sip@ietf.org
> Subject: Re: [Sip] SCTP and TLS
>=20
>=20
>=20
> sips means every hop needs TLS, transport=3Dtls is about one=20
> hop needing TLS.
>=20
> transport=3Dtls was deprecated in a SIPS uri (because it is=20
> redundant there)
> but I don't believe it was deprecated in a SIP uri. I was=20
> taking about the
> case of a sip uri.  Imagine A sends with a sip uri over UDP=20
> to B with a
> preloaded route to C and A wishes to indicated that a TLS=20
> protected protocol
> should be used from B to C. However C can send the message to=20
> D any way it
> wants so we don't want to upgrade to a sips.

A can insert a route-header with a sips URI. What is wrong with that?

/Hisham

>=20
> This is not an unreasonable scenario - A and D could be clients in an
> enterprise network that is protected in some way and B and C could be
> enterprise proxies that do TLS protection across the internet.
>=20
> There is also the Via problem...
>=20
> I see a few possibilities for this ... (yes and I agree with=20
> Adam on the
> confounding of transport type and security type into one tag=20
> was not the
> most elegant move but yet that is where we are)
>=20
> We need to be able to indicate in the VIA what the protocol=20
> is and also if
> TLS was used or not.
>=20
> Solution A
> - define SCTPS to be SCTP with TLS and UDPS to be UDP with=20
> TLS and let the
> via look like
>=20
> Via: SIP/2.0/SCTPS pc33.example.com;branch=3Dz9hG4bK776asdhds
>=20
> Solution B
> - change the SIP in the via to be either SIP or SIPS and=20
> treat this to be
> sort of like the SIP or SIPS in a NAPTR record
>=20
> Via: SIPS/2.0/SCTP pc33.example.com;branch=3Dz9hG4bK776asdhds
>=20
> Solution C
> - add a security tag that defines if security TLS is used or not
> Via: SIP/2.0/SCTP=20
> pc33.example.com;branch=3Dz9hG4bK776asdhds;security=3DTLS
>=20
> There are backwards compatibility issues with all of these.
>=20
> Next we need to be able to represent that we want security=20
> for a single hop
> in a URI ( could be a request URI or route or others)
>=20
> Solution 1
> make the transport tag be one of UDP, TCP, SCTP, DCCP and add=20
> a security tag
> that could be TLS
> sip:fluffy@securityforever.org;transport=3DSCTP;security=3DTLS
>=20
> Solution 2
> define some more transports of udp+tls sctp+tls, dccp,=20
> dccp+tls. Note that
> tls alone means tls over tcp.
> sip:fluffy@securityforever.org;transport=3DSCTP+TLS
> Also a URL with transport=3D+TLS would mean that any transport=20
> could be used
> but it must include TLS.
>=20
> Given we have two security protocols (nothing and tls) and a=20
> handful of
> transports (UDP, TCP, SCTP, DCCP) - Solution A and 2 could=20
> work without an
> explosion of labels. Solution 1 is tricky to make sure no one=20
> ever sends
> transport=3Dtcp;security=3Dtls to someone expecting=20
> transport=3Dtls. Solution B
> will no doubt break someone who gets a "Via: SIPS/2.0/TCP" on a TLS
> connection. Solution C would be the right things if we went=20
> with Solution 2
> instead of Solution 1.
>=20
> Sorry this is such a disorganized email. Thoughts?
>=20
> Thanks, Cullen
>=20
>=20
>=20
> On 11/24/03 8:50 AM, "Adam Roach" <adam@dynamicsoft.com> wrote:
>=20
> > Vijay K. Gurbani [mailto:vkg@lucent.com] wrote:
> >=20
> >> I seem to recall that "transport=3Dtls" was deprecated in favor
> >> of the sips URI which implies it.
> >=20
> > Yes, this is precisely the correct answer to Cullen's question.
> > Section 26.2.2 of RFC 3261 addresses this:
> >=20
> > Note that in the SIPS URI scheme, transport is independent of TLS,
> > and thus "sips:alice@atlanta.com;transport=3Dtcp" and
> > "sips:alice@atlanta.com;transport=3Dsctp" are both valid (although
> > note that UDP is not a valid transport for SIPS).  The use of
> > "transport=3Dtls" has consequently been deprecated, partly because
> > it was specific to a single hop of the request.  This is a change
> > since RFC 2543.
> >=20
> > Personally, I couldn't be more pleased, since calling tls a=20
> transport
> > always puts my teeth on edge.
> >=20
> > However, this leaves the issue of Via headers.
> >=20
> > I can say:
> >=20
> > Via: SIP/2.0/SCTP pc33.example.com;branch=3Dz9hG4bK776asdhds
> >=20
> > Or:
> >=20
> > Via: SIP/2.0/TLS pc33.example.com;branch=3Dz9hG4bK776asdhds
> >=20
> > But there is no standardized syntax that allows me to say
> > both at the same time.
> >=20
> > /a
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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Subject: RE: [Sip] PUBLISH message rates
Date: Tue, 25 Nov 2003 18:41:10 +0200
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Thread-Topic: [Sip] PUBLISH message rates
Thread-Index: AcOqu8UC2YIGGfwATFiXANwkoKPYVQItexRQ
To: <pkyzivat@cisco.com>, <hgs@cs.columbia.edu>
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Hi Paul,

 > -----Original Message-----
 > From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
 > Sent: 14 November, 2003 16:30
 > To: Henning Schulzrinne
 > Cc: Khartabil Hisham (NMP-MSW/Helsinki); Niemi Aki (NMP/Helsinki);
 > jdrosen@dynamicsoft.com; adam@dynamicsoft.com; sip@ietf.org
 > Subject: Re: [Sip] PUBLISH message rates
 >=20
 >=20
 >=20
 >=20
 > > hisham.khartabil@nokia.com wrote:
 > >=20
 > >> The only way to do this right is to do what RTCP does with report
 > >> interval, and that is to adjust the rate of PUBLISH using some
 > >> formula that takes into account the number of publishers=20
 > at a given
 > >> time. The more publishers that join, the less the rate at which
 > >> PUBLISH is sent at. The compositors need to communicate that
 > >> calculated rate to the publishers somehow. Do we want to=20
 > make it that
 > >> complicated? If not, then I suggest adding text the says if the
 > >> compositor is overloaded, it can send a 5xx response with
 > >> retry-after.
 >=20
 > Of course use of 5xx with retry-after wastes the effort the=20
 > compositor=20
 > has already expended in processing the publish up to that point.
 >=20
 > It might be interesting to make a change permitting use of the=20
 > Retry-After header with a 2xx response. Then the compositor=20
 > could accept=20
 >   a publish and at the same time ensure that the next one=20
 > doesn't come=20
 > too soon. This wouldn't be useful in all cases, but maybe in some.

I think this is an interesting idea. However, it doesn't seem to quite =
fit the Retry-After meaning. After all, the request did succeed since it =
received a 200 OK.=20

Cheers,
Aki

 > 	Paul
 >=20
 >=20

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



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[Trimming the cc ballast]

> I think this is an interesting idea. However, it doesn't seem to
> quite fit the Retry-After meaning. After all, the request did succeed
> since it received a 200 OK.

I think general SIP design philosophy is to avoid overloading of 
distinct meanings on the same parameter. We don't want to have to say 
'if the moon is in the first quarter, header Foo means that, but 
otherwise it means something different'. If this is valuable, then it 
deserves its own header field.

The particular approach also has the problem of introducing additional 
error cases and state: "sorry, PUBLISH, you were 5 seconds too early, 
please try again in 5 seconds".

That said, I share the doubts raised that any unconditional and 
content-unaware filtering is all that useful in practice.

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--NextPart

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

	Title		: The Internet Assigned Number Authority Universal 
			  Resource Identifier Parameter Registry for the 
			  Session Initiation Protocol
	Author(s)	: G. Camarillo
	Filename	: draft-ietf-sip-uri-parameter-reg-01.txt
	Pages		: 5
	Date		: 2003-11-25
	
This document creates an IANA registry for SIP URI and SIPS URI
parameters. It also lists the already existing parameters to be used
as initial values for that registry.

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

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Paul Kyzivat [mailto:pkyzivat@cisco.com] wrote:

> It might be interesting to make a change permitting use of the 
> Retry-After header with a 2xx response. Then the compositor 
> could accept 
>   a publish and at the same time ensure that the next one 
> doesn't come 
> too soon. This wouldn't be useful in all cases, but maybe in some.

I think you're starting down the same path that Robert has already
put some good work into. You probably want to take a look at:

http://www.ietf.org/internet-drafts/draft-sparks-sipping-load-00.txt

/a

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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		: The Internet Assigned Number Authority Header Field
			  Parameter Registry for the Session Initiation Protocol
	Author(s)	: G. Camarillo
	Filename	: draft-ietf-sip-parameter-registry-01.txt
	Pages		: 6
	Date		: 2003-11-25
	
This document creates an IANA registry for SIP header field
parameters. It also lists the already existing parameters to be used
as initial values for that registry.

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From exim@www1.ietf.org  Tue Nov 25 16:28:35 2003
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Henning Schulzrinne wrote:
> 
>> I think this is an interesting idea. However, it doesn't seem to
>> quite fit the Retry-After meaning. After all, the request did succeed
>> since it received a 200 OK.
> 
> I think general SIP design philosophy is to avoid overloading of 
> distinct meanings on the same parameter. We don't want to have to say 
> 'if the moon is in the first quarter, header Foo means that, but 
> otherwise it means something different'. If this is valuable, then it 
> deserves its own header field.

I agree that bundling Retry-After with a 2xx response is kind of zany, 
and there are probably better ways to solve the problem.

> The particular approach also has the problem of introducing additional 
> error cases and state: "sorry, PUBLISH, you were 5 seconds too early, 
> please try again in 5 seconds".
> 
> That said, I share the doubts raised that any unconditional and 
> content-unaware filtering is all that useful in practice.

At least when the problem is pushed back to the publisher (however that 
is achieved),  it is in the best position to deal with the problem in an 
intelligent way.

Adam Roach wrote:
> 
> I think you're starting down the same path that Robert has already
> put some good work into. You probably want to take a look at:
> 
> http://www.ietf.org/internet-drafts/draft-sparks-sipping-load-00.txt

Yes, this is clearly a special case of what Robert is talking about, and 
he has put way more thought into it than I did.

One difference - I *think* robert is trying to put pressure on a source 
to slow down everything from it. But maybe the pressure needs to be more 
selective, as in: slow down the PUBLISH requests, but its still ok to 
send an INVITE right away.

Of course that also makes an already intractible problem even harder. I 
have the feeling we are just groping in the dark here, but maybe that is 
how it has to start.

	Paul


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Cullen Jennings wrote:

> sips means every hop needs TLS, transport=tls is about one hop needing 
> TLS.
> This is not an unreasonable scenario - A and D could be clients in an
> enterprise network that is protected in some way and B and C could be
> enterprise proxies that do TLS protection across the internet.

At least this particular configuration is mitigated by loose-
routing the enterprise proxies.  The problem is, of course, how
does A know the URI of B's enterprise proxy in any authoritative
manner (outside of the 3GPP extensions)?  It could always guess
and construct a request of the form:

    INVITE sip:B@domain-b.com SIP/2.0
    Route: <sips:domain-b.com>
    ...

and send it using TLS to B's enterprise proxy, which then delivers
it to B.  At least the request transits the network in a secure
manner.

Alternatively, B can publish a URI of the form:
    sip:B@domain-b.com?Route=sips:domain-B.com

But I am unsure how many current UACs would form the correct
request given the above URI.  And furthermore, B has to be
cognizant enough to know that her UA does not support TLS but
her enterprise proxy does so she can leverage this knowledge
in publishing such a URI.

> There is also the Via problem...
[...]
> There are backwards compatibility issues with all of these.

Yes, although Solution C is probably the least intrusive.
Implementations that don't understand "security=TLS" parameter
will simply ignore it (and hopefully, re-generate it if they
are an intermediary).

> Next we need to be able to represent that we want security for a single 
> hop in a URI ( could be a request URI or route or others)
> 
> Solution 1
> make the transport tag be one of UDP, TCP, SCTP, DCCP and add a security 
> tag that could be TLS
> sip:fluffy@securityforever.org;transport=SCTP;security=TLS
> 
> Solution 2
> define some more transports of udp+tls sctp+tls, dccp, dccp+tls. Note that
> tls alone means tls over tcp.
> sip:fluffy@securityforever.org;transport=SCTP+TLS
> Also a URL with transport=+TLS would mean that any transport could be used
> but it must include TLS.
> 
> Given we have two security protocols (nothing and tls) and a handful of
> transports (UDP, TCP, SCTP, DCCP) - Solution A and 2 could work without an
> explosion of labels. Solution 1 is tricky to make sure no one ever sends
> transport=tcp;security=tls to someone expecting transport=tls. 

For that reason I prefer Solution 2 since it specifies a transport
plus a security tag.  But, the biggest problem is that these are
not backwards compatible to deployed gear.  So, besides an
academic exercise in curiosity, what are the chances of
implementation?

It appears to me that the most ubiquitous and least intrusive
solution is to use some form of loose source routing to
protect the requests in portions of the network that are not
completely trustworthy.

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


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Subject: Re: [Sip] SCTP and TLS
From: Cullen Jennings <fluffy@cisco.com>
To: Vijay Gurbani <vkg@lucent.com>
CC: Adam Roach <adam@dynamicsoft.com>, <sip@ietf.org>
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> plus a security tag.  But, the biggest problem is that these are
> not backwards compatible to deployed gear.  So, besides an
> academic exercise in curiosity, what are the chances of
> implementation?
>

There is not really any SCTP or TLS over SCTP or UDP or DCCP deployed so I
think we could be backwards compatible with existing TLS stuff. I'm not
seeing a way to be backwards compatible with everything but the current
stuff looks broken so I think the odds of something other than what is
currently specified deploying are fairly high. This is not a "I don't like
the current thing issues" this is the current thing can't construct a usable
via for a stateless proxy using SCTP with TLS.




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From exim@www1.ietf.org  Wed Nov 26 00:05:30 2003
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Subject: RE: [Sip] SCTP and TLS
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Adam Roach <adam@dynamicsoft.com>
Cc: "'Cullen Jennings'" <fluffy@cisco.com>, Vijay Gurbani <vkg@lucent.com>,
        sip@ietf.org
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	 <9BF66EBF6BEFD942915B4D4D45C051F3E86659@dyn-tx-exch-001.dynamicsoft.com>
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;transport=tls was utterly deprecated.

This was done quickly, and it missed the case that somebody
wanted to specify using tls for just the next hop by putting
something in the URI.

All we have available for that now is DNS SRV which operates
at the whole-host level, not the individual resource level.

I see this as an unnecessary shortcoming.

RjS

On Tue, 2003-11-25 at 00:28, Adam Roach wrote:
> Cullen Jennings [mailto:fluffy@cisco.com] wrote:
> 
> > transport=tls was deprecated in a SIPS uri (because it is 
> > redundant there)
> > but I don't believe it was deprecated in a SIP uri.
> 
> I think the intention was to deprecate it everywhere, not just
> in sips: URIs. To emphasize the final sentence of the passage
> I quoted (regarding deprecation of the "transport=tls"
> indication):
> 
>   "This is a change since RFC 2543."
> 
> Since the "sips:" scheme is not defined in 2543, it seems a
> bit of a stretch to call this deprecation a change unless it
> applies at least to "sip:" URIs.
> 
> If you find 3261 less than coherent on this topic, there
> is a perfectly reasonable explanation, best summarized as:
> 
>   "We apologise for the fault in the hop-by-hop security model.
>    Those responsible have been sacked, and hop-by-hop security
>    has been completed in an entirely different style at great
>    expense at the last minute."
> 
> /a
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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Cc: Adam Roach <adam@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] SCTP and TLS
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Cullen Jennings [mailto:fluffy@cisco.com] wrote:

> There is not really any SCTP or TLS over SCTP or UDP or DCCP 
> deployed so I think we could be backwards compatible with
> existing TLS stuff. I'm not seeing a way to be backwards
> compatible with everything but the current stuff looks broken
> so I think the odds of something other than what is currently
> specified deploying are fairly high. This is not a "I don't
> like the current thing issues" this is the current thing can't 
> construct a usable via for a stateless proxy using SCTP with TLS.

Actually, I think we have a choice between backwards-compatibility
and elegance.

For example, we could choose designations like the following
for a sort of elegance:

UDP:       SIP/2.0/UDP
UDP+TLS:   SIP/2.0/TLS/UDP
TCP:       SIP/2.0/TCP
TCP+TLS:   SIP/2.0/TLS/TCP
SCTP:      SIP/2.0/SCTP
SCTP+TLS:  SIP/2.0/TLS/SCTP
DCCP:      SIP/2.0/DCCP
DCCP+TLS:  SIP/2.0/TLS/DCCP

Alternately, we could retain backwards compatibility (but have
people forever wonder about the TCP aberration) by using
designations like:

UDP:       SIP/2.0/UDP
UDP+TLS:   SIP/2.0/TLS/UDP
TCP:       SIP/2.0/TCP
TCP+TLS:   SIP/2.0/TLS
SCTP:      SIP/2.0/SCTP
SCTP+TLS:  SIP/2.0/TLS/SCTP
DCCP:      SIP/2.0/DCCP
DCCP+TLS:  SIP/2.0/TLS/DCCP

Because of practical considerations, I would propose that the
second approach is probably the best path forward.

/a

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Subject: RE: [Sip] SCTP and TLS
Date: Wed, 26 Nov 2003 09:01:57 +0200
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Why isn't it enough to state that if a request was received using TLS, =
the response must sent using TLS? It requires proxies and UASs to keep a =
state though.

Regards,
Hisham

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
> Adam Roach
> Sent: 26.November.2003 07:10
> To: 'Cullen Jennings'; Vijay Gurbani
> Cc: Adam Roach; sip@ietf.org
> Subject: RE: [Sip] SCTP and TLS
>=20
>=20
> Cullen Jennings [mailto:fluffy@cisco.com] wrote:
>=20
> > There is not really any SCTP or TLS over SCTP or UDP or DCCP=20
> > deployed so I think we could be backwards compatible with
> > existing TLS stuff. I'm not seeing a way to be backwards
> > compatible with everything but the current stuff looks broken
> > so I think the odds of something other than what is currently
> > specified deploying are fairly high. This is not a "I don't
> > like the current thing issues" this is the current thing can't=20
> > construct a usable via for a stateless proxy using SCTP with TLS.
>=20
> Actually, I think we have a choice between backwards-compatibility
> and elegance.
>=20
> For example, we could choose designations like the following
> for a sort of elegance:
>=20
> UDP:       SIP/2.0/UDP
> UDP+TLS:   SIP/2.0/TLS/UDP
> TCP:       SIP/2.0/TCP
> TCP+TLS:   SIP/2.0/TLS/TCP
> SCTP:      SIP/2.0/SCTP
> SCTP+TLS:  SIP/2.0/TLS/SCTP
> DCCP:      SIP/2.0/DCCP
> DCCP+TLS:  SIP/2.0/TLS/DCCP
>=20
> Alternately, we could retain backwards compatibility (but have
> people forever wonder about the TCP aberration) by using
> designations like:
>=20
> UDP:       SIP/2.0/UDP
> UDP+TLS:   SIP/2.0/TLS/UDP
> TCP:       SIP/2.0/TCP
> TCP+TLS:   SIP/2.0/TLS
> SCTP:      SIP/2.0/SCTP
> SCTP+TLS:  SIP/2.0/TLS/SCTP
> DCCP:      SIP/2.0/DCCP
> DCCP+TLS:  SIP/2.0/TLS/DCCP
>=20
> Because of practical considerations, I would propose that the
> second approach is probably the best path forward.
>=20
> /a
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Subject: [Sip] Entity-tags in PUBLISH
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Hi,

As we discussed in Minneapolis, the PUBLISH draft needs additional text =
covering the use of etags, and the relationship to HTTP in this respect. =
One related thing that came up was that although the etags in PUBLISH =
have been copied from RFC 2616, there are some minor differences, mainly =
due to the fact that with PUBLISH, we don't have use for features used =
with for example the GET and HEAD methods:

   - We have dropped the weak etag concept, because PUBLISH etags are =
basically=20
     always strong. As a side effect, the syntax is also different, =
i.e., an etag=20
     is a token, instead of quoted-string with an optional "weakness" =
tag prefix.=20
     This seemed reasonable, since this is the common way to =
representing=09
     similar things in SIP, and after removing the weak tag, the quotes =
were=20
     superfluous.

   - We have dropped multiple etags in If-Match header, because in =
PUBLISH, the=20
     EPA will always only modify/remove/refresh its own publications. =
This=20
     corresponds to a conditional PUT in HTTP/1.1, where only a single =
entity-tag=20
     is present in If-Match.

   - We have dropped the "*" valued If-Match. An "If-Match: *" =
corresponds to a=20
     PUBLISH without the If-Match header, so there is no need for this =
special=20
     value.

   - In HTTP/1.1, the origin server SHOULD create an etag, whereas in =
PUBLISH,=20
     the ESC MUST always create an etag.

The main functionality of etags is the same in my opinion, even with the =
above "focused" use of them. But I would still like some input on =
whether the working group feels these differences are considerable =
enough that we should *not* inherit the syntax for these headers from =
HTTP/1.1, but create similar but distinct header field names, and =
response code.

Cheers,
Aki


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From: "Medhavi Bhatia" <mbhatia@nextone.com>
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Folks,

Is this draft being revived ? We had some questions/comments on the syntax
specified in it and were wondering if this was being worked on or dropped.

Medhavi.

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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charset=3Diso-8859-1">
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<DIV><FONT face=3DCourier size=3D2>Folks,</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier size=3D2>Is this draft being revived ? We had =
some=20
questions/comments on the syntax specified in it and were wondering if =
this was=20
being worked on or dropped.</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier size=3D2>Medhavi.</FONT></DIV></BODY></HTML>

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From exim@www1.ietf.org  Thu Nov 27 02:42:57 2003
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From: "Hu Yijun,BISC R&D SA(BJ)" <Yijun.Hu@BISC.SIEMENS.COM.CN>
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Subject: [Sip] a question about offer/answer model
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hi all,
 
I have a question about the SDP offer/answer model.
 
In the section 6 of rfc3264, it claimed that the answerer can reject an offer stream by setting the port number to zero. 
 
My question is: if an SDP offer, included in INVITE, is reject by the answerer, and the corresponding SDP answer is included in 200 OK, then how to deal with this answer in the offerer(UAC)? Should the UAC treat it as a successful SDP negotiation, or send a BYE immediately after sending the ACK?

-----------------------------------------------------------------------------
Best regards,
Hu Yijun
BISC, R&D SA
Tel.:+86-10-64346-660
E-mail: yijun.hu@bisc.siemens.com.cn 

 

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<DIV><FONT face=&#23435;&#20307; size=2><SPAN class=781060907-27112003>hi 
all,</SPAN></FONT></DIV>
<DIV><FONT face=&#23435;&#20307; size=2><SPAN 
class=781060907-27112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=&#23435;&#20307; size=2><SPAN class=781060907-27112003>I have a question about 
the SDP offer/answer model.</SPAN></FONT></DIV>
<DIV><FONT face=&#23435;&#20307; size=2><SPAN 
class=781060907-27112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=&#23435;&#20307; size=2><SPAN class=781060907-27112003>In the section 6 of 
rfc3264, it claimed that the answerer can reject an offer stream&nbsp;by setting 
the port number to zero. </SPAN></FONT></DIV>
<DIV><FONT face=&#23435;&#20307; size=2><SPAN 
class=781060907-27112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=&#23435;&#20307; size=2><SPAN class=781060907-27112003>My question is: if an 
SDP offer, included in INVITE, is reject by the answerer, and&nbsp;the 
corresponding SDP answer is included in 200 OK, then how to deal with this 
answer in the offerer(UAC)?&nbsp;Should the UAC treat it as a successful SDP 
negotiation, or send a BYE immediately after sending the 
ACK?</SPAN></FONT></DIV>
<P><FONT 
size=2>-----------------------------------------------------------------------------<BR>Best 
regards,<BR>Hu Yijun<BR>BISC, R&amp;D SA<BR>Tel.:+86-10-64346-660<BR>E-mail: 
yijun.hu@bisc.siemens.com.cn</FONT> </P>
<DIV>&nbsp;</DIV></BODY></HTML>

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Subject: [Sip] Question:transaction layer: timer B
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implementation question..

when shud timer B be stopped after reaching "proceeding" state or
"confirmed" state?






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From: "Christian Stredicke" <stredicke@snom.de>
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Subject: [Sip] From and To-header confusion in draft-ietf-sip-replaces-04.txt
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Hi!

 

I think there is some from and to-header confusion in the Replaces
header.

 

Depending on who initiates the call, the to and from must be exchanged.
Maybe it would be better to use the term local and remote-tag instead or
to allow mixing from and to.

 

Christian

--

Christian Stredicke

tel:+49.30.39833.401 (ENUM & PSTN)


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<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Verdana'>Hi!</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Verdana'>I think there is some from and to-header =
confusion
in the Replaces header.</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Verdana'>Depending on who initiates the call, the to =
and
from must be exchanged. Maybe it would be better to use the term local =
and
remote-tag instead or to allow mixing from and to.</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Verdana'>Christian</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Verdana'>Christian Stredicke</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Verdana'>tel:+49.30.39833.401 (ENUM &amp; =
PSTN)</span></font></p>

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Cc: "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] From and To-header confusion in draft-ietf-sip-replaces
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Christian,
 
If I understand you correctly, you are pointing out the following.
 
If a target receives an INVITE with Replaces headers, it has to somehow use
the Replaces To/From fields to identify a dialog.  If this dialog were
originally started by the target then there may be a mapping of:
 
To field of original INVITE ==> target's idea of remote tag
From field of original INVITE ==> target's idea of local tag.
 
However if the target did not start the dialog, the mapping is reversed.
 
To field of original INVITE ==> target's idea of local tag
From field of original INVITE ==> target's idea of remote tag.
 
And your question is... how does the Replaces To/From get mapped to the
targets idea of local/remote tags?
 
Unfortunately, this can't be resolved into a simple mapping.  A third party
REFER modifying an INVITE may not know who started the dialog and so
renaming the Replaces fields makes no difference since the referrer still
doesn't know which tag to place in which field.  The target has to perform a
lookup using both mappings - often one will find a dialog and the other will
not.
 
There is a possible issue with "loopback" type dialogs where a UA, with
multiple personalities (i.e. a gateway) may have two dialogs, one in each
direction.  In this case, it may be impossible to map to a single dialog,
which is where the following sentence from page 5 of the draft is headed...
 
  "If the Replaces header field matches more than one dialog, the UA MUST
act as if no match is found."
 
Is this a correct understanding of your posting?
 
Regards,
Paul DS.
Paul D.Smith 
Network Protocols Group 
Data Connection Ltd (DCL) 
Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com 
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
<http://www.dataconnection.com/>  
-----Original Message-----
From: Christian Stredicke [mailto:stredicke@snom.de]
Sent: 27 November 2003 17:45
To: SIP (E-mail)
Subject: [Sip] From and To-header confusion in
draft-ietf-sip-replaces-04.txt



Hi!

 

I think there is some from and to-header confusion in the Replaces header.

 

Depending on who initiates the call, the to and from must be exchanged.
Maybe it would be better to use the term local and remote-tag instead or to
allow mixing from and to.

 

Christian

--

Christian Stredicke

tel:+49.30.39833.401 (ENUM & PSTN)


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<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 
size=2>Christian,</FONT></SPAN></DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 size=2>If I 
understand you correctly,&nbsp;you are pointing out the 
following.</FONT></SPAN></DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 size=2>If a 
target receives an INVITE with Replaces headers, it has to somehow use the 
Replaces To/From fields to identify a dialog.&nbsp; If this dialog were 
originally started by the target then there may be a mapping 
of:</FONT></SPAN></DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 size=2>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 size=2>To 
field of original INVITE ==&gt; target's idea of remote tag</FONT></SPAN></DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 size=2>From 
field of original INVITE ==&gt; target's idea of local 
tag.</FONT></SPAN></DIV></FONT></SPAN></DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 
size=2>However if the target did not start the dialog, the mapping is 
reversed.</FONT></SPAN></DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 size=2>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 size=2>To 
field of original INVITE ==&gt; target's idea of local tag</FONT></SPAN></DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 size=2>From 
field of original INVITE ==&gt; target's idea of remote 
tag.</FONT></SPAN></DIV></FONT></SPAN></DIV>
<DIV><SPAN class=319184209-28112003></SPAN>&nbsp;</DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 size=2>And 
your question is... how does the Replaces To/From get mapped to the targets idea 
of local/remote tags?</FONT></SPAN></DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><FONT face=Arial><FONT color=#008000><FONT size=2><SPAN 
class=319184209-28112003>Unfortunately, this can't&nbsp;be resolved into a 
simple mapping.&nbsp; A third party REFER modifying an INVITE may not know who 
started the dialog and so renaming the Replaces fields makes no difference since 
the referrer still doesn't know which tag to place in which 
field.</SPAN>&nbsp;<SPAN class=319184209-28112003> T</SPAN><SPAN 
class=319184209-28112003>he target has to perform a lookup using both mappings - 
often one will&nbsp;find a dialog and the other will 
not.</SPAN></FONT></FONT></FONT></DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 size=2>There 
is a possible issue with "loopback" type dialogs where a UA, with multiple 
personalities (i.e. a gateway) may have two dialogs, one in each 
direction.&nbsp; In this case, it may be impossible to map to a single dialog, 
which is where the following sentence from page 5 of the draft is 
headed...</FONT></SPAN></DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=319184209-28112003><FONT face=Arial color=#008000 size=2>&nbsp; 
"If the Replaces header field matches more than one dialog, the UA&nbsp;MUST act 
as if no match is found."</FONT></SPAN></DIV>
<DIV><SPAN class=319184209-28112003></SPAN><SPAN 
class=319184209-28112003></SPAN><FONT face=Tahoma><FONT face=Arial color=#008000 
size=2></FONT><FONT face=Arial color=#008000 size=2></FONT><FONT face=Arial 
color=#008000 size=2></FONT><FONT face=Arial color=#008000 size=2></FONT><FONT 
face=Arial color=#008000 size=2></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial color=#008000 size=2><SPAN class=319184209-28112003>Is 
this a correct understanding of your posting?</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#008000 size=2><SPAN 
class=319184209-28112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial color=#008000 size=2><SPAN 
class=319184209-28112003>Regards,</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#008000 size=2><SPAN class=319184209-28112003>Paul 
DS.</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#008000 size=2><SPAN class=319184209-28112003>
<P><FONT face="Courier New" size=2>Paul D.Smith</FONT> <BR><FONT 
face="Courier New" size=2>Network Protocols Group </FONT><BR><FONT 
face="Courier New" size=2>Data Connection Ltd (DCL)</FONT> <BR><FONT 
face="Courier New" size=2>Tel: +44 20 8366 1177&nbsp; Email: 
paul.d.smith@dataconnection.com </FONT><BR><FONT face="Courier New" size=2>Fax: 
+44 20 8363 1039&nbsp; Web:&nbsp;&nbsp; <A href="http://www.dataconnection.com/" 
target=_blank>http://www.dataconnection.com</A></FONT> </SPAN></FONT><FONT 
face=Tahoma><BR><FONT size=2>-----Original Message-----<BR><B>From:</B> 
Christian Stredicke [mailto:stredicke@snom.de]<BR><B>Sent:</B> 27 November 2003 
17:45<BR><B>To:</B> SIP (E-mail)<BR><B>Subject:</B> [Sip] From and To-header 
confusion in draft-ietf-sip-replaces-04.txt<BR><BR></P></DIV></FONT></FONT>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
  <DIV class=Section1>
  <P class=MsoNormal><FONT face=Verdana size=2><SPAN lang=EN-US 
  style="FONT-SIZE: 10pt; FONT-FAMILY: Verdana">Hi!</SPAN></FONT></P>
  <P class=MsoNormal><FONT face=Verdana size=2><SPAN lang=EN-US 
  style="FONT-SIZE: 10pt; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>
  <P class=MsoNormal><FONT face=Verdana size=2><SPAN lang=EN-US 
  style="FONT-SIZE: 10pt; FONT-FAMILY: Verdana">I think there is some from and 
  to-header confusion in the Replaces header.</SPAN></FONT></P>
  <P class=MsoNormal><FONT face=Verdana size=2><SPAN lang=EN-US 
  style="FONT-SIZE: 10pt; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>
  <P class=MsoNormal><FONT face=Verdana size=2><SPAN lang=EN-US 
  style="FONT-SIZE: 10pt; FONT-FAMILY: Verdana">Depending on who initiates the 
  call, the to and from must be exchanged. Maybe it would be better to use the 
  term local and remote-tag instead or to allow mixing from and 
  to.</SPAN></FONT></P>
  <P class=MsoNormal><FONT face=Verdana size=2><SPAN lang=EN-US 
  style="FONT-SIZE: 10pt; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>
  <P class=MsoNormal><FONT face=Verdana size=2><SPAN lang=EN-US 
  style="FONT-SIZE: 10pt; FONT-FAMILY: Verdana">Christian</SPAN></FONT></P>
  <P class=MsoNormal><FONT face=Verdana size=2><SPAN lang=EN-GB 
  style="FONT-SIZE: 10pt; FONT-FAMILY: Verdana">--</SPAN></FONT></P>
  <P class=MsoNormal><FONT face=Verdana size=2><SPAN lang=EN-GB 
  style="FONT-SIZE: 10pt; FONT-FAMILY: Verdana">Christian 
  Stredicke</SPAN></FONT></P>
  <P class=MsoNormal><FONT face=Verdana size=2><SPAN lang=EN-GB 
  style="FONT-SIZE: 10pt; FONT-FAMILY: Verdana">tel:+49.30.39833.401 (ENUM &amp; 
  PSTN)</SPAN></FONT></P></DIV></BLOCKQUOTE></BODY></HTML>

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From: "Christian Stredicke" <stredicke@snom.de>
To: "'Paul D.Smith'" <Paul.D.Smith@dataconnection.com>
Cc: <sip@ietf.org>
Subject: AW: [Sip] From and To-header confusion in draft-ietf-sip-replaces-04.txt
Date: Fri, 28 Nov 2003 11:19:46 +0100
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Hi Paul,

=20

correct. The problem is that users get angry if the pickup does not work
just because of standards confusion.=20

=20

Our current workaround is to relax the requirement a little bit and
accept to- and from-tags exchanged. That works fine, however there is
some risk that this can lead to some false pickups. However, I don=92t
know a scenario where this can be an issue.

=20

Christian

=20

-----Urspr=FCngliche Nachricht-----
Von: Paul D.Smith [mailto:Paul.D.Smith@dataconnection.com]=20
Gesendet: Freitag, 28. November 2003 11:05
An: 'Christian Stredicke'
Cc: 'sip@ietf.org'
Betreff: RE: [Sip] From and To-header confusion in
draft-ietf-sip-replaces-04.txt

=20

Christian,

=20

If I understand you correctly, you are pointing out the following.

=20

If a target receives an INVITE with Replaces headers, it has to somehow
use the Replaces To/From fields to identify a dialog.  If this dialog
were originally started by the target then there may be a mapping of:

=20

To field of original INVITE =3D=3D> target's idea of remote tag

From field of original INVITE =3D=3D> target's idea of local tag.

=20

However if the target did not start the dialog, the mapping is reversed.

=20

To field of original INVITE =3D=3D> target's idea of local tag

From field of original INVITE =3D=3D> target's idea of remote tag.

=20

And your question is... how does the Replaces To/From get mapped to the
targets idea of local/remote tags?

=20

Unfortunately, this can't be resolved into a simple mapping.  A third
party REFER modifying an INVITE may not know who started the dialog and
so renaming the Replaces fields makes no difference since the referrer
still doesn't know which tag to place in which field.  The target has to
perform a lookup using both mappings - often one will find a dialog and
the other will not.

=20

There is a possible issue with "loopback" type dialogs where a UA, with
multiple personalities (i.e. a gateway) may have two dialogs, one in
each direction.  In this case, it may be impossible to map to a single
dialog, which is where the following sentence from page 5 of the draft
is headed...

=20

  "If the Replaces header field matches more than one dialog, the UA
MUST act as if no match is found."

=20

Is this a correct understanding of your posting?

=20

Regards,

Paul DS.

Paul D.Smith=20
Network Protocols Group=20
Data Connection Ltd (DCL)=20
Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com=20
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
<http://www.dataconnection.com/> =20
-----Original Message-----
From: Christian Stredicke [mailto:stredicke@snom.de]
Sent: 27 November 2003 17:45
To: SIP (E-mail)
Subject: [Sip] From and To-header confusion in
draft-ietf-sip-replaces-04.txt

Hi!

=20

I think there is some from and to-header confusion in the Replaces
header.

=20

Depending on who initiates the call, the to and from must be exchanged.
Maybe it would be better to use the term local and remote-tag instead or
to allow mixing from and to.

=20

Christian

--

Christian Stredicke

tel:+49.30.39833.401 (ENUM & PSTN)


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
style=3D'font-size:
10.0pt;font-family:Verdana;color:blue'>Hi Paul,</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
style=3D'font-size:
10.0pt;font-family:Verdana;color:blue'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>correct. The =
problem is
that users get angry if the pickup does not work just because of =
standards confusion.
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>&nbsp;</span></=
font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>Our current =
workaround
is to relax the requirement a little bit and accept to- and from-tags =
exchanged.
That works fine, however there is some risk that this can lead to some =
false
pickups. However, I don&#8217;t know a scenario where this can be an =
issue.</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>&nbsp;</span></=
font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>Christian</span=
></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DVerdana><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Verdana;color:blue'>&nbsp;</span></=
font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0cm 0cm =
0cm 4.0pt'>

<p class=3DMsoNormal><font size=3D2 face=3DTahoma><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Tahoma'>-----Urspr=FCngliche Nachricht-----<br>
<b><span style=3D'font-weight:bold'>Von:</span></b> Paul =
D.Sm</span></font><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>ith
[mailto:Paul.D.Smith@dataconnection.com] <br>
<b><span style=3D'font-weight:bold'>Gesendet:</span></b> Freitag, 28. =
November
2003 11:05<br>
<b><span style=3D'font-weight:bold'>An:</span></b> 'Christian =
Stredicke'<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> '</span></font><font =
size=3D2
 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>sip@ietf.org</span></font><=
font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>'<br>
<b><span style=3D'font-weight:bold'>Betreff:</span></b> RE: [Sip] From =
and
To-header confusion in draft-ietf-sip-replaces-04.txt</span></font></p>

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

<div>

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

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>If I understand you =
correctly,&nbsp;you
are pointing out the following.</span></font></p>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>If a target receives an INVITE =
with
Replaces headers, it has to somehow use the Replaces To/From fields to =
identify
a dialog.&nbsp; If this dialog were originally started by the target =
then there
may be a mapping of:</span></font></p>

</div>

<div>

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

</div>

<div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>To field of original INVITE =
=3D=3D&gt;
target's idea of remote tag</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>From field of original INVITE =
=3D=3D&gt;
target's idea of local tag.</span></font></p>

</div>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>However if the target did not =
start the
dialog, the mapping is reversed.</span></font></p>

</div>

<div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>To field of original INVITE =
=3D=3D&gt;
target's idea of local tag</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>From field of original INVITE =
=3D=3D&gt;
target's idea of remote tag.</span></font></p>

</div>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>And your question is... how does =
the
Replaces To/From get mapped to the targets idea of local/remote =
tags?</span></font></p>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>Unfortunately, this can't&nbsp;be
resolved into a simple mapping.&nbsp; A third party REFER modifying an =
INVITE
may not know who started the dialog and so renaming the Replaces fields =
makes
no difference since the referrer still doesn't know which tag to place =
in which
field.&nbsp; The target has to perform a lookup using both mappings - =
often one
will&nbsp;find a dialog and the other will not.</span></font></p>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>There is a possible issue with
&quot;loopback&quot; type dialogs where a UA, with multiple =
personalities (i.e.
a gateway) may have two dialogs, one in each direction.&nbsp; In this =
case, it
may be impossible to map to a single dialog, which is where the =
following
sentence from page 5 of the draft is headed...</span></font></p>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>&nbsp; &quot;If the Replaces =
header field
matches more than one dialog, the UA&nbsp;MUST act as if no match is
found.&quot;</span></font></p>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>Is this a correct understanding of =
your
posting?</span></font></p>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>Regards,</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:green'>Paul DS.</span></font></p>

</div>

<div>

<p style=3D'margin-bottom:12.0pt'><font size=3D2 color=3Dgreen =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:green'>Paul =
D.Smith</span></font><font
size=3D2 color=3Dgreen face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
color:green'> <br>
</span></font><font size=3D2 color=3Dgreen face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:green'>Network
Protocols Group </span></font><font size=3D2 color=3Dgreen =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:green'><br>
</span></font><font size=3D2 color=3Dgreen face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:green'>Data =
Connection
Ltd (DCL)</span></font><font size=3D2 color=3Dgreen face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:green'> <br>
</span></font><font size=3D2 color=3Dgreen face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:green'>Tel: =
+44 20 8366
1177&nbsp; Email: paul.d.smith@dataconnection.com </span></font><font =
size=3D2
color=3Dgreen face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
color:green'><br>
</span></font><font size=3D2 color=3Dgreen face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:green'>Fax: =
+44 20 8363
1039&nbsp; Web:&nbsp;&nbsp; <a href=3D"http://www.dataconnection.com/"
target=3D"_blank">http://www.dataconnection.com</a></span></font><font =
size=3D2
color=3Dgreen face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
color:green'> </span></font><font face=3DTahoma><span =
style=3D'font-family:Tahoma'><br>
</span></font><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma'>-----Original Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> Christian Stredicke
[mailto:stredicke@snom.de]<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 27 November 2003 =
17:45<br>
<b><span style=3D'font-weight:bold'>To:</span></b> SIP (E-mail)<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] From and =
To-header
confusion in draft-ietf-sip-replaces-04.txt</span></font></p>

</div>

<blockquote =
style=3D'margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt'>

<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Verdana'>Hi!</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Verdana'>I think there is some from and to-header =
confusion
in the Replaces header.</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Verdana'>Depending on who initiates the call, the to =
and
from must be exchanged. Maybe it would be better to use the term local =
and
remote-tag instead or to allow mixing from and to.</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Verdana'>Christian</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Verdana'>Christian Stredicke</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DVerdana><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Verdana'>tel:+49.30.39833.401 (ENUM &amp; =
PSTN)</span></font></p>

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Hi:
I have a question on session timer message flow.

 If a UA does not support session timer and if it receives=20

Require: timer
Session Expires: 3600=20

in the 200 OK to re-INVITE what is the correct way to process this =
response?=20

Should UAC Ack the 200 OK and then send a bye or ignore the Require and =
session Expires: headers and continue with rest of the processing?

Veda



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</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I have a question on session =
timer&nbsp;message=20
flow.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;If a UA does not support session =
timer and if=20
it receives </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Require: timer</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Session Expires: 3600 </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>in the 200 OK to re-INVITE what is the =
correct way=20
to process this response? </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Should UAC Ack the 200 OK and then send =
a bye or=20
ignore the Require and session Expires: headers and continue with rest =
of the=20
processing?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Veda</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

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From exim@www1.ietf.org  Sat Nov 29 07:14:42 2003
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From: Gokul Krishnan <gktvm2@yahoo.com>
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Subject: [Sip] Query on PRACK
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Hi All,
 
Does PRACK trigger another 2XX response ? If yes, does that mean UAC should send an ACK? Can any one please provide a message sequence flow?
 
  If a PRACK request is received by the UA core that does not
  match any unacknowledged reliable provisional response, the 
  UAS MUST respond to the PRACK with a 481 response.  If the
  PRACK does match an unacknowledged reliable provisional
  response, it MUST be responded to with a 2xx response.  
  (RFC 3262, Page 5)


Thanks and Regards
Gokul



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<DIV>
<DIV><FONT face=Verdana color=#800000 size=2><SPAN class=320184811-29112003>Hi All,</SPAN></FONT></DIV>
<DIV><FONT face=Verdana color=#800000 size=2></FONT>&nbsp;</DIV>
<DIV><FONT face=Verdana><FONT color=#800000 size=2>Does PRACK trigger another 2XX response ?&nbsp;<SPAN class=320184811-29112003>If yes, does that mean&nbsp;</SPAN><SPAN class=320184811-29112003>UAC should send an</SPAN>&nbsp;ACK? Can&nbsp;<SPAN class=320184811-29112003>any one</SPAN>&nbsp;please provide a message sequence&nbsp;<SPAN class=320184811-29112003>flow</SPAN>?</FONT></FONT></DIV>
<DIV><FONT face=Verdana color=#800000 size=2></FONT>&nbsp;</DIV>
<DIV><FONT face=Verdana color=#800000><FONT size=2><FONT face="Courier New" color=#000000>&nbsp; If a PRACK request is received by the UA core that does not</FONT></FONT></FONT></DIV>
<DIV><FONT face=Verdana color=#800000><FONT size=2><FONT face="Courier New" color=#000000>&nbsp;&nbsp;match any&nbsp;unacknowledged reliable provisional response, the </FONT></FONT></FONT></DIV>
<DIV><FONT face=Verdana color=#800000><FONT size=2><FONT face="Courier New" color=#000000>&nbsp; UAS MUST respond to&nbsp;the PRACK with a 481 response.&nbsp; If the</FONT></FONT></FONT></DIV>
<DIV><FONT face=Verdana color=#800000><FONT size=2><FONT face="Courier New" color=#000000>&nbsp; PRACK does match an&nbsp;unacknowledged reliable provisional</FONT></FONT></FONT></DIV>
<DIV><FONT face=Verdana color=#800000><FONT size=2><FONT face="Courier New" color=#000000>&nbsp; response, it MUST be responded to&nbsp;with a 2xx response.</FONT>&nbsp;<SPAN class=320184811-29112003><FONT face="Courier New" color=#000000> </FONT></SPAN></FONT></FONT></DIV>
<DIV><FONT face=Verdana color=#800000><FONT size=2><SPAN class=320184811-29112003><FONT face="Courier New" color=#000000>&nbsp; (<STRONG>RFC 3262, Page 5</STRONG>)</FONT></SPAN></FONT></FONT></DIV>
<DIV>
<DIV><SPAN class=320184811-29112003><FONT face=Verdana color=#800000 size=2></FONT></SPAN></DIV><SPAN class=320184811-29112003><FONT face=Verdana color=#800000 size=2></FONT></SPAN></DIV>
<DIV><SPAN class=320184811-29112003><FONT face=Verdana color=#800000 size=2>Thanks and Regards</FONT></SPAN></DIV>
<DIV><SPAN class=320184811-29112003><FONT face=Verdana color=#800000 size=2>Gokul</FONT></SPAN></DIV></DIV><p><hr SIZE=1>
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Date: Sat, 29 Nov 2003 09:18:51 -0800
Subject: Re: AW: [Sip] From and To-header confusion in
	draft-ietf-sip-replaces-04.txt
From: Cullen Jennings <fluffy@cisco.com>
To: Christian Stredicke <stredicke@snom.de>,
        "'Paul D.Smith'" <Paul.D.Smith@dataconnection.com>
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Sounds like we need to fix the draft.

On 11/28/03 2:19 AM, "Christian Stredicke" <stredicke@snom.de> wrote:

> Hi Paul,
> =20
> correct. The problem is that users get angry if the pickup does not work =
just
> because of standards confusion.
> =20
> Our current workaround is to relax the requirement a little bit and accep=
t to-
> and from-tags exchanged. That works fine, however there is some risk that=
 this
> can lead to some false pickups. However, I don=B9t know a scenario where th=
is
> can be an issue.
> =20
> Christian
> =20
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: Paul D.Smith [mailto:Paul.D.Smith@dataconnection.com]
> Gesendet: Freitag, 28. November 2003 11:05
> An: 'Christian Stredicke'
> Cc: 'sip@ietf.org'
> Betreff: RE: [Sip] From and To-header confusion in
> draft-ietf-sip-replaces-04.txt
> =20
>=20
> Christian,
>=20
> =20
>=20
> If I understand you correctly, you are pointing out the following.
>=20
> =20
>=20
> If a target receives an INVITE with Replaces headers, it has to somehow u=
se
> the Replaces To/From fields to identify a dialog.  If this dialog were
> originally started by the target then there may be a mapping of:
>=20
> =20
>=20
> To field of original INVITE =3D=3D> target's idea of remote tag
>=20
> From field of original INVITE =3D=3D> target's idea of local tag.
>=20
> =20
>=20
> However if the target did not start the dialog, the mapping is reversed.
>=20
> =20
>=20
> To field of original INVITE =3D=3D> target's idea of local tag
>=20
> From field of original INVITE =3D=3D> target's idea of remote tag.
>=20
> =20
>=20
> And your question is... how does the Replaces To/From get mapped to the
> targets idea of local/remote tags?
>=20
> =20
>=20
> Unfortunately, this can't be resolved into a simple mapping.  A third par=
ty
> REFER modifying an INVITE may not know who started the dialog and so rena=
ming
> the Replaces fields makes no difference since the referrer still doesn't =
know
> which tag to place in which field.  The target has to perform a lookup us=
ing
> both mappings - often one will find a dialog and the other will not.
>=20
> =20
>=20
> There is a possible issue with "loopback" type dialogs where a UA, with
> multiple personalities (i.e. a gateway) may have two dialogs, one in each
> direction.  In this case, it may be impossible to map to a single dialog,
> which is where the following sentence from page 5 of the draft is headed.=
..
>=20
> =20
>=20
>   "If the Replaces header field matches more than one dialog, the UA MUST=
 act
> as if no match is found."
>=20
> =20
>=20
> Is this a correct understanding of your posting?
>=20
> =20
>=20
> Regards,
>=20
> Paul DS.
>=20
> Paul D.Smith=20
> Network Protocols Group
> Data Connection Ltd (DCL)
> Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com
> Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
> <http://www.dataconnection.com/>
> -----Original Message-----
> From: Christian Stredicke [mailto:stredicke@snom.de]
> Sent: 27 November 2003 17:45
> To: SIP (E-mail)
> Subject: [Sip] From and To-header confusion in draft-ietf-sip-replaces-04=
.txt
>>=20
>> Hi!
>> =20
>> I think there is some from and to-header confusion in the Replaces heade=
r.
>> =20
>> Depending on who initiates the call, the to and from must be exchanged. =
Maybe
>> it would be better to use the term local and remote-tag instead or to al=
low
>> mixing from and to.
>> =20
>> Christian
>> --
>> Christian Stredicke
>> tel:+49.30.39833.401 (ENUM & PSTN)
>=20



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<HTML>
<HEAD>
<TITLE>Re: AW: [Sip] From and To-header confusion in draft-ietf-sip-replace=
s-04.txt</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><BR>
Sounds like we need to fix the draft.<BR>
<BR>
On 11/28/03 2:19 AM, &quot;Christian Stredicke&quot; &lt;stredicke@snom.de&=
gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0p=
x'><FONT COLOR=3D"#0000FF">Hi Paul,<BR>
&nbsp;<BR>
correct. The problem is that users get angry if the pickup does not work ju=
st because of standards confusion. <BR>
&nbsp;<BR>
Our current workaround is to relax the requirement a little bit and accept =
to- and from-tags exchanged. That works fine, however there is some risk tha=
t this can lead to some false pickups. However, I don&#8217;t know a scenari=
o where this can be an issue.<BR>
&nbsp;<BR>
Christian<BR>
&nbsp;<BR>
</FONT><BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Tahoma">-----Urspr=
&uuml;ngliche Nachricht-----<BR>
<B>Von:</B> Paul D.Smith [<a href=3D"mailto:Paul.D.Smith@dataconnection.com]"=
>mailto:Paul.D.Smith@dataconnection.com]</a> <BR>
<B>Gesendet:</B> Freitag, 28. November 2003 11:05<BR>
<B>An:</B> 'Christian Stredicke'<BR>
<B>Cc:</B> 'sip@ietf.org'<BR>
<B>Betreff:</B> RE: [Sip] From and To-header confusion in draft-ietf-sip-re=
places-04.txt<BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#008000"><FONT FA=
CE=3D"Arial">Christian,<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#008000"><FONT FA=
CE=3D"Arial">If I understand you correctly, you are pointing out the following=
.<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#008000"><FONT FA=
CE=3D"Arial">If a target receives an INVITE with Replaces headers, it has to s=
omehow use the Replaces To/From fields to identify a dialog. &nbsp;If this d=
ialog were originally started by the target then there may be a mapping of:<=
BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#008000"><FONT FA=
CE=3D"Arial">To field of original INVITE =3D=3D&gt; target's idea of remote tag<BR=
>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT><FONT COLOR=3D"#008000"><FONT FACE=3D"Arial">From field of original INVI=
TE =3D=3D&gt; target's idea of local tag.<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#008000"><FONT FA=
CE=3D"Arial">However if the target did not start the dialog, the mapping is re=
versed.<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT><FONT COLOR=3D"#008000"><FONT FACE=3D"Arial"> <BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT><FONT COLOR=3D"#008000"><FONT FACE=3D"Arial">To field of original INVITE=
 =3D=3D&gt; target's idea of local tag<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT><FONT COLOR=3D"#008000"><FONT FACE=3D"Arial">From field of original INVI=
TE =3D=3D&gt; target's idea of remote tag.<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#008000"><FONT FA=
CE=3D"Arial">And your question is... how does the Replaces To/From get mapped =
to the targets idea of local/remote tags?<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#008000"><FONT FA=
CE=3D"Arial">Unfortunately, this can't be resolved into a simple mapping. &nbs=
p;A third party REFER modifying an INVITE may not know who started the dialo=
g and so renaming the Replaces fields makes no difference since the referrer=
 still doesn't know which tag to place in which field. &nbsp;The target has =
to perform a lookup using both mappings - often one will find a dialog and t=
he other will not.<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#008000"><FONT FA=
CE=3D"Arial">There is a possible issue with &quot;loopback&quot; type dialogs =
where a UA, with multiple personalities (i.e. a gateway) may have two dialog=
s, one in each direction. &nbsp;In this case, it may be impossible to map to=
 a single dialog, which is where the following sentence from page 5 of the d=
raft is headed...<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#008000"><FONT FA=
CE=3D"Arial"> &nbsp;&quot;If the Replaces header field matches more than one d=
ialog, the UA MUST act as if no match is found.&quot;<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#008000"><FONT FA=
CE=3D"Arial">Is this a correct understanding of your posting?<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#008000"><FONT FA=
CE=3D"Arial">Regards,<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT><FONT COLOR=3D"#008000"><FONT FACE=3D"Arial">Paul DS.<BR>
</FONT></FONT><FONT FACE=3D"Verdana"><BR>
</FONT><FONT COLOR=3D"#008000"><FONT FACE=3D"Courier New">Paul D.Smith</FONT><F=
ONT FACE=3D"Arial"> <BR>
</FONT><FONT FACE=3D"Courier New">Network Protocols Group <BR>
Data Connection Ltd (DCL)</FONT><FONT FACE=3D"Arial"> <BR>
</FONT><FONT FACE=3D"Courier New">Tel: +44 20 8366 1177 &nbsp;Email: paul.d.s=
mith@dataconnection.com <BR>
Fax: +44 20 8363 1039 &nbsp;Web: &nbsp;&nbsp;<a href=3D"http://www.dataconnec=
tion.com">http://www.dataconnection.com</a> <a href=3D"http://www.dataconnecti=
on.com/">&lt;http://www.dataconnection.com/&gt;</a> </FONT><FONT FACE=3D"Arial=
"> <BR>
</FONT></FONT><FONT FACE=3D"Tahoma">-----Original Message-----<BR>
<B>From:</B> Christian Stredicke [<a href=3D"mailto:stredicke@snom.de]">mailt=
o:stredicke@snom.de]</a><BR>
<B>Sent:</B> 27 November 2003 17:45<BR>
<B>To:</B> SIP (E-mail)<BR>
<B>Subject:</B> [Sip] From and To-header confusion in draft-ietf-sip-replac=
es-04.txt<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Verdan=
a"><BR>
Hi!<BR>
</FONT></SPAN><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'>I =
think there is some from and to-header confusion in the Replaces header.<BR>
</SPAN></FONT><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'>De=
pending on who initiates the call, the to and from must be exchanged. Maybe =
it would be better to use the term local and remote-tag instead or to allow =
mixing from and to.<BR>
</SPAN></FONT><FONT SIZE=3D"4"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font=
-size:18.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'>Ch=
ristian<BR>
--<BR>
Christian Stredicke<BR>
tel:+49.30.39833.401 (ENUM &amp; PSTN)<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0=
px'><BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0=
px'><BR>
</SPAN></FONT>
</BODY>
</HTML>


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Date: Sat, 29 Nov 2003 09:20:57 -0800
Subject: Re: [Sip] draft-ietf-iptel-trunk-group-00.txt
From: Cullen Jennings <fluffy@cisco.com>
To: Medhavi Bhatia <mbhatia@nextone.com>, <iptel-admin@ietf.org>
CC: <sip@ietf.org>
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It is being revised right now =AD it is an IPTEL draft =AD please use that emai=
l
list. If you reply to this, please take the sip mailing list off the email
thread. Thanks, Cullen
=20


On 11/26/03 8:21 PM, "Medhavi Bhatia" <mbhatia@nextone.com> wrote:

> Folks,
> =20
> Is this draft being revived ? We had some questions/comments on the synta=
x
> specified in it and were wondering if this was being worked on or dropped=
.
> =20
> Medhavi.
>=20



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<HTML>
<HEAD>
<TITLE>Re: [Sip] draft-ietf-iptel-trunk-group-00.txt</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><BR>
It is being revised right now &#8211; it is an IPTEL draft &#8211; please u=
se that email list. If you reply to this, please take the sip mailing list o=
ff the email thread. Thanks, Cullen<BR>
&nbsp;<BR>
<BR>
<BR>
On 11/26/03 8:21 PM, &quot;Medhavi Bhatia&quot; &lt;mbhatia@nextone.com&gt;=
 wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Courie=
r">Folks,<BR>
</FONT><FONT FACE=3D"Verdana"> <BR>
</FONT><FONT FACE=3D"Courier">Is this draft being revived ? We had some quest=
ions/comments on the syntax specified in it and were wondering if this was b=
eing worked on or dropped.<BR>
</FONT><FONT FACE=3D"Verdana"> <BR>
</FONT><FONT FACE=3D"Courier">Medhavi.<BR>
</FONT><FONT FACE=3D"Verdana"><BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Verda=
na"><BR>
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Date: Sat, 29 Nov 2003 20:25:38 +0100
From: Rene Bartsch <ml@bartschnet.de>
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Subject: [Sip] ENUM-support in SIP-protocol?
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Hi,

first I'm new to the list, so hello to all! :-)

My question is if there is any draft of plan to implement ENUM-lookups 
into the SIP-protocol?

Rene Bartsch


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