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From: "Joanne McMillen" <joanne@avaya.com>
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Subject: [Sip] RFC 3261 and media
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I seem to recall discussions a long time ago regarding SIP sessions without
media.
Currently, RFC 3261 is very vague on what these procedures would be. Given
the
offer/answer model, how would a UA receiving an INVITE without an offer know
how to
distinguish between a "delayed-media" case and a "no-media" case? Is there a
specific
session content type  that should be used for this? I cannot find it
documented
anywhere.

Please help.

Regards,
Joanne

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
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<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2>I seem to recall discussions a long time ago =
regarding=20
SIP&nbsp;sessions without media</FONT><FONT size=3D2>.</FONT></DIV>
<DIV><FONT size=3D2>Currently, RFC 3261 is very vague on what these =
procedures=20
would be. Given the</FONT></DIV>
<DIV><FONT size=3D2>offer/answer model, how would a UA receiving an =
INVITE without=20
an offer know how to</FONT></DIV>
<DIV><FONT size=3D2>distinguish between a "delayed-media" case and a =
"no-media"=20
case? Is there a specific </FONT></DIV>
<DIV><FONT size=3D2>session content type &nbsp;that should be used =
</FONT><FONT=20
size=3D2>for this? I cannot find&nbsp;it documented </FONT></DIV>
<DIV><FONT size=3D2>anywhere. </FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Please help.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Regards,</FONT></DIV>
<DIV><FONT size=3D2>Joanne</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

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Joanne McMillen wrote:
> I seem to recall discussions a long time ago regarding SIP sessions 
> without media.
> Currently, RFC 3261 is very vague on what these procedures would be. 
> Given the
> offer/answer model, how would a UA receiving an INVITE without an offer 
> know how to
> distinguish between a "delayed-media" case and a "no-media" case? Is 
> there a specific
> session content type  that should be used for this? I cannot find it 
> documented
> anywhere.

If one of the messages that is permitted to carry an offer or answer 
contains a body of Content-Type application/sdp, and content-disposition 
of "session", then it is an offer or answer. To conform, the body then 
must actually contain content conforming to the syntax of sdp. If that 
sdp has no m-lines, (or only m-lines with a port of zero) then it is a 
media-less offer or answer.

OTOH, a delayed-media invite contains no sdp at all.

Individual endpoints have to make their own decisions about whether to 
accept no-media invites, and if so how long to keep them around.

	Paul


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From exim@www1.ietf.org  Tue Feb  3 01:49:44 2004
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: "Niemi Aki (Nokia-M/Helsinki)" <aki.niemi@nokia.com>
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Subject: Re: [Sip] querying the current event state of a publication
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Firstly, I'm inclined to back away from my prior suggesting of using 
URIs instead of etags, primarily because I think we are just too far 
along here and "if it aint broke, don't fix it". We can probably get the 
missing piece of functionality - finding out who else is publishing - in 
ways that dont change what we have documented.

Your suggestion below is reasonable towards that end, but makes me 
nervous that it introduces a behavior that is *like* registration, but 
isnt the same (no  contacts in the publish request, for example).

Another possible idea is to create a new event package which is the 
"esc" event package. When you subscribe to a URI for that package, you 
get notified of all published documents made against that URI. The 
current state of the resource for that package is a list of the 
currently active documents for that URI. By doing this as an event 
package, we can leave PUBLISH alone, and add this functionality in the 
future, as a separate item.

-Jonathan R.

Niemi Aki (Nokia-M/Helsinki) wrote:

> Hi,
> 
> Inline.
> 
> ext Jonathan Rosenberg wrote:
> 
>> In draft-ietf-sip-publish-02, section 5.7 proposes a way to query a 
>> specific published event state. It recommends subscribing to the AOR 
>> to learn it. However, it gives this caveat:
>>
>>  Note that a subscription to the event package will likely deliver
>>       results of the event composition process of the state agent, which
>>       may be a subset or a superset of the current published event
>>       state.
>>
>> which is another way of saying, that subscribing to the aor doesnt 
>> really allow you to query for a specific published event state.
>>
>> If we want this functionality, I have an idea for an alternative 
>> approach that works much better, triggered by some conversations I had 
>> with Ben C. recently.
>>
>> In the response to a PUBLISH request, the ESC returns a URI that 
>> identifies that particular event state publication (effectively, a URI 
>> that is bound to the AOR+event+etag of a publication). This URI is 
>> basically a GRUU. If a user wishes to learn the state of this 
>> particular event state, they merely subscribe to that URI. Indeed, a 
>> particularly interestinig way to do this is to define sIP etags as 
>> URIs, rather than reusing the definition from http. I think that 
>> having an etag as a URI is far more elegant, as it would allow for 
>> things like this function in a natural way. The alternative would be 
>> to possibly use the Contact header field, as the meaning is the same 
>> as in a 2xx to INVITE - identifying the specific *instance* to which 
>> you were connected when you contacted the AOR. This makes sense if you 
>> think of the event state publication URI as a GRUU.
> 
> 
> But a PUA has to know the state it just published, simply because it was 
> the source of that state in the first place. What it can't learn is the 
> state of the other active PUAs.
> 
> And this is actually something currently missing from PUBLISH. A given 
> PUA can't find out about publications other than thiose it itself has 
> made. As a consequence, it can't for example, override the publications 
> of a zombie PUA (a use case we have discussed in the past at length).
> 
> In order to enable a PUA to find out about the *other* publications to a 
> given resource, the ESC would have to return not only a pointer to this 
> PUAs publication, but to all active publications for that AoR. This is 
> very similar to a registrar returning all active address bindings of an 
> AoR.
> 
>> Indeed, one might even then PUBLISH to that URI, and it would have the 
>> exact result you would expect - that particular event state is 
>> updated. In such a case the server woould NOT return an etag or 
>> another URI in the response.
> 
> 
> When you add the feature I describe above, etags would still be needed 
> since it would allow two PUAs to publish to the same instance of state. 
> Etags would be needed for versioning in that case.
> 
> <snip />
> 
>> This would mean we wouldnt need the SIP-If-match.
>>
>> Another alternative is to say that we are simply too far along in 
>> PUBLISH, and the time has passed for major changes as this would be.
> 
> 
> I do think we are quite far along in PUBLISH for adding this sort of 
> functionality. However, I see some low-hanging fruit here, and to be 
> honest, it has always bothered me a bit that there is no way for a PUA 
> to know about the state other PUAs are publishing.
> 
> There is a requirement that calls for basically exactly this in 
> draft-ietf-simple-publish-reqs-00 and it states:
> 
>    14.  PUAs must have a capability that allows them to query for the
>         identifiers of all of the segments of presence information that
>         have currently been published for a presentity (provided that
>         the PUA is authorized to receive this information).
> 
> Using the register-like behavior for returning the set of active state 
> in the 2xx response to PUBLISH + having this set be a set of GRUUs would 
> fulfill the above. The PUA would be able to query the state (using 
> SUBSCRIBE), but also override it (using PUBLISH). This is also a 
> requireemnt in the reqs draft:
> 
>   15.  It must be possible for the publication of presence information
>         for a particular segment to overwrite existing information for
>         that segment, even if the existing information had been
>         published by a different PUA. ...
> 
> So I would be willing to add this functionality simply because it would 
> make the solution more complete. As far as adding complexity, sure, but 
> an ESC whose policy doesn't allow cross PUA state could always just 
> refuse to send the publication contacts in the PUBLISH response.
> 
> Does this sound reasonable?
> 
> Cheers,
> Aki
> 
> P.S. I added an example message flow to illustrate the above ramblings...
> 
> ---
> 
> PUA publishes its (initial) state:
> 
>       PUBLISH sip:presentity@example.com SIP/2.0
>       To: <sip:presentity@example.com>
>       From: <sip:presentity@example.com>;tag=54321mm
>       Expires: 3600
>       Event: presence
>       ...
> 
> ESC returns with an etag for that state, plus all active publications in 
> the form of URIs:
> 
>       SIP/2.0 200 OK
>       To: <sip:presentity@example.com>;tag=effe22aa
>       From: <sip:presentity@example.com>;tag=54321mm
>       SIP-ETag: qwi982ks
>       Expires: 3600
>       Contact: sip:kaskjsdh@pa.example.com;etag=qwi982ks;expires=3600
>       Contact: sip:jhsk23jj@pa.example.com;etag=hh3j4kll;expires=3229
>       Contact: sip:hhfkjhjk@pa.example.com;etag=3ihy4hhk;expires=1200
> 
> 
>> Thoughts?
>>
>> -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: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and target
 refresh request
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Christian Jansson wrote:

> Perhaps a little late, and hopefully not off topic. Comments inline.

Not too late, not off topic.

> 
> 
>>Jonathan R. wrote
>>This, however, leads me into thinking a bit on the lifecycle
>>management for GRUUs. Currently, GRUU is bound to a single 
>>registration, where the registration is defined by the Contact URI. If
> 
> 
>>the UA should "move", in the sense that it acquires a new IP address, 
>>this will be a new Contact URI, and therefore, a new GRUU. It would be
>>nice if, instead, the GRUU were truly bound to the UA instance, and 
>>that even if the UA instance changed IP addresses, the GRUU could 
>>remain unchanged. In such a case, if the UA moves, there is actually 
>>no need to issue any target refresh requests. The UA would simply 
>>re-register, update the binding of its GRUU to its new IP 
>>address, and 
>>existing calls would be correctly routed.
> 
> 
> But if the ip actually changes, it would be necessary to send an updated
> SDP with the new ip in the c= line.

Yes, that is true. If media relays are used (i.e., TURN), it might be 
possible to avoid even this, but lets leave that aside for now.

> So your idea won't save you from
> having to send a SIP request to the other end every time the ip changes
> in an INVITE session. 

Right. However, it does solve the problem that, previously, if both 
endpoints moved at the same time, the call would fail - neither side 
could signal the other.


> For a non-INVITE session though the idea would
> save a lot of request from being sent if the session only contains
> signaling.

Right, in particular, a long lived presence subscription, for example.

-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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Date: Tue, 03 Feb 2004 12:37:10 +0530
From: Natesan Kannan <nkannan@huawei.com>
Subject: Re: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and target
 refresh request
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Christian Jansson <christian.jansson@hotsip.com>
Cc: Paul Kyzivat <pkyzivat@cisco.com>, Jussi.Turunen@nokia.com,
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I will attempt a wild swing too...
Though we are talking of terminal mobility, it seems to me that this
mobility is restricted because of the fixed routeset in the dialog.
I can see that GRUU mapping to instance id, and the fact that GRUU can be
globally routed, paves a way for an un-restricted terminal mobility, just in
case nobody record-routed.
Was wondering if this could be of any use...
-Kannan
----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Christian Jansson" <christian.jansson@hotsip.com>
Cc: "Paul Kyzivat" <pkyzivat@cisco.com>; <Jussi.Turunen@nokia.com>;
<nataraju.alilaghatta@wipro.com>; <sanjsinh@cisco.com>; <sip@ietf.org>
Sent: Tuesday, February 03, 2004 12:24 PM
Subject: Re: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and
target refresh request


>
>
> Christian Jansson wrote:
>
> > Perhaps a little late, and hopefully not off topic. Comments inline.
>
> Not too late, not off topic.
>
> >
> >
> >>Jonathan R. wrote
> >>This, however, leads me into thinking a bit on the lifecycle
> >>management for GRUUs. Currently, GRUU is bound to a single
> >>registration, where the registration is defined by the Contact URI. If
> >
> >
> >>the UA should "move", in the sense that it acquires a new IP address,
> >>this will be a new Contact URI, and therefore, a new GRUU. It would be
> >>nice if, instead, the GRUU were truly bound to the UA instance, and
> >>that even if the UA instance changed IP addresses, the GRUU could
> >>remain unchanged. In such a case, if the UA moves, there is actually
> >>no need to issue any target refresh requests. The UA would simply
> >>re-register, update the binding of its GRUU to its new IP
> >>address, and
> >>existing calls would be correctly routed.
> >
> >
> > But if the ip actually changes, it would be necessary to send an updated
> > SDP with the new ip in the c= line.
>
> Yes, that is true. If media relays are used (i.e., TURN), it might be
> possible to avoid even this, but lets leave that aside for now.
>
> > So your idea won't save you from
> > having to send a SIP request to the other end every time the ip changes
> > in an INVITE session.
>
> Right. However, it does solve the problem that, previously, if both
> endpoints moved at the same time, the call would fail - neither side
> could signal the other.
>
>
> > For a non-INVITE session though the idea would
> > save a lot of request from being sent if the session only contains
> > signaling.
>
> Right, in particular, a long lived presence subscription, for example.
>
> -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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This list is for NEW development of the core SIP Protocol
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Hi Jonathan,

Thanks a lot for the excellent comments. Most of them I will directly =
fold into -03. Few clarifications below.

Cheers,
Aki

Jonathan Rosenberg wrote:

<snip />

 > * section 3, paragraph 3:
 >=20
 > > That is, the steady-state of
 > >    this event package in the absence of, or in addition=20
 > to, soft state
 > >    provided through the PUBLISH mechanism
 >=20
 > this makes it sound like steady state is a function of the event=20
 > packae only. the hard state is a function of the resource uRI and=20
 > event package. Indeed, one may think of an event package as merely=20
 > sub-resource within the full resource. I wonder if it wouldn't be=20
 > useful to come up with a term for this - for example, "scoped=20
 > resource" is a combination of a URI and event package, and=20
 > identifies=20
 > a particular piece of event state.

I don't quite follow,  but I agree the quoted text is rather vague. To =
me the hard state, or "preconfigured" state is just another source of =
event state to the composer. The mechanism to set that state might be =
other than PUBLISH.=20

<snip />

 > * section 5.1:
 >=20
 > > As with any other SIP message, the PUBLISH mechanism MAY use the
 > >    content indirection mechanism defined in [10].
 >=20
 > this statement makes [10] a normative reference, I believe. It is=20
 > listed as informative in the references section.

I'm not sure I agree. I've understood the RFC editor's policy so that an =
informative reference is not required for implementing the RFC. It's an =
interesting question as to how this relates to RFC 2119 terminology. But =
I think a MAY makes the feature optional so that it fits the informative =
reference definition.

<snip />

 > * section 5.3 is out of place. Section 5 is about generating=20
 > requests.=20
 > the subsections for the most part deal with the specific 4 sub-cases=20
 > you identified in 5.1. However, 5.3 talks about setting expirations,=20
 > which is something that is actually done for the cases in 5.2, 5.4,=20
 > 5.5 and 5.6. I would suggest that eaech of those four=20
 > sections rather=20
 > describe how the Expires header field is constructed for each case.=20
 > That is consistent with having a column in table 1 on=20
 > expires. Indeed,=20
 > each section should also talk about bodies and etag. Along those=20
 > lines, section 5.2 does not actually say that a body has to be=20
 > present; it shouold be stated given the importance of this fact from=20
 > table 1.

I think the reason for this is that I envisioned a PUBLISH carrying =
state in some other place than the body. However, I don't know if such a =
case would ever exist, and other people have also commented on this. I =
think it is safe to assume that the state is always carried in the body, =
as opposed to "typically".

I'll change the text in 5.2 accordingly.
=20
<snip />

 > * section 5.7 is out of place, since it has nothing to do with=20
 > constructing publish at all. indeed, since the note basically says=20
 > that this doesnt work, i would suggest removing the section entirely.

I think I agree with you. The separate email thread about querying the =
state of a particular publication ought to resolve this issue.=20

 > It occurs to me that its a huge shame that etags are not=20
 > URIs. If they=20
 > weree, we would have a rerally neat way to find out the=20
 > specific state=20
 > of a particular publication - you woould subscribe to that etag. The=20
 > next best thing might be to define a header field in the PUBLISH=20
 > response that identifies this particular published instance. Hmm,=20
 > Contact might even be a sensible choice there... worth discussing.=20
 > I'll send a separate email on the topic.

I responded to this separately.=20
=20
<snip />
=20
 > * section 6:
 >=20
 > >  PUBLISH requests MUST be processed in the order that they are
 > >    received.=20
 >=20
 > isnt this only true for requuests with the same r-uri?

Correct. I will add some text to this effect.
=20
<snip />

 > * the steps in section 6 omit handling for the case where=20
 > there was no=20
 > SIP-If-Match header field. step 6 says to check for it, and all the=20
 > following steps assume it was there. I would suggest adding a bullet=20
 > to step 6 which says, if there was no SIP-If-Match header field, the=20
 > ESC creates a new etag, and associates it with as-yet empty event=20
 > state. Processing continues in the steps below. This way, you always=20
 > exit step 6 with an etag.

The attempt was to make etag generation part of step 9. This way, an =
entity-tag is generated for each successful publication and returned in =
2xx, irrespective of whether the publication was an initial publication, =
a refresh, or a modification. Indeed, a 2xx to a removal could be =
treated the same, although such an entity-tag is useless at that point.

So I think it is ok to exit step 6 without an entity tag, as long as =
step 9 is clear that entity-tags are generated for all successful =
publications.

 >=20
 > * step 8 of section 6:
 >=20
 > >  The ESC processes the published event state, typically contained
 > >        in the body of the PUBLISH request. If the request=20
 > contains no
 > >        body (when it should contain one),=20
 >=20
 > i think we need to list the conditions for when it should=20
 > contain one.=20
 > Indeed, since you are always allowed to publish without a body in=20
 > order to refresh or delete, under what conditions is it illegal for=20
 > the body to not be there?

I don't think there are any. I will reword and remove that part.

<snip />

 > * step 9:
 >=20
 > > The response MUST also contain a SIP-ETag header field for
 > >        which the ESC MUST generate and store a locally=20
 > unique entity-tag
 > >        for identifying the publication.
 >=20
 > You should clarify that it generates a new one in each response, and=20
 > does so for both initial and refreshes. Also, clarify that this etag=20
 > replaces the previous, such that the previous etag is no=20
 > longer valid=20
 > for identifying the publication.

Good point.
=20
<snip />

 > * sectiton 11.1:
 >=20
 > > Authentication issues are discussed in SIP [2]. The exact=20
 > methods for
 > >    creation and manipulation of the ESC authorization policies are
 > >    outside the scope of this document.
 >=20
 > you will get dinged on this by Bellovin, guaranteed. He wants to see=20
 > statements about baseline mandtory to implement mechahnisms.=20
 > Its fine=20
 > to say that this mechanism is digest, as mandated by rfc3261.
 >=20
 > >  To prevent replay attacks, implementations SHOULD require
 > >    authentication with anti-replay protection.=20
 > Authentication issues are
 > >    discussed in SIP [2].
 >=20
 > as above.... does this mean digest with next-nonce or are=20
 > you talking=20
 > sips?
 >=20
 > and section 11.4:
 >=20
 > > To prevent such attacks, implementations SHOULD, at a minimum,
 > >    provide integrity protection across the To, From, Event,
 > >    SIP-If-Match, Route, and Expires headers and the bodies=20
 > of PUBLISH
 > >    requests.
 >=20
 > as above, he will ask for a mechanism. This requiurement makes the=20
 > choice sips.
 >=20
 > Same in 11.5.
=20
I see what you are saying. But is it really so that each new method that =
gets added to SIP needs to explicitly mandate certain security =
mechanisms? At least going beyond the security considerations in RFC =
3261 seems unnecessary, since the properties of PUBLISH are very similar =
to, say, REGISTER.=20

I was hoping we could get by with discussing PUBLISH specific threats, =
and merely referencing relevant parts of RFC3261 when it comes to =
mitigating those threats.

Cheers,
Aki



<snip />

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From: "Suha Cubukcuoglu" <suhac@nplusone.net>
To: <sip@ietf.org>
Date: Tue, 3 Feb 2004 11:20:19 -0000
Organization: edNET
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Subject: [Sip] Music On Hold
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Hi,

During an attended call transfer attempt, the transferred party
(A-Party) is put on 'hold' by the transferor (B-Party) - i.e. no media
stream flows from B to A and vice versa. While the A-Party is on hold,
the B-Party can INVITE a third party to play 'Music on Hold' tune to the
A-Party until the transfer attempt succeeds. We think that there are
complications in this approach. Firstly, the SIP Proxy must now where to
route the new INVITE generated by the B-Party. If the destination URI is
'moh@someserver.com', the SIP Proxy must either do a DNS query or use
its IP dial-plans to know where to route the call. Secondly, the B-Party
might eventually end-up receiving the music tune whereas it was
initially intended for the A-Party - we've had problems in the past
because of this. This is all dependent on how the third party MOH server
handles/established/terminates requests. 

We think that third party invitation is a bad idea, and it would rather
be more suitable to have a dedicated server (IVR unit perhaps) which the
SIP Proxy Server can transfer calls to/from upon request for a MOH
service. Could you tell us your opinions about this please?

Thanks.

Regards,    

Suha Cubukcuoglu
nplusone
t: +44 131 514 2000
d: +44 131 514 4037


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From exim@www1.ietf.org  Tue Feb  3 08:47:03 2004
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Cc: "'aki.niemi@nokia.com'" <aki.niemi@nokia.com>,
        "'jdrosen@dynamicsoft.com'" <jdrosen@dynamicsoft.com>
Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-publish-02.txt
Date: Tue, 3 Feb 2004 13:27:28 -0000 
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I think you are confusing two different concepts.

If it is not normative it is informative.

If it is optional it is not mandatory.

The two concepts are orthogonal.

By capitalizing the "MAY" you are emphasizing that this is a requirement, and therefore normative, even though it is an optional requirement.

Keith

> -----Original Message-----
> From: aki.niemi@nokia.com [mailto:aki.niemi@nokia.com]
> Sent: 03 February 2004 08:52
> To: jdrosen@dynamicsoft.com
> Cc: sip@ietf.org
> Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-publish-02.txt
> 
> 
> Hi Jonathan,
> 
> Thanks a lot for the excellent comments. Most of them I will 
> directly fold into -03. Few clarifications below.
> 
> Cheers,
> Aki
> 
> Jonathan Rosenberg wrote:
> 
> <snip />
> 
>  > * section 5.1:
>  > 
>  > > As with any other SIP message, the PUBLISH mechanism MAY use the
>  > >    content indirection mechanism defined in [10].
>  > 
>  > this statement makes [10] a normative reference, I believe. It is 
>  > listed as informative in the references section.
> 
> I'm not sure I agree. I've understood the RFC editor's policy 
> so that an informative reference is not required for 
> implementing the RFC. It's an interesting question as to how 
> this relates to RFC 2119 terminology. But I think a MAY makes 
> the feature optional so that it fits the informative 
> reference definition.
> 
> <snip />
> 

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From exim@www1.ietf.org  Tue Feb  3 10:09:42 2004
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Thread-Topic: REL cause code after interwork timer in RFC3398
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From: "El Moussawi, Ali" <ali.el-moussawi@hp.com>
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Hello all,
In RFC 3398, page 16, what do you think the cause code should be in the
REL sent to the PSTN network after interwork tinmer expires ?
Thanks a lot for your help
Ali

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I have the following remarks regarding the GRUU draft:

1. The draft states that one of the GRUU properties is =20
"   o It MUST NOT be possible, based on inspection of the URI, to
determine the associated Contact URI or Address of Record."

It MUST be possible to achieve such a property for a GRUU (if the
Registrar wants to enforce such a policy for its domain), but it is not
part of the requirements for the GRUU concept to always enforce such a
policy.

2. It is expressed throughout the draft that=20
"This extension allows a UA to obtain a GRUU, and to use a GRUU. These
two mechanisms are separate, in that a UA can obtain a GRUU in any way
it likes, and use the mechanisms in this specification to use them.
Similarly, a UA can obtain a GRUU but never use it."

For consistency, I suggest in section "6.2 Using the GRUU" change the
first sentence from

   "A UA first obtains a GRUU using the procedures of Section 6.1."
to=20
   "A UA first obtains a GRUU using the procedures of Section 6.1 or by
other means".

Thanks,
Orit.

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
Jonathan Rosenberg
Sent: Monday, February 02, 2004 10:54 PM
To: Christian Jansson
Cc: Paul Kyzivat; Jussi.Turunen@nokia.com;
nataraju.alilaghatta@wipro.com; sanjsinh@cisco.com; sip@ietf.org
Subject: Re: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and
target refresh request



Christian Jansson wrote:

> Perhaps a little late, and hopefully not off topic. Comments inline.

Not too late, not off topic.

>=20
>=20
>>Jonathan R. wrote
>>This, however, leads me into thinking a bit on the lifecycle
>>management for GRUUs. Currently, GRUU is bound to a single=20
>>registration, where the registration is defined by the Contact URI. If
>=20
>=20
>>the UA should "move", in the sense that it acquires a new IP address,=20
>>this will be a new Contact URI, and therefore, a new GRUU. It would be
>>nice if, instead, the GRUU were truly bound to the UA instance, and=20
>>that even if the UA instance changed IP addresses, the GRUU could=20
>>remain unchanged. In such a case, if the UA moves, there is actually=20
>>no need to issue any target refresh requests. The UA would simply=20
>>re-register, update the binding of its GRUU to its new IP=20
>>address, and=20
>>existing calls would be correctly routed.
>=20
>=20
> But if the ip actually changes, it would be necessary to send an
updated
> SDP with the new ip in the c=3D line.

Yes, that is true. If media relays are used (i.e., TURN), it might be=20
possible to avoid even this, but lets leave that aside for now.

> So your idea won't save you from
> having to send a SIP request to the other end every time the ip
changes
> in an INVITE session.=20

Right. However, it does solve the problem that, previously, if both=20
endpoints moved at the same time, the call would fail - neither side=20
could signal the other.


> For a non-INVITE session though the idea would
> save a lot of request from being sent if the session only contains
> signaling.

Right, in particular, a long lived presence subscription, for example.

-Jonathan R.

--=20
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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Orit Levin wrote:
> I have the following remarks regarding the GRUU draft:
> 
> 1. The draft states that one of the GRUU properties is  
> "   o It MUST NOT be possible, based on inspection of the URI, to
> determine the associated Contact URI or Address of Record."
> 
> It MUST be possible to achieve such a property for a GRUU (if the
> Registrar wants to enforce such a policy for its domain), but it is not
> part of the requirements for the GRUU concept to always enforce such a
> policy.

I have to admit I concur with this position. There might be very useful 
situations where the anonymity aspect of the GRUU would be undesirable.

I can even envision models where the GRUU is cryptographically 
validatable as representing an instance associated with a specific AOR, 
including variations where the validation can or cannot be done without 
prior knowledge of the associated AOR.

As far as I can tell, reversibility does not impact the given rationale 
for the requirements, that:

    With these rules, it is possible, though not required, to construct a
    GRUU without requiring the maintenance of any additional state. To do
    that, the URI would be constructed in the following fashion:

So I believe I would support making this a SHOULD strength requirement. 
Conceivably some discussion of the security implications of NOT meeting 
this requirement would be appropriate.

--
Dean




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aki.niemi@nokia.com wrote:

> Hi Jonathan,
> 
> Thanks a lot for the excellent comments. Most of them I will directly
> fold into -03. Few clarifications below.
> 
> Cheers, Aki
> 
> Jonathan Rosenberg wrote:
> 
> <snip />
> 
>> * section 3, paragraph 3:
>> 
>>> That is, the steady-state of this event package in the absence
>>> of, or in addition
>> to, soft state
>>> provided through the PUBLISH mechanism
>> 
>> this makes it sound like steady state is a function of the event 
>> packae only. the hard state is a function of the resource uRI and 
>> event package. Indeed, one may think of an event package as merely
>>  sub-resource within the full resource. I wonder if it wouldn't be
>>  useful to come up with a term for this - for example, "scoped 
>> resource" is a combination of a URI and event package, and 
>> identifies a particular piece of event state.
> 
> I don't quite follow,  but I agree the quoted text is rather vague.
> To me the hard state, or "preconfigured" state is just another source
> of event state to the composer. The mechanism to set that state might
> be other than PUBLISH.

Right. I agree with that. Let me try again. Requoting the paragraph:

    PUBLISH requests create soft state in the ESC. This state has a
    defined lifetime and will expire after a negotiated amount of time,
    requiring the publication to be refreshed by subsequent PUBLISH
    requests. Local policy at the compositor may in turn define hard
    state for a particular event package. That is, the steady-state of
    this event package in the absence of, or in addition to, soft state
    provided through the PUBLISH mechanism. Setting this hard state or
    configuring the composer policy is out of the scope of this
    specification.

The words "hard state for a particular event package" read to me like it 
meant that hard state was defined on a package by package basis. But, 
its not. There would be hard state generally for a particular 
URI/package basis. So, I'd suggest:

     There may also be hard state provisioned for each resource for
     a particular event package. This hard state represents the resource
     state that is present at all times, and does not expire.


My other comment was that the concept of "for each resource for each 
package" is a common concept, for which I was looking for a term to define.

> 
> <snip />
> 
>> * section 5.1:
>> 
>>> As with any other SIP message, the PUBLISH mechanism MAY use the 
>>> content indirection mechanism defined in [10].
>> 
>> this statement makes [10] a normative reference, I believe. It is 
>> listed as informative in the references section.
> 
> I'm not sure I agree. I've understood the RFC editor's policy so that
> an informative reference is not required for implementing the RFC.
> It's an interesting question as to how this relates to RFC 2119
> terminology. But I think a MAY makes the feature optional so that it
> fits the informative reference definition.

No - Keith is right in his note. Think of it this way - would the 
feature interop if, one day, the referenced I-D evaporated and people 
implemented it differently? In this case, no. If they chose to implement 
the content indirection, and there was no spec, they would do it 
differently and there would be no interop.

I would just drop the text to avoid the dependency.

> 
> <snip />
> 
>> * section 5.3 is out of place. Section 5 is about generating 
>> requests. the subsections for the most part deal with the specific
>> 4 sub-cases you identified in 5.1. However, 5.3 talks about setting
>> expirations, which is something that is actually done for the cases
>> in 5.2, 5.4, 5.5 and 5.6. I would suggest that eaech of those four
>>  sections rather describe how the Expires header field is
>> constructed for each case. That is consistent with having a column
>> in table 1 on expires. Indeed, each section should also talk about
>> bodies and etag. Along those lines, section 5.2 does not actually
>> say that a body has to be present; it shouold be stated given the
>> importance of this fact from table 1.
> 
> I think the reason for this is that I envisioned a PUBLISH carrying
> state in some other place than the body. However, I don't know if
> such a case would ever exist, and other people have also commented on
> this. I think it is safe to assume that the state is always carried
> in the body, as opposed to "typically".

Yes.


>> * the steps in section 6 omit handling for the case where there was
>> no SIP-If-Match header field. step 6 says to check for it, and all
>> the following steps assume it was there. I would suggest adding a
>> bullet to step 6 which says, if there was no SIP-If-Match header
>> field, the ESC creates a new etag, and associates it with as-yet
>> empty event state. Processing continues in the steps below. This
>> way, you always exit step 6 with an etag.
> 
> The attempt was to make etag generation part of step 9. This way, an
> entity-tag is generated for each successful publication and returned
> in 2xx, irrespective of whether the publication was an initial
> publication, a refresh, or a modification. Indeed, a 2xx to a removal
> could be treated the same, although such an entity-tag is useless at
> that point.
> 
> So I think it is ok to exit step 6 without an entity tag, as long as
> step 9 is clear that entity-tags are generated for all successful
> publications.
> 

Step 8 requires an entity tag to be defined, however.


>> * sectiton 11.1:
>> 
>>> Authentication issues are discussed in SIP [2]. The exact
>> methods for
>>> creation and manipulation of the ESC authorization policies are 
>>> outside the scope of this document.
>> 
>> you will get dinged on this by Bellovin, guaranteed. He wants to
>> see statements about baseline mandtory to implement mechahnisms. 
>> Its fine to say that this mechanism is digest, as mandated by
>> rfc3261.
>> 
>>> To prevent replay attacks, implementations SHOULD require 
>>> authentication with anti-replay protection.
>> Authentication issues are
>>> discussed in SIP [2].
>> 
>> as above.... does this mean digest with next-nonce or are you
>> talking sips?
>> 
>> and section 11.4:
>> 
>>> To prevent such attacks, implementations SHOULD, at a minimum, 
>>> provide integrity protection across the To, From, Event, 
>>> SIP-If-Match, Route, and Expires headers and the bodies
>> of PUBLISH
>>> requests.
>> 
>> as above, he will ask for a mechanism. This requiurement makes the
>>  choice sips.
>> 
>> Same in 11.5.
> 
> I see what you are saying. But is it really so that each new method
> that gets added to SIP needs to explicitly mandate certain security
> mechanisms? At least going beyond the security considerations in RFC
> 3261 seems unnecessary, since the properties of PUBLISH are very
> similar to, say, REGISTER.

I have made this same comment in past discussions of specs going through 
iesg. However, the security ADs believe strongly in reptition of 
security requirements in specifications. Trust me, you WILL get dinged 
on this. It is much easier to just add text stating the same normative 
requiremetns as in rfc3261, then to get a discuss during iesg evaluation.

-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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Dean Willis wrote:

> Orit Levin wrote:
> 
>> I have the following remarks regarding the GRUU draft:
>>
>> 1. The draft states that one of the GRUU properties is  "   o It MUST 
>> NOT be possible, based on inspection of the URI, to
>> determine the associated Contact URI or Address of Record."
>>
>> It MUST be possible to achieve such a property for a GRUU (if the
>> Registrar wants to enforce such a policy for its domain), but it is not
>> part of the requirements for the GRUU concept to always enforce such a
>> policy.
> 
> 
> I have to admit I concur with this position. There might be very useful 
> situations where the anonymity aspect of the GRUU would be undesirable.
> 
> I can even envision models where the GRUU is cryptographically 
> validatable as representing an instance associated with a specific AOR, 
> including variations where the validation can or cannot be done without 
> prior knowledge of the associated AOR.
> 
> As far as I can tell, reversibility does not impact the given rationale 
> for the requirements, that:
> 
>    With these rules, it is possible, though not required, to construct a
>    GRUU without requiring the maintenance of any additional state. To do
>    that, the URI would be constructed in the following fashion:
> 
> So I believe I would support making this a SHOULD strength requirement. 
> Conceivably some discussion of the security implications of NOT meeting 
> this requirement would be appropriate.

I am ok to lessen the requirement.

The question becomes, do we want a way to allow the client to signal 
that a randomized gruu is desired, or is it up to the server? I would 
much prefer to leave it up to the server, in which case the only thing 
that we need to change is the requirement, above. I changed the text to 
read:

<t>In many cases, it will be desirable to construct the GRUU in such a
way that it will not be possible, based on inspection of the URI, to
determine the associated Contact URI or Address of Record. Whether or
not a GRUU should be constructed with this property is a local policy
decision of the administrator.
</t>


Orit writes:
> 2. It is expressed throughout the draft that 
> "This extension allows a UA to obtain a GRUU, and to use a GRUU. These
> two mechanisms are separate, in that a UA can obtain a GRUU in any way
> it likes, and use the mechanisms in this specification to use them.
> Similarly, a UA can obtain a GRUU but never use it."
> 
> For consistency, I suggest in section "6.2 Using the GRUU" change the
> first sentence from
> 
>    "A UA first obtains a GRUU using the procedures of Section 6.1."
> to 
>    "A UA first obtains a GRUU using the procedures of Section 6.1 or by
> other means".

OK, with the addition of "outside the scope of this specification.".

-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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Could somebody please confirm that if I am writing a call control
application offering call hold and transfer then the definitive document to
follow is draft-ietf-sipping-service-examples-05.txt?

Matthew Gardiner
Software Engineer
Aculab 
Tel: +44 (0) 1908 273 911
Fax: +44 (0) 1908 273 801
Email: mailto:matthew.gardiner@aculab.com
Website: <http://www.aculab.com>




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<P><FONT SIZE=3D2 FACE=3D"Arial">Could somebody please confirm that if =
I am writing a call control application offering call hold and transfer =
then the definitive document to follow is =
draft-ietf-sipping-service-examples-05.txt?</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Matthew Gardiner</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Software Engineer<BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">Aculab</FONT><FONT SIZE=3D2 =
FACE=3D"Times New Roman"><BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">Tel: +44 (0) 1908 273 =
911</FONT><BR>
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HREF=3D"mailto:matthew.gardiner@aculab.com">mailto:matthew.gardiner@acul=
ab.com</A></FONT>
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Matthew.Gardiner@aculab.com wrote:
> Could somebody please confirm that if I am writing a call control 
> application offering call hold and transfer then the definitive document 
> to follow is draft-ietf-sipping-service-examples-05.txt?

No. The *definitive* document is RFC3261 and related documents. The 
details of call control are derivative from that in somewhat the same 
way the much of physics is derivative from F=MA.

Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) an 
*example*. It is not normative.

However, that draft is certainly helpful and educational. You would do 
well to understand what is in there before completing your task.

	Paul


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Hi folks,
   It's an interesting point.
Paul's comment seems to be "there are many different ways of skinning a 
cat, and the
only normative definition is of the knife".
The IETF defines the details of knife behaviour, and it's up to 
everyone else to sort
out how to use it for their purpose.

My understanding is that the IETF (and the SIP WG in particular) will 
not produce normative
definitions of these (or any other) teleservices, only definitions of 
protocol elements that
might be used for these (or other) purposes.
SIPPING will consider such services and decide on the protocol elements 
they think are needed
(that don't exist already) and then SIP grinds through the details.

Thus the only teleservice examples are in the service-examples document 
(collected together
as a convenience), these are non-normative, and there will never BE 
normative teleservice
definitions produced by these groups. Is this correct?

all the best,
   Lawrence

On 4 Feb 2004, at 3:07 pm, Paul Kyzivat wrote:

>
>
> Matthew.Gardiner@aculab.com wrote:
>> Could somebody please confirm that if I am writing a call control 
>> application offering call hold and transfer then the definitive 
>> document to follow is draft-ietf-sipping-service-examples-05.txt?
>
> No. The *definitive* document is RFC3261 and related documents. The 
> details of call control are derivative from that in somewhat the same 
> way the much of physics is derivative from F=MA.
>
> Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) an 
> *example*. It is not normative.
>
> However, that draft is certainly helpful and educational. You would do 
> well to understand what is in there before completing your task.
>
> 	Paul
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

_______________________________________________
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To: lwc@roke.co.uk, pkyzivat@cisco.com
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Subject: RE: [Sip] Call transfer using SIP
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This vagueness makes it very difficult to engineer a telephony solution
which interoperates with other vendors equipment. In the telephony arena
getting call transfer right is of fundamental importance. A simple document
titled "Call transfer in SIP" would suffice in de-mystifying this area.

-----Original Message-----
From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
Sent: 04 February 2004 16:10
To: Paul Kyzivat
Cc: Matthew.Gardiner@aculab.com; sip@ietf.org
Subject: Re: [Sip] Call transfer using SIP


Hi folks,
   It's an interesting point.
Paul's comment seems to be "there are many different ways of skinning a 
cat, and the
only normative definition is of the knife".
The IETF defines the details of knife behaviour, and it's up to 
everyone else to sort
out how to use it for their purpose.

My understanding is that the IETF (and the SIP WG in particular) will 
not produce normative
definitions of these (or any other) teleservices, only definitions of 
protocol elements that
might be used for these (or other) purposes.
SIPPING will consider such services and decide on the protocol elements 
they think are needed
(that don't exist already) and then SIP grinds through the details.

Thus the only teleservice examples are in the service-examples document 
(collected together
as a convenience), these are non-normative, and there will never BE 
normative teleservice
definitions produced by these groups. Is this correct?

all the best,
   Lawrence

On 4 Feb 2004, at 3:07 pm, Paul Kyzivat wrote:

>
>
> Matthew.Gardiner@aculab.com wrote:
>> Could somebody please confirm that if I am writing a call control 
>> application offering call hold and transfer then the definitive 
>> document to follow is draft-ietf-sipping-service-examples-05.txt?
>
> No. The *definitive* document is RFC3261 and related documents. The 
> details of call control are derivative from that in somewhat the same 
> way the much of physics is derivative from F=MA.
>
> Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) an 
> *example*. It is not normative.
>
> However, that draft is certainly helpful and educational. You would do 
> well to understand what is in there before completing your task.
>
> 	Paul
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

-- 
Registered Office: Roke Manor Research Ltd, Siemens House, Oldbury,
Bracknell,
Berkshire. RG12 8FZ

The information contained in this e-mail and any attachments is confidential
to
Roke Manor Research Ltd and must not be passed to any third party without
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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2653.12">
<TITLE>RE: [Sip] Call transfer using SIP</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>This vagueness makes it very difficult to engineer a =
telephony solution which interoperates with other vendors equipment. In =
the telephony arena getting call transfer right is of fundamental =
importance. A simple document titled &quot;Call transfer in SIP&quot; =
would suffice in de-mystifying this area.</FONT></P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Conroy, Lawrence (SMTP) [<A =
HREF=3D"mailto:lwc@roke.co.uk">mailto:lwc@roke.co.uk</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: 04 February 2004 16:10</FONT>
<BR><FONT SIZE=3D2>To: Paul Kyzivat</FONT>
<BR><FONT SIZE=3D2>Cc: Matthew.Gardiner@aculab.com; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: Re: [Sip] Call transfer using SIP</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Hi folks,</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; It's an interesting point.</FONT>
<BR><FONT SIZE=3D2>Paul's comment seems to be &quot;there are many =
different ways of skinning a </FONT>
<BR><FONT SIZE=3D2>cat, and the</FONT>
<BR><FONT SIZE=3D2>only normative definition is of the =
knife&quot;.</FONT>
<BR><FONT SIZE=3D2>The IETF defines the details of knife behaviour, and =
it's up to </FONT>
<BR><FONT SIZE=3D2>everyone else to sort</FONT>
<BR><FONT SIZE=3D2>out how to use it for their purpose.</FONT>
</P>

<P><FONT SIZE=3D2>My understanding is that the IETF (and the SIP WG in =
particular) will </FONT>
<BR><FONT SIZE=3D2>not produce normative</FONT>
<BR><FONT SIZE=3D2>definitions of these (or any other) teleservices, =
only definitions of </FONT>
<BR><FONT SIZE=3D2>protocol elements that</FONT>
<BR><FONT SIZE=3D2>might be used for these (or other) purposes.</FONT>
<BR><FONT SIZE=3D2>SIPPING will consider such services and decide on =
the protocol elements </FONT>
<BR><FONT SIZE=3D2>they think are needed</FONT>
<BR><FONT SIZE=3D2>(that don't exist already) and then SIP grinds =
through the details.</FONT>
</P>

<P><FONT SIZE=3D2>Thus the only teleservice examples are in the =
service-examples document </FONT>
<BR><FONT SIZE=3D2>(collected together</FONT>
<BR><FONT SIZE=3D2>as a convenience), these are non-normative, and =
there will never BE </FONT>
<BR><FONT SIZE=3D2>normative teleservice</FONT>
<BR><FONT SIZE=3D2>definitions produced by these groups. Is this =
correct?</FONT>
</P>

<P><FONT SIZE=3D2>all the best,</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; Lawrence</FONT>
</P>

<P><FONT SIZE=3D2>On 4 Feb 2004, at 3:07 pm, Paul Kyzivat wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; Matthew.Gardiner@aculab.com wrote:</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; Could somebody please confirm that if I am =
writing a call control </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; application offering call hold and transfer =
then the definitive </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; document to follow is =
draft-ietf-sipping-service-examples-05.txt?</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; No. The *definitive* document is RFC3261 and =
related documents. The </FONT>
<BR><FONT SIZE=3D2>&gt; details of call control are derivative from =
that in somewhat the same </FONT>
<BR><FONT SIZE=3D2>&gt; way the much of physics is derivative from =
F=3DMA.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; Draft-ietf-sipping-service-examples-05.txt is =
(not surprisingly) an </FONT>
<BR><FONT SIZE=3D2>&gt; *example*. It is not normative.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; However, that draft is certainly helpful and =
educational. You would do </FONT>
<BR><FONT SIZE=3D2>&gt; well to understand what is in there before =
completing your task.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
</P>

<P><FONT SIZE=3D2>-- </FONT>
<BR><FONT SIZE=3D2>Registered Office: Roke Manor Research Ltd, Siemens =
House, Oldbury, Bracknell,</FONT>
<BR><FONT SIZE=3D2>Berkshire. RG12 8FZ</FONT>
</P>

<P><FONT SIZE=3D2>The information contained in this e-mail and any =
attachments is confidential to</FONT>
<BR><FONT SIZE=3D2>Roke Manor Research Ltd and must not be passed to =
any third party without</FONT>
<BR><FONT SIZE=3D2>permission. This communication is for information =
only and shall not create or</FONT>
<BR><FONT SIZE=3D2>change any contractual relationship.</FONT>
</P>
<BR>

<P><B><FONT SIZE=3D1>For Aculab's privacy policy and E-Mail disclaimer, =
Click Here : <A HREF=3D"http://www.aculab.com/company/legal_notice.htm" =
TARGET=3D"_blank">http://www.aculab.com/company/legal_notice.htm</A>. =
</FONT></B>
</P>
<BR>

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



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Subject: RE: [Sip] Call transfer using SIP
Date: Wed, 4 Feb 2004 16:35:00 -0000
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Thread-Topic: [Sip] Call transfer using SIP
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From: "Chris Boulton" <cboulton@ubiquity.net>
To: "Conroy, Lawrence (SMTP)" <lwc@roke.co.uk>,
        "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: <Matthew.Gardiner@aculab.com>, <sip@ietf.org>
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Lawrence - remember that SIP is a transport protocol which allows an
objective to be reached in many differing ways.  There is no correct
behavior in many scenarios as long as they fall between the guidelines
set out.  It would be wrong to define normative behavior for such
scenarios and so a push in the right direction towards preferred methods
is the best approach.

Chris.


>-----Original Message-----
>From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
>Sent: 04 February 2004 16:10
>To: Paul Kyzivat
>Cc: Matthew.Gardiner@aculab.com; sip@ietf.org
>Subject: Re: [Sip] Call transfer using SIP
>
>Hi folks,
>   It's an interesting point.
>Paul's comment seems to be "there are many different ways of skinning a
>cat, and the
>only normative definition is of the knife".
>The IETF defines the details of knife behaviour, and it's up to
>everyone else to sort
>out how to use it for their purpose.
>
>My understanding is that the IETF (and the SIP WG in particular) will
>not produce normative
>definitions of these (or any other) teleservices, only definitions of
>protocol elements that
>might be used for these (or other) purposes.
>SIPPING will consider such services and decide on the protocol elements
>they think are needed
>(that don't exist already) and then SIP grinds through the details.
>
>Thus the only teleservice examples are in the service-examples document
>(collected together
>as a convenience), these are non-normative, and there will never BE
>normative teleservice
>definitions produced by these groups. Is this correct?
>
>all the best,
>   Lawrence
>
>On 4 Feb 2004, at 3:07 pm, Paul Kyzivat wrote:
>
>>
>>
>> Matthew.Gardiner@aculab.com wrote:
>>> Could somebody please confirm that if I am writing a call control
>>> application offering call hold and transfer then the definitive
>>> document to follow is draft-ietf-sipping-service-examples-05.txt?
>>
>> No. The *definitive* document is RFC3261 and related documents. The
>> details of call control are derivative from that in somewhat the same
>> way the much of physics is derivative from F=3DMA.
>>
>> Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) an
>> *example*. It is not normative.
>>
>> However, that draft is certainly helpful and educational. You would
do
>> well to understand what is in there before completing your task.
>>
>> 	Paul
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


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Cc: Paul Kyzivat <pkyzivat@cisco.com>, Matthew.Gardiner@aculab.com,
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Subject: Re: [Sip] Call transfer using SIP
Date: Wed, 4 Feb 2004 17:22:54 +0000
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Hi Chris, folks,
   That's what I said - the SIP/SIPPING groups will not produce normative
teleservice definitions - we're in TSV and a teleservice is in/above 
APPs.

I do sympathise with those folks who want a definition so they can 
interwork
with other vendor's kit in implementing those teleservices.

Short answer is - yup - it would be a good idea, but not by the IETF.

I note in passing that Alan Johnston works for a fine Telecom Service 
Provider
that might be expected to have an interest in the way teleservices are 
implemented
by different vendors.

all the best,
   Lawrence

On 4 Feb 2004, at 4:35 pm, Chris Boulton wrote:

> Lawrence - remember that SIP is a transport protocol which allows an
> objective to be reached in many differing ways.  There is no correct
> behavior in many scenarios as long as they fall between the guidelines
> set out.  It would be wrong to define normative behavior for such
> scenarios and so a push in the right direction towards preferred 
> methods
> is the best approach.
>
> Chris.
>
>
>> -----Original Message-----
>> From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
>> Sent: 04 February 2004 16:10
>> To: Paul Kyzivat
>> Cc: Matthew.Gardiner@aculab.com; sip@ietf.org
>> Subject: Re: [Sip] Call transfer using SIP
>>
>> Hi folks,
>>   It's an interesting point.
>> Paul's comment seems to be "there are many different ways of skinning 
>> a
>> cat, and the
>> only normative definition is of the knife".
>> The IETF defines the details of knife behaviour, and it's up to
>> everyone else to sort
>> out how to use it for their purpose.
>>
>> My understanding is that the IETF (and the SIP WG in particular) will
>> not produce normative
>> definitions of these (or any other) teleservices, only definitions of
>> protocol elements that
>> might be used for these (or other) purposes.
>> SIPPING will consider such services and decide on the protocol 
>> elements
>> they think are needed
>> (that don't exist already) and then SIP grinds through the details.
>>
>> Thus the only teleservice examples are in the service-examples 
>> document
>> (collected together
>> as a convenience), these are non-normative, and there will never BE
>> normative teleservice
>> definitions produced by these groups. Is this correct?
>>
>> all the best,
>>   Lawrence
>>
>> On 4 Feb 2004, at 3:07 pm, Paul Kyzivat wrote:
>>
>>>
>>>
>>> Matthew.Gardiner@aculab.com wrote:
>>>> Could somebody please confirm that if I am writing a call control
>>>> application offering call hold and transfer then the definitive
>>>> document to follow is draft-ietf-sipping-service-examples-05.txt?
>>>
>>> No. The *definitive* document is RFC3261 and related documents. The
>>> details of call control are derivative from that in somewhat the same
>>> way the much of physics is derivative from F=MA.
>>>
>>> Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) an
>>> *example*. It is not normative.
>>>
>>> However, that draft is certainly helpful and educational. You would
> do
>>> well to understand what is in there before completing your task.
>>>
>>> 	Paul
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
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>> Use sipping@ietf.org for new developments on the application of sip
>
>
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> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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> Use sipping@ietf.org for new developments on the application of sip
>

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Subject: RE: [Sip] draft-ietf-sip-gruu-00
Date: Wed, 4 Feb 2004 09:51:08 -0800
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Thread-Topic: [Sip] draft-ietf-sip-gruu-00
Thread-Index: AcPq5KM+wSeiEdDQSo+0FYPzkOQTtQAYqXTQ
From: "Orit Levin" <oritl@microsoft.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Dean Willis" <dean.willis@softarmor.com>
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Sounds perfect to me.
Thanks,
Orit.

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]=20
Sent: Tuesday, February 03, 2004 10:03 PM
To: Dean Willis
Cc: Orit Levin; sip@ietf.org
Subject: Re: [Sip] draft-ietf-sip-gruu-00

inline.

Dean Willis wrote:

> Orit Levin wrote:
>=20
>> I have the following remarks regarding the GRUU draft:
>>
>> 1. The draft states that one of the GRUU properties is  "   o It MUST

>> NOT be possible, based on inspection of the URI, to
>> determine the associated Contact URI or Address of Record."
>>
>> It MUST be possible to achieve such a property for a GRUU (if the
>> Registrar wants to enforce such a policy for its domain), but it is
not
>> part of the requirements for the GRUU concept to always enforce such
a
>> policy.
>=20
>=20
> I have to admit I concur with this position. There might be very
useful=20
> situations where the anonymity aspect of the GRUU would be
undesirable.
>=20
> I can even envision models where the GRUU is cryptographically=20
> validatable as representing an instance associated with a specific
AOR,=20
> including variations where the validation can or cannot be done
without=20
> prior knowledge of the associated AOR.
>=20
> As far as I can tell, reversibility does not impact the given
rationale=20
> for the requirements, that:
>=20
>    With these rules, it is possible, though not required, to construct
a
>    GRUU without requiring the maintenance of any additional state. To
do
>    that, the URI would be constructed in the following fashion:
>=20
> So I believe I would support making this a SHOULD strength
requirement.=20
> Conceivably some discussion of the security implications of NOT
meeting=20
> this requirement would be appropriate.

I am ok to lessen the requirement.

The question becomes, do we want a way to allow the client to signal=20
that a randomized gruu is desired, or is it up to the server? I would=20
much prefer to leave it up to the server, in which case the only thing=20
that we need to change is the requirement, above. I changed the text to=20
read:

<t>In many cases, it will be desirable to construct the GRUU in such a
way that it will not be possible, based on inspection of the URI, to
determine the associated Contact URI or Address of Record. Whether or
not a GRUU should be constructed with this property is a local policy
decision of the administrator.
</t>


Orit writes:
> 2. It is expressed throughout the draft that=20
> "This extension allows a UA to obtain a GRUU, and to use a GRUU. These
> two mechanisms are separate, in that a UA can obtain a GRUU in any way
> it likes, and use the mechanisms in this specification to use them.
> Similarly, a UA can obtain a GRUU but never use it."
>=20
> For consistency, I suggest in section "6.2 Using the GRUU" change the
> first sentence from
>=20
>    "A UA first obtains a GRUU using the procedures of Section 6.1."
> to=20
>    "A UA first obtains a GRUU using the procedures of Section 6.1 or
by
> other means".

OK, with the addition of "outside the scope of this specification.".

-Jonathan R.



--=20
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] draft-ietf-sip-gruu-00
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Jonathan Rosenberg wrote:
> <t>In many cases, it will be desirable to construct the GRUU in such a
> way that it will not be possible, based on inspection of the URI, to
> determine the associated Contact URI or Address of Record. Whether or
> not a GRUU should be constructed with this property is a local policy
> decision of the administrator.
> </t>

This works for me. You might even strike " of the administrator" from 
the  end.

--
Dean


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CC: Chris Boulton <cboulton@ubiquity.net>, Paul Kyzivat <pkyzivat@cisco.com>,
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Subject: Re: [Sip] Call transfer using SIP
References: <45730E094814E44488F789C1CDED27AE0219B186@gbnewp0758m.eu.ubiquity.net> <C4548039-5736-11D8-80DF-00039355516C@roke.co.uk>
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Mathew asked:

 > Could somebody please confirm that if I am writing a call control
 > application offering call hold and transfer then the definitive
 > document to follow is draft-ietf-sipping-service-examples-05.txt?

and Lawrence said:

> Short answer is - yup - it would be a good idea, but not by the IETF.

So, I might ask, would it be worth encouraging our friends at SIP Forum 
to publish some "best practices" in this regard?

--
Dean


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



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Subject: Re: [Sip] Call transfer using SIP
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See also:

http://www.ietf.org/internet-drafts/draft-ietf-sip-referredby-03.txt

The following, another pertinent draft,  seems to have expired for some
reason:

http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-transfer-02.txt

Alex.



Matthew.Gardiner@aculab.com wrote:

>
>
> This vagueness makes it very difficult to engineer a telephony
> solution which interoperates with other vendors equipment. In the
> telephony arena getting call transfer right is of fundamental
> importance. A simple document titled "Call transfer in SIP" would
> suffice in de-mystifying this area.
>
> -----Original Message-----
> From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
> Sent: 04 February 2004 16:10
> To: Paul Kyzivat
> Cc: Matthew.Gardiner@aculab.com; sip@ietf.org
> Subject: Re: [Sip] Call transfer using SIP
>
> Hi folks,
>    It's an interesting point.
> Paul's comment seems to be "there are many different ways of skinning
> a
> cat, and the
> only normative definition is of the knife".
> The IETF defines the details of knife behaviour, and it's up to
> everyone else to sort
> out how to use it for their purpose.
>
> My understanding is that the IETF (and the SIP WG in particular) will
> not produce normative
> definitions of these (or any other) teleservices, only definitions of
> protocol elements that
> might be used for these (or other) purposes.
> SIPPING will consider such services and decide on the protocol
> elements
> they think are needed
> (that don't exist already) and then SIP grinds through the details.
>
> Thus the only teleservice examples are in the service-examples
> document
> (collected together
> as a convenience), these are non-normative, and there will never BE
> normative teleservice
> definitions produced by these groups. Is this correct?
>
> all the best,
>    Lawrence
>
> On 4 Feb 2004, at 3:07 pm, Paul Kyzivat wrote:
>
> >
> >
> > Matthew.Gardiner@aculab.com wrote:
> >> Could somebody please confirm that if I am writing a call control
> >> application offering call hold and transfer then the definitive
> >> document to follow is draft-ietf-sipping-service-examples-05.txt?
> >
> > No. The *definitive* document is RFC3261 and related documents. The
> > details of call control are derivative from that in somewhat the
> same
> > way the much of physics is derivative from F=MA.
> >
> > Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) an
> > *example*. It is not normative.
> >
> > However, that draft is certainly helpful and educational. You would
> do
> > well to understand what is in there before completing your task.
> >
> >       Paul
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
> --
> Registered Office: Roke Manor Research Ltd, Siemens House, Oldbury,
> Bracknell,
> Berkshire. RG12 8FZ
>
> The information contained in this e-mail and any attachments is
> confidential to
> Roke Manor Research Ltd and must not be passed to any third party
> without
> permission. This communication is for information only and shall not
> create or
> change any contractual relationship.
>
> For Aculab's privacy policy and E-Mail disclaimer, Click Here :
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See also:
<p><A HREF="http://www.ietf.org/internet-drafts/draft-ietf-sip-referredby-03.txt">http://www.ietf.org/internet-drafts/draft-ietf-sip-referredby-03.txt</A>
<p>The following, another pertinent draft,&nbsp; seems to have expired
for some reason:
<br>&nbsp; <A HREF="http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-transfer-02.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-transfer-02.txt</A>
<p>Alex.
<br>&nbsp;
<br>&nbsp;
<p>Matthew.Gardiner@aculab.com wrote:
<blockquote TYPE=CITE>&nbsp;
<p><font size=-1>This vagueness makes it very difficult to engineer a telephony
solution which interoperates with other vendors equipment. In the telephony
arena getting call transfer right is of fundamental importance. A simple
document titled "Call transfer in SIP" would suffice in de-mystifying this
area.</font>
<p><font size=-1>-----Original Message-----</font>
<br><font size=-1>From: Conroy, Lawrence (SMTP) [<a href="mailto:lwc@roke.co.uk">mailto:lwc@roke.co.uk</a>]</font>
<br><font size=-1>Sent: 04 February 2004 16:10</font>
<br><font size=-1>To: Paul Kyzivat</font>
<br><font size=-1>Cc: Matthew.Gardiner@aculab.com; sip@ietf.org</font>
<br><font size=-1>Subject: Re: [Sip] Call transfer using SIP</font>
<p><font size=-1>Hi folks,</font>
<br><font size=-1>&nbsp;&nbsp; It's an interesting point.</font>
<br><font size=-1>Paul's comment seems to be "there are many different
ways of skinning a</font>
<br><font size=-1>cat, and the</font>
<br><font size=-1>only normative definition is of the knife".</font>
<br><font size=-1>The IETF defines the details of knife behaviour, and
it's up to</font>
<br><font size=-1>everyone else to sort</font>
<br><font size=-1>out how to use it for their purpose.</font>
<p><font size=-1>My understanding is that the IETF (and the SIP WG in particular)
will</font>
<br><font size=-1>not produce normative</font>
<br><font size=-1>definitions of these (or any other) teleservices, only
definitions of</font>
<br><font size=-1>protocol elements that</font>
<br><font size=-1>might be used for these (or other) purposes.</font>
<br><font size=-1>SIPPING will consider such services and decide on the
protocol elements</font>
<br><font size=-1>they think are needed</font>
<br><font size=-1>(that don't exist already) and then SIP grinds through
the details.</font>
<p><font size=-1>Thus the only teleservice examples are in the service-examples
document</font>
<br><font size=-1>(collected together</font>
<br><font size=-1>as a convenience), these are non-normative, and there
will never BE</font>
<br><font size=-1>normative teleservice</font>
<br><font size=-1>definitions produced by these groups. Is this correct?</font>
<p><font size=-1>all the best,</font>
<br><font size=-1>&nbsp;&nbsp; Lawrence</font>
<p><font size=-1>On 4 Feb 2004, at 3:07 pm, Paul Kyzivat wrote:</font>
<p><font size=-1>></font>
<br><font size=-1>></font>
<br><font size=-1>> Matthew.Gardiner@aculab.com wrote:</font>
<br><font size=-1>>> Could somebody please confirm that if I am writing
a call control</font>
<br><font size=-1>>> application offering call hold and transfer then the
definitive</font>
<br><font size=-1>>> document to follow is draft-ietf-sipping-service-examples-05.txt?</font>
<br><font size=-1>></font>
<br><font size=-1>> No. The *definitive* document is RFC3261 and related
documents. The</font>
<br><font size=-1>> details of call control are derivative from that in
somewhat the same</font>
<br><font size=-1>> way the much of physics is derivative from F=MA.</font>
<br><font size=-1>></font>
<br><font size=-1>> Draft-ietf-sipping-service-examples-05.txt is (not
surprisingly) an</font>
<br><font size=-1>> *example*. It is not normative.</font>
<br><font size=-1>></font>
<br><font size=-1>> However, that draft is certainly helpful and educational.
You would do</font>
<br><font size=-1>> well to understand what is in there before completing
your task.</font>
<br><font size=-1>></font>
<br><font size=-1>>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul</font>
<br><font size=-1>></font>
<br><font size=-1>></font>
<br><font size=-1>> _______________________________________________</font>
<br><font size=-1>> Sip mailing list&nbsp; <a href="https://www1.ietf.org/mailman/listinfo/sip" TARGET="_blank">https://www1.ietf.org/mailman/listinfo/sip</a></font>
<br><font size=-1>> This list is for NEW development of the core SIP Protocol</font>
<br><font size=-1>> Use sip-implementors@cs.columbia.edu for questions
on current sip</font>
<br><font size=-1>> Use sipping@ietf.org for new developments on the application
of sip</font>
<br><font size=-1>></font>
<p><font size=-1>--</font>
<br><font size=-1>Registered Office: Roke Manor Research Ltd, Siemens House,
Oldbury, Bracknell,</font>
<br><font size=-1>Berkshire. RG12 8FZ</font>
<p><font size=-1>The information contained in this e-mail and any attachments
is confidential to</font>
<br><font size=-1>Roke Manor Research Ltd and must not be passed to any
third party without</font>
<br><font size=-1>permission. This communication is for information only
and shall not create or</font>
<br><font size=-1>change any contractual relationship.</font>
<p><b><font size=-2>For Aculab's privacy policy and E-Mail disclaimer,
Click Here : <a href="http://www.aculab.com/company/legal_notice.htm" TARGET="_blank">http://www.aculab.com/company/legal_notice.htm</a>.</font></b>
<br>&nbsp;</blockquote>
</html>

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



From exim@www1.ietf.org  Wed Feb  4 17:15:51 2004
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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'PROUVOST Sébastien FTRD/DAC/ISS'"
	 <sebastien.prouvost@francetelecom.com>,
        "Jesske, R" <R.Jesske@t-com.net>
Cc: sip@ietf.org
Subject: RE: RE : [Sip] draft-ietf-sip-history-info-01.txt
Date: Wed, 4 Feb 2004 16:12:30 -0600 
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Sébastien and Roland,

I apologize for the delay in responding.  

You are correct in that one privacy aspect currently addressed for
History-Info is around the UA's ability to require privacy for elements of a
message that can somehow divulge information about the end client's
identity. In some ways, History-Info could be considered in the category of
Record Route,  Via, etc.  However, HI is an optional header and per the
recommendation in RFC 3323, the decision was made that if Header privacy was
requested, then History-Info SHOULD not be included.   In addition, if
through local policy or some other local information available wrt the
retargeted URIs, there is knowledge that there should be privacy applied to
these URIs, then as discussed in RFC 3323, the intermediary processing that
request can also add the Privacy header to ensure that History-Info is also
not captured by subsequent intermediaries.  This latter point perhaps is not
so clearly or explicitly spelled out in the current document in the privacy
sections, but there is discussion in the detailed proxy processing behaviour
in 2.3.3 that the local policy can define whether the History-Info captured
would leave a specific domain (this is of course for optionality, as well,
which is another issue that I'll address in a separate email).      

It seems that what's being discussed in these emails is introducing a
specific privacy tag against those entries that local policy defines should
not be forwarded outside a specific domain,  so that they can be removed
(and don't need to be kept persistent by a privacy service).  Right now,
there's an asumption that these entries are somehow known by the proxy, but
perhaps an explicit tag would be useful.  OR are you talking about extending
the current model in RFC 3323 to specifically address the use of a privacy
service for History-Info headers?  

Regards,
Mary H. Barnes
mary.barnes@nortelnetworks.com
972-684-5432


-----Original Message-----
From: PROUVOST Sébastien FTRD/DAC/ISS
[mailto:sebastien.prouvost@francetelecom.com]
Sent: Thursday, January 22, 2004 4:32 AM
To: Jesske, R; sip@ietf.org; Barnes, Mary [NGC:B622:EXCH]; Watson, Mark
[MOP:EP10:EXCH]; fluffy@cisco.com
Subject: RE : [Sip] draft-ietf-sip-history-info-01.txt


I fully support this proposal. 

The privacy regarding the history-info header relates to the privacy policy
accorded to the target of the request (before it is retargeted). Therefore
it shall not depend on the privacy policy accorded to the calling UA. The
history-info header shall have its own privacy value. 
Moreover in case the request is retargeted more than once (particularly if
the targets represent different users), it shall be possible to request
privacy for a target independently of the others.  

Regards,
Sébastien Prouvost
France Télécom R&D/DAC/CAR

-----Message d'origine-----
De : Jesske, R [mailto:R.Jesske@t-com.net] 
Envoyé : vendredi 16 janvier 2004 10:29
À : sip@ietf.org
Cc : Poetzl, Joachim; Alexeitsev, D
Objet : [Sip] draft-ietf-sip-history-info-01.txt


Dear all,
in 1.3 Ensuring the Privacy of History_Info it is proposed to use a
priv-value of Session or Header level privacy. I would like to propose a own
privacy level for the history info, because the Information shows also
information about the network. With regard to the provider operating a
network some of the providers don't like to transmit information about call
history. Or a redirecting party don't like to show the redirecting address.
Therefore it should be possible to delete the history info at domain
boundaries based on the privacy-value.

I propose the priv-value   =   "history_info"  for the whole History-Info
header.

In addition a option would be useful where only parts (e.g Indes1.1) of the
history info are restricted. That could be done with a privacy statement
within the History-Info .

Best Regards

Roland


Deutsche Telekom AG
T Com Zentrale
Roland Jesske, T332-2
Section T33; Signalling, Gateways and Switching Systems 
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940 
Fax:      +49 6151 83-4577 
email:   r.jesske@t-com.net




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


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



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Subject: Re: [Sip] Session timer question
From: Hiroaki Hata <hata@sphere.ad.jp>
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Hi
Even if UAC does not support timer and ignores sesssion expires header
in a 2xx response, a session could be refreshed in case that UAC can response
a 2xx to re-INVITE from UAS. If UAC is not able to recongize a difference between
INVITE and re-INVITE and responds 4xx to the request of the refresh, a session
would be terminated. It depends just upon the implementation of UAC, doesn't it?

PS.The table 2 Jonathan mentioned below is the Figure 3 in draft-13 but it is still 
called table 3 in text page20.

Hiro
--
Hiroaki Hata
NTTPC Communications,Inc.
+81-3-5212-2101


> inline.
> 
> Veda Hegde wrote:
> 
> > Keith: Thanks for pointing this out and I will go with this in my
> > implementation.
> > 
> > I think, the 2nd paragraph in "Overview of operation" in the draft
> > should be reworded. It says, 
> > ----------------------------------------------------------------------------
> >  --------- This extension has the property that it works even when
> > only one UA in a dialog supports it. The processing steps differ
> > for handling each of the four cases (UAC supports it, or doesn't,
> > and UAS supports it, or doesn't). 
> > ----------------------------------------------------------------------------
> >  --------
> > 
> > Doesn't this mean even if UAC does not support timer, UAS could
> > support it?
> 
> Yes, and this mode is supported. In such a case, the INVITE from the 
> UAC would lack a Supported header field. Proxies insert 
> Session-Expires headers. The UAS receives it. Based on table 2 in the 
> session timer spec, since the UAS supports the extension, it sets the 
> refresher parameter to "uas", and generates a 2xx with the final 
> session timer in the session expires header. This header is ignored by 
> the UAC. The called party will generate refresh requests for the 
> remainder of the call.
> 
> Note that there is no Require in the 2xx in this case.
> 
> -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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This list is for NEW development of the core SIP Protocol
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Subject: Re: [Sip] draft-ietf-sip-gruu-00
References: <DD07841287D0AD428833021705E0D14E015025C3@RED-MSG-52.redmond.corp.microsoft.com> <402035B7.5030206@softarmor.com> <40208B24.5020004@dynamicsoft.com> <4021470D.7050104@softarmor.com>
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Dean Willis wrote:

> Jonathan Rosenberg wrote:
> 
>> <t>In many cases, it will be desirable to construct the GRUU in such a
>> way that it will not be possible, based on inspection of the URI, to
>> determine the associated Contact URI or Address of Record. Whether or
>> not a GRUU should be constructed with this property is a local policy
>> decision of the administrator.
>> </t>
> 
> 
> This works for me. You might even strike " of the administrator" from 
> the  end.

Done.

-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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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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To: lwc@roke.co.uk, cboulton@ubiquity.net
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Subject: RE: [Sip] Call transfer using SIP
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Maybe the author of "Draft-ietf-sipping-service-examples-05.txt" could
confirm how definitive the message flows regarding call transfer are
therein. (Or point out a preferred document of reference).

-----Original Message-----
From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
Sent: 04 February 2004 17:23
To: Chris Boulton
Cc: Paul Kyzivat; Matthew.Gardiner@aculab.com; sip@ietf.org
Subject: Re: [Sip] Call transfer using SIP


Hi Chris, folks,
   That's what I said - the SIP/SIPPING groups will not produce normative
teleservice definitions - we're in TSV and a teleservice is in/above 
APPs.

I do sympathise with those folks who want a definition so they can 
interwork
with other vendor's kit in implementing those teleservices.

Short answer is - yup - it would be a good idea, but not by the IETF.

I note in passing that Alan Johnston works for a fine Telecom Service 
Provider
that might be expected to have an interest in the way teleservices are 
implemented
by different vendors.

all the best,
   Lawrence

On 4 Feb 2004, at 4:35 pm, Chris Boulton wrote:

> Lawrence - remember that SIP is a transport protocol which allows an
> objective to be reached in many differing ways.  There is no correct
> behavior in many scenarios as long as they fall between the guidelines
> set out.  It would be wrong to define normative behavior for such
> scenarios and so a push in the right direction towards preferred 
> methods
> is the best approach.
>
> Chris.
>
>
>> -----Original Message-----
>> From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
>> Sent: 04 February 2004 16:10
>> To: Paul Kyzivat
>> Cc: Matthew.Gardiner@aculab.com; sip@ietf.org
>> Subject: Re: [Sip] Call transfer using SIP
>>
>> Hi folks,
>>   It's an interesting point.
>> Paul's comment seems to be "there are many different ways of skinning 
>> a
>> cat, and the
>> only normative definition is of the knife".
>> The IETF defines the details of knife behaviour, and it's up to
>> everyone else to sort
>> out how to use it for their purpose.
>>
>> My understanding is that the IETF (and the SIP WG in particular) will
>> not produce normative
>> definitions of these (or any other) teleservices, only definitions of
>> protocol elements that
>> might be used for these (or other) purposes.
>> SIPPING will consider such services and decide on the protocol 
>> elements
>> they think are needed
>> (that don't exist already) and then SIP grinds through the details.
>>
>> Thus the only teleservice examples are in the service-examples 
>> document
>> (collected together
>> as a convenience), these are non-normative, and there will never BE
>> normative teleservice
>> definitions produced by these groups. Is this correct?
>>
>> all the best,
>>   Lawrence
>>
>> On 4 Feb 2004, at 3:07 pm, Paul Kyzivat wrote:
>>
>>>
>>>
>>> Matthew.Gardiner@aculab.com wrote:
>>>> Could somebody please confirm that if I am writing a call control
>>>> application offering call hold and transfer then the definitive
>>>> document to follow is draft-ietf-sipping-service-examples-05.txt?
>>>
>>> No. The *definitive* document is RFC3261 and related documents. The
>>> details of call control are derivative from that in somewhat the same
>>> way the much of physics is derivative from F=MA.
>>>
>>> Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) an
>>> *example*. It is not normative.
>>>
>>> However, that draft is certainly helpful and educational. You would
> do
>>> well to understand what is in there before completing your task.
>>>
>>> 	Paul
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
>
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> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2653.12">
<TITLE>RE: [Sip] Call transfer using SIP</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Maybe the author of =
&quot;Draft-ietf-sipping-service-examples-05.txt&quot; could confirm =
how definitive the message flows regarding call transfer are therein. =
(Or point out a preferred document of reference).</FONT></P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Conroy, Lawrence (SMTP) [<A =
HREF=3D"mailto:lwc@roke.co.uk">mailto:lwc@roke.co.uk</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: 04 February 2004 17:23</FONT>
<BR><FONT SIZE=3D2>To: Chris Boulton</FONT>
<BR><FONT SIZE=3D2>Cc: Paul Kyzivat; Matthew.Gardiner@aculab.com; =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: Re: [Sip] Call transfer using SIP</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Hi Chris, folks,</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; That's what I said - the SIP/SIPPING =
groups will not produce normative</FONT>
<BR><FONT SIZE=3D2>teleservice definitions - we're in TSV and a =
teleservice is in/above </FONT>
<BR><FONT SIZE=3D2>APPs.</FONT>
</P>

<P><FONT SIZE=3D2>I do sympathise with those folks who want a =
definition so they can </FONT>
<BR><FONT SIZE=3D2>interwork</FONT>
<BR><FONT SIZE=3D2>with other vendor's kit in implementing those =
teleservices.</FONT>
</P>

<P><FONT SIZE=3D2>Short answer is - yup - it would be a good idea, but =
not by the IETF.</FONT>
</P>

<P><FONT SIZE=3D2>I note in passing that Alan Johnston works for a fine =
Telecom Service </FONT>
<BR><FONT SIZE=3D2>Provider</FONT>
<BR><FONT SIZE=3D2>that might be expected to have an interest in the =
way teleservices are </FONT>
<BR><FONT SIZE=3D2>implemented</FONT>
<BR><FONT SIZE=3D2>by different vendors.</FONT>
</P>

<P><FONT SIZE=3D2>all the best,</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; Lawrence</FONT>
</P>

<P><FONT SIZE=3D2>On 4 Feb 2004, at 4:35 pm, Chris Boulton =
wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; Lawrence - remember that SIP is a transport =
protocol which allows an</FONT>
<BR><FONT SIZE=3D2>&gt; objective to be reached in many differing =
ways.&nbsp; There is no correct</FONT>
<BR><FONT SIZE=3D2>&gt; behavior in many scenarios as long as they fall =
between the guidelines</FONT>
<BR><FONT SIZE=3D2>&gt; set out.&nbsp; It would be wrong to define =
normative behavior for such</FONT>
<BR><FONT SIZE=3D2>&gt; scenarios and so a push in the right direction =
towards preferred </FONT>
<BR><FONT SIZE=3D2>&gt; methods</FONT>
<BR><FONT SIZE=3D2>&gt; is the best approach.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; Chris.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; From: Conroy, Lawrence (SMTP) [<A =
HREF=3D"mailto:lwc@roke.co.uk">mailto:lwc@roke.co.uk</A>]</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; Sent: 04 February 2004 16:10</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; To: Paul Kyzivat</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; Cc: Matthew.Gardiner@aculab.com; =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; Subject: Re: [Sip] Call transfer using =
SIP</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; Hi folks,</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&nbsp;&nbsp; It's an interesting =
point.</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; Paul's comment seems to be &quot;there are =
many different ways of skinning </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; a</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; cat, and the</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; only normative definition is of the =
knife&quot;.</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; The IETF defines the details of knife =
behaviour, and it's up to</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; everyone else to sort</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; out how to use it for their purpose.</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; My understanding is that the IETF (and the =
SIP WG in particular) will</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; not produce normative</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; definitions of these (or any other) =
teleservices, only definitions of</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; protocol elements that</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; might be used for these (or other) =
purposes.</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; SIPPING will consider such services and =
decide on the protocol </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; elements</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; they think are needed</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; (that don't exist already) and then SIP =
grinds through the details.</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; Thus the only teleservice examples are in =
the service-examples </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; document</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; (collected together</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; as a convenience), these are non-normative, =
and there will never BE</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; normative teleservice</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; definitions produced by these groups. Is =
this correct?</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; all the best,</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&nbsp;&nbsp; Lawrence</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; On 4 Feb 2004, at 3:07 pm, Paul Kyzivat =
wrote:</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; Matthew.Gardiner@aculab.com =
wrote:</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;&gt; Could somebody please confirm that =
if I am writing a call control</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;&gt; application offering call hold and =
transfer then the definitive</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;&gt; document to follow is =
draft-ietf-sipping-service-examples-05.txt?</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; No. The *definitive* document is =
RFC3261 and related documents. The</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; details of call control are derivative =
from that in somewhat the same</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; way the much of physics is derivative =
from F=3DMA.</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; =
Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) =
an</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; *example*. It is not normative.</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; However, that draft is certainly =
helpful and educational. You would</FONT>
<BR><FONT SIZE=3D2>&gt; do</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; well to understand what is in there =
before completing your task.</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; &nbsp;&nbsp;&nbsp; Paul</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; This list is for NEW development of the =
core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; Use sip-implementors@cs.columbia.edu =
for questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; Use sipping@ietf.org for new =
developments on the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt;&gt; This list is for NEW development of the =
core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; Use sipping@ietf.org for new developments =
on the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; This message has been scanned for viruses by =
MailControl - </FONT>
<BR><FONT SIZE=3D2>&gt; www.mailcontrol.com</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
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<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
</P>

<P><FONT SIZE=3D2>-- </FONT>
<BR><FONT SIZE=3D2>Registered Office: Roke Manor Research Ltd, Siemens =
House, Oldbury, Bracknell,</FONT>
<BR><FONT SIZE=3D2>Berkshire. RG12 8FZ</FONT>
</P>

<P><FONT SIZE=3D2>The information contained in this e-mail and any =
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<BR><FONT SIZE=3D2>Roke Manor Research Ltd and must not be passed to =
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From exim@www1.ietf.org  Thu Feb  5 02:51:40 2004
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Date: Thu, 05 Feb 2004 02:48:24 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Hiroaki Hata <hata@sphere.ad.jp>
CC: vhegde@san.rr.com, drage@lucent.com, pkyzivat@cisco.com,
        Brian.Rosen@marconi.com, sip@ietf.org
Subject: Re: [Sip] Session timer question
References: <475FF955A05DD411980D00508B6D5FB00439EF42@en0033exch001u.uk.lucent.com>	<001001c3b841$b5c99d60$2506c042@WXPvhegde>	<3FD0E40A.7020008@dynamicsoft.com> <200402051452.BCE64762.BBIU@sphere.ad.jp>
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Hiroaki Hata wrote:

> Hi
> Even if UAC does not support timer and ignores sesssion expires header
> in a 2xx response, a session could be refreshed in case that UAC can response
> a 2xx to re-INVITE from UAS. If UAC is not able to recongize a difference between
> INVITE and re-INVITE and responds 4xx to the request of the refresh, a session
> would be terminated. It depends just upon the implementation of UAC, doesn't it?

Why would the caller reject the re-INVITE?

> 
> PS.The table 2 Jonathan mentioned below is the Figure 3 in draft-13 but it is still 
> called table 3 in text page20.

Thanks for pointing this out.

-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: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and target
 refresh request
References: <FE03AFC4B33E7447979123987BD65F450EDB15@exchange.hotsip.com> <401F4591.2040500@dynamicsoft.com> <004901c3ea24$576c1850$f504120a@in.huawei.com>
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Natesan Kannan wrote:

> I will attempt a wild swing too...
> Though we are talking of terminal mobility, it seems to me that this
> mobility is restricted because of the fixed routeset in the dialog.
> I can see that GRUU mapping to instance id, and the fact that GRUU can be
> globally routed, paves a way for an un-restricted terminal mobility, just in
> case nobody record-routed.
> Was wondering if this could be of any use...

It is true that the signaling path would be nailed down. This might be 
problematic for cases where a user roams into a totally different 
service provider, and that provider requires signaling to go through 
their signaling network in order for media to flow through their access 
network.

In such a case, what makes sense is for the client to generate a totally 
new INVITE, and use Replaces to replace the dialog at the called party. 
The original dialog is then torn down. Lest people complain about 
billing implications, the replaces header would allow for sufficient 
correlation of the two dialogs to account for this as if it was still 
one single call.

-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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I have a question about the proxy behavior under the following scenario,
could someone please provide a clarification.

1. UAC sends an INVITE via proxy.
2. Proxy responds back with 100 trying message to UAC and forwards INVITE to
UAS.
3. UAS sends 200 OK which is relayed by the proxy to UAC.
4. UAC sends an ACK which is being processed by the proxy to be forwarded to
the UAS.
** Here ACK has sequence number 1
5. UAC sends another INVITE and incidentally this gets forwarded to the UAS
before the ACK from previous INVITE by the proxy.
** Here INVITE has sequence number 2
6. Since UAS did not receive the ACK for previous INVITE (nor did it
received CANCEL/BYE), but received another INVITE with same Call-ID etc. it
throws a "500 Internal Server Error" error.

According to RFC 3261, the UAS behavior to throw a "500 Internal Server
Error" is correct. But I am not sure about the proxy behavior here, could
not be clearly figured out of 3261.

Is proxy supposed to correlate all the messages based on the sequence number
and call-id and essentially send an ACK before sending out second INVITE. Or
they are completely different transactions and this is a normal proxy
behavior.

Thanks,

Rajesh.



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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        "Conroy, Lawrence (SMTP)"
	 <lwc@roke.co.uk>
Cc: Chris Boulton <cboulton@ubiquity.net>, Paul Kyzivat
	 <pkyzivat@cisco.com>,
        Matthew.Gardiner@aculab.com, sip@ietf.org
Subject: RE: [Sip] Call transfer using SIP
Date: Thu, 5 Feb 2004 11:27:54 -0000 
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Two points:

-	Standardisation of protocol by information flow is not a good idea.

-	The underlying IETF protocol standards (REFER etc) must be complied with, and with creation of a separate set of standards, this may well be bypassed.

regards

Keith

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 04 February 2004 19:28
> To: Conroy, Lawrence (SMTP)
> Cc: Chris Boulton; Paul Kyzivat; Matthew.Gardiner@aculab.com;
> sip@ietf.org
> Subject: Re: [Sip] Call transfer using SIP
> 
> 
> Mathew asked:
> 
>  > Could somebody please confirm that if I am writing a call control
>  > application offering call hold and transfer then the definitive
>  > document to follow is draft-ietf-sipping-service-examples-05.txt?
> 
> and Lawrence said:
> 
> > Short answer is - yup - it would be a good idea, but not by 
> the IETF.
> 
> So, I might ask, would it be worth encouraging our friends at 
> SIP Forum 
> to publish some "best practices" in this regard?
> 
> --
> Dean
> 
> 
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From exim@www1.ietf.org  Thu Feb  5 09:44:16 2004
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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
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Subject: [Sip] Optionality for History-Info
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I'm in the process of updating History Info
(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-01.txt)
based upon a few comments I've received and merging the requirements into
the solution document.  However, I really need more feedback from the
working group to address the primary point raised during the discussion in
Minneapolis on the optionality.  Initial analysis way long ago resulted in
the decision that Local Policy was the best approach to the optionality,
since it didn't seem reasonable to use Require for an optional header and
lack of the header should not prevent a basic request from being processed.
However, a concern has been raised that Local Policy isn't sufficient as it
requires coordination between clients and proxies (i.e. there's no aspect of
the protocol that allows a client to define or discover that the Proxy will
add the header).  Supported, of course, is used to let the Proxy know that
the information, if available, should be included in the responses.  

In taking a few steps back and re-evaluating all the possibilities for
making the optionality more under the control of the protocol and I came up
with the following list of possibilities. It may not be complete, nor are
the list of pros and cons intended to be exhaustive, but I think it serves
as a useful basis for discussion.

1) Use Require 
Pros: 
- Puts the use of the header explicitly under control of the client
initiating the request. 

Cons: 
- Seems extremely restrictive and functionally limiting, although it would
depend upon the application at the end client that wants to make use of the
information as to how detrimental this might be. 
- Doesn't make sense since the applicability of History-Info isn't
restricted to the initiator of the request. 


2) Make use of the Request-Disposition header field as defined in the caller
preferences draft to explicit request History-Info be included:
history-directive

Pros: 
- Consistent with the intent of that header in that it's a mechanism for a
client to request how a server/proxy should handle a request 

Cons:
- that field isn't extensible 
- doesn't really accomplish more than what can be implied by the Supported
header (i.e. if the client wants the information in a response, then if a
server is capable of adding the header, it SHOULD). 

3) Make use of or define something similar to the Session specific policy,
whereby the client "authorizes" a proxy to capture History-Info and that
it's only captured in that case.   

Pros:
- Explicit.

Cons:
- Seems like a lot of overhead, with little advantage. 

4) Follow the approach of existing optional SIP headers added by
intermediaries (e.g. Path and Reason), which is that it's purely a local
policy decision and doesn't require any coordination with the UA as to
whether it should or shouldn't be added. 

Pros: 
- It's fairly straightforward and consistent with the intent of optional
headers.

Cons: 
- Does require some guidelines to ensure that applications that might use
the optional information are robust enough to handle the lack of
information. 

Feedback on these proposals is appreciated and hopefully this discussion
will allow us to reach a final, constructive conclusion.  

Obviously, my preference is the 4th option and perhaps one reason why we've
gotten so stuck on this point is that the guidelines for applications on the
usage of the header are perhaps not obvious.  So, it might be useful to add
an additional section to the document (or perhaps include additional
information for each of the scenarios in the appendix) on application
guidelines around the usage of the information and default behavior if the
information is not available.

Regards, 
Mary H. Barnes
mary.barnes@nortelnetworks.com
972-684-5432



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From exim@www1.ietf.org  Thu Feb  5 10:01:42 2004
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Thread-Topic: [Sip] Optionality for History-Info
Thread-Index: AcPr9dWPOopRQe/aTnWwYsuUIRbehwAAMNdw
From: "Vijay Gaur" <vgaur@aastra.com>
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Subject: [Sip] Transport parameter in request URI
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Hi,
   My setup is like this:   =20

      UA1----->(TLS)----->Proxy------>(UDP)--->UA2

Here UA1 makes call and dialog gets established between UA1 and UA2. =
Contact of UA2 contains transport parameter tls (Contact: =
sip@ua1.com;transport=3Dtls). Now when UA2 generates BYE it doesn't put =
any transport parameter in request URI (BYE sip:ua1.com SIP/2.0). =
Because of no transport parameter BYE doesn't get 200OK.
Now my question is which transport (TLS or UDP) UA2 should put in =
request URI?
RFC 3261 says that in request message agent should put transport =
parameter in request URI with transport which it is using.
If I put transport=3Dudp in request URI UA2 BYE doesn't get 200OK, but =
in case of transport=3Dtls it works fine.

I appreciate your help.
Thanks,
Vijay

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Subject: RE: [Sip] Transport parameter in request URI
Date: Thu, 5 Feb 2004 09:58:03 -0500
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Thread-Topic: [Sip] Optionality for History-Info
Thread-Index: AcPr9dWPOopRQe/aTnWwYsuUIRbehwAAMNdwAAChV6A=
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Hi,
   My setup is like this:   =20

      UA1----->(TLS)----->Proxy------>(UDP)--->UA2

Here UA1 makes call and dialog gets established between UA1 and UA2. =
Contact of UA1 contains transport parameter tls (Contact: =
sip@ua1.com;transport=3Dtls). Now when UA2 generates BYE it doesn't put =
any transport parameter in request URI (BYE sip:ua1.com SIP/2.0). =
Because of no transport parameter BYE doesn't get 200OK.
Now my question is which transport (TLS or UDP) UA2 should put in =
request URI?
RFC 3261 says that in request message agent should put transport =
parameter in request URI with transport which it is using.
If I put transport=3Dudp in request URI UA2 BYE doesn't get 200OK, but =
in case of transport=3Dtls it works fine.

I appreciate your help.
Thanks,
Vijay
-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Vijay
Gaur
Sent: Thursday, February 05, 2004 9:52 AM
To: sip@ietf.org
Subject: [Sip] Transport parameter in request URI


Hi,
   My setup is like this:   =20

      UA1----->(TLS)----->Proxy------>(UDP)--->UA2

Here UA1 makes call and dialog gets established between UA1 and UA2. =
Contact of UA2 contains transport parameter tls (Contact: =
sip@ua1.com;transport=3Dtls). Now when UA2 generates BYE it doesn't put =
any transport parameter in request URI (BYE sip:ua1.com SIP/2.0). =
Because of no transport parameter BYE doesn't get 200OK.
Now my question is which transport (TLS or UDP) UA2 should put in =
request URI?
RFC 3261 says that in request message agent should put transport =
parameter in request URI with transport which it is using.
If I put transport=3Dudp in request URI UA2 BYE doesn't get 200OK, but =
in case of transport=3Dtls it works fine.

I appreciate your help.
Thanks,
Vijay

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

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



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This is a multi-part message in MIME format.

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Having tried to implement HTTP Authentication for SIP, we faced a few =
problems. This draft tries to highlight those problems and in some cases =
suggests a solution.

Please read and comment.

Thanks,
Hisham

> -----Original Message-----
> From: owner-ietf-announce@ietf.org
> [mailto:owner-ietf-announce@ietf.org]On Behalf Of ext
> Internet-Drafts@ietf.org
> Sent: 05.February.2004 22:42
> Subject: I-D ACTION:draft-khartabil-sip-auth-analysis-00.txt
>=20
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
>=20
>=20
> 	Title		: Analysis: HTTP Authentication for SIP
> 	Author(s)	: H. Khartabil
> 	Filename	: draft-khartabil-sip-auth-analysis-00.txt
> 	Pages		: 9
> 	Date		: 2004-2-5
> =09
> The Session Initiation Protocol (SIP) provides a stateless,
>    challenge-based mechanism for  authentication that is based on
>    authentication in HTTP. RFC 3261 fails to indicate the behaviour of
>    SIP intermediaries and User Agents in certain scenarios. This
>    documents presents such scenarios, analysis the currently available
>    text, and where possible, offers a solution.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-khartabil-sip-auth-a
nalysis-00.txt

To remove yourself from the IETF Announcement list, send a message to=20
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message.

Internet-Drafts are also available by anonymous FTP. Login with the =
username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-khartabil-sip-auth-analysis-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html=20
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-khartabil-sip-auth-analysis-00.txt".
=09
NOTE:	The mail server at ietf.org can return the document in
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_______________________________________________
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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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Thread-Topic: [Sipping] Conference policy URI
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This will most likely be a SIP WG item.

/Hisham

> -----Original Message-----
> From: sipping-admin@ietf.org=20
> [mailto:sipping-admin@ietf.org]On Behalf Of
> ext hisham.khartabil@nokia.com
> Sent: 06.February.2004 09:52
> To: jdrosen@dynamicsoft.com
> Cc: sipping@ietf.org
> Subject: RE: [Sipping] Conference policy URI
>=20
>=20
> Thanks Jonathan. A draft has been submitted to the SIP WG.
>=20
> http://www.ietf.org/internet-drafts/draft-khartabil-sip-policy
> -uri-call-info-purpose-00.txt
>=20
> Regards,
> Hisham
>=20
> > -----Original Message-----
> > From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: 24.January.2004 04:44
> > To: Khartabil Hisham (Nokia-TP/Helsinki)
> > Cc: sipping@ietf.org
> > Subject: Re: [Sipping] Conference policy URI
> >=20
> >=20
> > I think this is a reasonable usage of Call-Info. However,=20
> it can only=20
> > be done if a "purpose" tag value is registered with IANA, as=20
> > indicated=20
> > by section 20.9. However, it ocurred to me, when reading=20
> > this, that we=20
> > never set up an IANA registry for call-info purpose tokens! Oops!=20
> > Thats easily rectifiable; when the draft defining the=20
> > 'policy-uri' (or=20
> > whatever its called) purpose tag is written, it should also=20
> > create the=20
> > registry and register the other tokens defined in rfc3261.
> >=20
> > -Jonathan R.
> >=20
> > hisham.khartabil@nokia.com wrote:
> >=20
> > > Hi,
> > >=20
> > > There was consensus on the XCON mailing list that the conference
> > > state event package needs to include in the notification the
> > > conference policy URI.
> > >=20
> > > There was also consensus that some participants may not want to
> > > subscribe to the conference package but would still like to learn
> > > the conference policy URI. In this case, it was suggested that the
> > > Call-info header carries such information. Perhaps the
> > > cc-conferencing draft can include examples of that.
> > >=20
> > > Thanks, Hisham
> > >=20
> > > _______________________________________________ Sipping mailing
> > > list  https://www1.ietf.org/mailman/listinfo/sipping This list is
> > > for NEW development of the application of SIP Use
> > > sip-implementors@cs.columbia.edu for questions on current sip Use
> > > sip@ietf.org for new developments of core SIP
> > >=20
> >=20
> > --=20
> > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > Chief Technology Officer                    Parsippany, NJ=20
> 07054-2711
> > dynamicsoft
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> >=20
> >=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Is there any progres in this area? I dont see any discussions after =
draft-miller-sip-tcap-00.txt released..

Thanks,
K.S. Keshava Murthy
Technical Architect

IntelliNet Technologies
Foundations for Global Networks

Phone +91 80 5125 6018 x 147
Web www.intellinet-tech.com=20

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IntelliNet Technologies India Pvt Ltd
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<DIV><FONT face=3DArial size=3D2>Is there any progres in this area? I =
dont see any=20
discussions after draft-miller-sip-tcap-00.txt released..</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>K.S. Keshava Murthy<BR>Technical=20
Architect</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>IntelliNet Technologies<BR>Foundations =
for Global=20
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<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Phone&nbsp;+91 80 5125 6018 x =
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</FONT></DIV>
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Subject: [Sip] Some details in RFC 3398
Date: Wed, 4 Feb 2004 14:44:40 +0100
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Thread-Topic: [Sip] Some details in RFC 3398
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From: "El Moussawi, Ali" <ali.el-moussawi@hp.com>
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Hi all,

I have two tiny questions concerning RFC3398.

In  7.1.6 , what do you think the cause code should be in the REL sent
to the PSTN network after interwork tinmer expires ? =20

On the other hand, in 8.2.3, it says that upon receipt of a SIP 181
response, the gateway should send an early ACM to the PSTN side, plus a
CPG with event code 6. What is the use of CPG since the other side will
already know that its call is beying handled when it receives a early
ACM ? Is the CPG mandatory ?

Thanks a lot=20


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<BODY>
<DIV><SPAN class=3D484473713-04022004><FONT =
size=3D2>Hi&nbsp;all,</FONT></DIV>
<DIV>
<P><FONT size=3D2><SPAN class=3D484473713-04022004>I have two tiny =
questions=20
concerning RFC3398.</SPAN></FONT></P>
<P><FONT size=3D2>In&nbsp;<SPAN class=3D484473713-04022004><FONT =
face=3DArial=20
color=3D#0000ff>&nbsp;7.1.6&nbsp;</FONT></SPAN>, what do you think the =
cause code=20
should be in the REL sent to the PSTN network after interwork tinmer =
expires=20
?&nbsp;<SPAN class=3D484473713-04022004><FONT face=3DArial=20
color=3D#0000ff>&nbsp;</FONT></SPAN></FONT></P>
<P><FONT size=3D2><SPAN class=3D484473713-04022004><FONT face=3DArial =
color=3D#0000ff>On=20
the other hand, in 8.2.3, it says that upon receipt of&nbsp;a SIP 181 =
response,=20
the gateway should&nbsp;send an early ACM to the&nbsp;PSTN side, plus a =
CPG with=20
event code 6. What is the use of CPG since the other side will already =
know that=20
its call is beying handled when it receives a early ACM ? Is the CPG =
mandatory=20
?</FONT></SPAN></FONT></P>
<P><FONT size=3D2><SPAN class=3D484473713-04022004><FONT face=3DArial=20
color=3D#0000ff>Thanks a=20
lot</FONT>&nbsp;</SPAN></FONT></P></SPAN></DIV></BODY></HTML>
=00
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Reply-To: <marco.pissarello@fastwebnet.it>
From: "Marco Pissarello" <marco.pissarello@agsmtel.it>
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Date: Thu, 5 Feb 2004 07:43:34 +0100
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Ing. Marco Pissarello

mobile: +39 3484786881
mobile: +39 3486090637
phone:  +39 0454856103
mail:   marco.pissarello@fastwebnet.it

-----Messaggio originale-----
Da: sip-admin@ietf.org [mailto:sip-admin@ietf.org] Per conto di
sip-request@ietf.org
Inviato: mercoled=EC 4 febbraio 2004 18.00
A: sip@ietf.org
Oggetto: Sip digest, Vol 1 #1250 - 9 msgs

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Today's Topics:

   1. draft-ietf-sip-gruu-00 (Orit Levin)
   2. Re: draft-ietf-sip-gruu-00 (Dean Willis)
   3. Re: I-D ACTION:draft-ietf-sip-publish-02.txt (Jonathan Rosenberg)
   4. Re: draft-ietf-sip-gruu-00 (Jonathan Rosenberg)
   5. Call transfer using SIP (Matthew.Gardiner@aculab.com)
   6. Re: Call transfer using SIP (Paul Kyzivat)
   7. Re: Call transfer using SIP (Conroy, Lawrence (SMTP))
   8. RE: Call transfer using SIP (Matthew.Gardiner@aculab.com)
   9. RE: Call transfer using SIP (Chris Boulton)

--__--__--

Message: 1
Date: Tue, 3 Feb 2004 14:58:25 -0800
From: "Orit Levin" <oritl@microsoft.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>, <sip@ietf.org>
Subject: [Sip] draft-ietf-sip-gruu-00

I have the following remarks regarding the GRUU draft:

1. The draft states that one of the GRUU properties is =3D20
"   o It MUST NOT be possible, based on inspection of the URI, to
determine the associated Contact URI or Address of Record."

It MUST be possible to achieve such a property for a GRUU (if the
Registrar wants to enforce such a policy for its domain), but it is not
part of the requirements for the GRUU concept to always enforce such a
policy.

2. It is expressed throughout the draft that=3D20
"This extension allows a UA to obtain a GRUU, and to use a GRUU. These
two mechanisms are separate, in that a UA can obtain a GRUU in any way
it likes, and use the mechanisms in this specification to use them.
Similarly, a UA can obtain a GRUU but never use it."

For consistency, I suggest in section "6.2 Using the GRUU" change the
first sentence from

   "A UA first obtains a GRUU using the procedures of Section 6.1."
to=3D20
   "A UA first obtains a GRUU using the procedures of Section 6.1 or by
other means".

Thanks,
Orit.

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
Jonathan Rosenberg
Sent: Monday, February 02, 2004 10:54 PM
To: Christian Jansson
Cc: Paul Kyzivat; Jussi.Turunen@nokia.com;
nataraju.alilaghatta@wipro.com; sanjsinh@cisco.com; sip@ietf.org
Subject: Re: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and
target refresh request



Christian Jansson wrote:

> Perhaps a little late, and hopefully not off topic. Comments inline.

Not too late, not off topic.

>=3D20
>=3D20
>>Jonathan R. wrote
>>This, however, leads me into thinking a bit on the lifecycle
>>management for GRUUs. Currently, GRUU is bound to a single=3D20
>>registration, where the registration is defined by the Contact URI. If
>=3D20
>=3D20
>>the UA should "move", in the sense that it acquires a new IP =
address,=3D20
>>this will be a new Contact URI, and therefore, a new GRUU. It would be
>>nice if, instead, the GRUU were truly bound to the UA instance, =
and=3D20
>>that even if the UA instance changed IP addresses, the GRUU could=3D20
>>remain unchanged. In such a case, if the UA moves, there is =
actually=3D20
>>no need to issue any target refresh requests. The UA would simply=3D20
>>re-register, update the binding of its GRUU to its new IP=3D20
>>address, and=3D20
>>existing calls would be correctly routed.
>=3D20
>=3D20
> But if the ip actually changes, it would be necessary to send an
updated
> SDP with the new ip in the c=3D3D line.

Yes, that is true. If media relays are used (i.e., TURN), it might =
be=3D20
possible to avoid even this, but lets leave that aside for now.

> So your idea won't save you from
> having to send a SIP request to the other end every time the ip
changes
> in an INVITE session.=3D20

Right. However, it does solve the problem that, previously, if both=3D20
endpoints moved at the same time, the call would fail - neither =
side=3D20
could signal the other.


> For a non-INVITE session though the idea would
> save a lot of request from being sent if the session only contains
> signaling.

Right, in particular, a long lived presence subscription, for example.

-Jonathan R.

--=3D20
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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--__--__--

Message: 2
Date: Tue, 03 Feb 2004 17:58:47 -0600
From: Dean Willis <dean.willis@softarmor.com>
To: Orit Levin <oritl@microsoft.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: Re: [Sip] draft-ietf-sip-gruu-00

Orit Levin wrote:
> I have the following remarks regarding the GRUU draft:
>=20
> 1. The draft states that one of the GRUU properties is =20
> "   o It MUST NOT be possible, based on inspection of the URI, to
> determine the associated Contact URI or Address of Record."
>=20
> It MUST be possible to achieve such a property for a GRUU (if the
> Registrar wants to enforce such a policy for its domain), but it is =
not
> part of the requirements for the GRUU concept to always enforce such a
> policy.

I have to admit I concur with this position. There might be very useful=20
situations where the anonymity aspect of the GRUU would be undesirable.

I can even envision models where the GRUU is cryptographically=20
validatable as representing an instance associated with a specific AOR,=20
including variations where the validation can or cannot be done without=20
prior knowledge of the associated AOR.

As far as I can tell, reversibility does not impact the given rationale=20
for the requirements, that:

    With these rules, it is possible, though not required, to construct =
a
    GRUU without requiring the maintenance of any additional state. To =
do
    that, the URI would be constructed in the following fashion:

So I believe I would support making this a SHOULD strength requirement.=20
Conceivably some discussion of the security implications of NOT meeting=20
this requirement would be appropriate.

--
Dean





--__--__--

Message: 3
Date: Wed, 04 Feb 2004 00:54:24 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
To: aki.niemi@nokia.com
CC: sip@ietf.org
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-publish-02.txt



aki.niemi@nokia.com wrote:

> Hi Jonathan,
>=20
> Thanks a lot for the excellent comments. Most of them I will directly
> fold into -03. Few clarifications below.
>=20
> Cheers, Aki
>=20
> Jonathan Rosenberg wrote:
>=20
> <snip />
>=20
>> * section 3, paragraph 3:
>>=20
>>> That is, the steady-state of this event package in the absence
>>> of, or in addition
>> to, soft state
>>> provided through the PUBLISH mechanism
>>=20
>> this makes it sound like steady state is a function of the event=20
>> packae only. the hard state is a function of the resource uRI and=20
>> event package. Indeed, one may think of an event package as merely
>>  sub-resource within the full resource. I wonder if it wouldn't be
>>  useful to come up with a term for this - for example, "scoped=20
>> resource" is a combination of a URI and event package, and=20
>> identifies a particular piece of event state.
>=20
> I don't quite follow,  but I agree the quoted text is rather vague.
> To me the hard state, or "preconfigured" state is just another source
> of event state to the composer. The mechanism to set that state might
> be other than PUBLISH.

Right. I agree with that. Let me try again. Requoting the paragraph:

    PUBLISH requests create soft state in the ESC. This state has a
    defined lifetime and will expire after a negotiated amount of time,
    requiring the publication to be refreshed by subsequent PUBLISH
    requests. Local policy at the compositor may in turn define hard
    state for a particular event package. That is, the steady-state of
    this event package in the absence of, or in addition to, soft state
    provided through the PUBLISH mechanism. Setting this hard state or
    configuring the composer policy is out of the scope of this
    specification.

The words "hard state for a particular event package" read to me like it =

meant that hard state was defined on a package by package basis. But,=20
its not. There would be hard state generally for a particular=20
URI/package basis. So, I'd suggest:

     There may also be hard state provisioned for each resource for
     a particular event package. This hard state represents the resource
     state that is present at all times, and does not expire.


My other comment was that the concept of "for each resource for each=20
package" is a common concept, for which I was looking for a term to =
define.

>=20
> <snip />
>=20
>> * section 5.1:
>>=20
>>> As with any other SIP message, the PUBLISH mechanism MAY use the=20
>>> content indirection mechanism defined in [10].
>>=20
>> this statement makes [10] a normative reference, I believe. It is=20
>> listed as informative in the references section.
>=20
> I'm not sure I agree. I've understood the RFC editor's policy so that
> an informative reference is not required for implementing the RFC.
> It's an interesting question as to how this relates to RFC 2119
> terminology. But I think a MAY makes the feature optional so that it
> fits the informative reference definition.

No - Keith is right in his note. Think of it this way - would the=20
feature interop if, one day, the referenced I-D evaporated and people=20
implemented it differently? In this case, no. If they chose to implement =

the content indirection, and there was no spec, they would do it=20
differently and there would be no interop.

I would just drop the text to avoid the dependency.

>=20
> <snip />
>=20
>> * section 5.3 is out of place. Section 5 is about generating=20
>> requests. the subsections for the most part deal with the specific
>> 4 sub-cases you identified in 5.1. However, 5.3 talks about setting
>> expirations, which is something that is actually done for the cases
>> in 5.2, 5.4, 5.5 and 5.6. I would suggest that eaech of those four
>>  sections rather describe how the Expires header field is
>> constructed for each case. That is consistent with having a column
>> in table 1 on expires. Indeed, each section should also talk about
>> bodies and etag. Along those lines, section 5.2 does not actually
>> say that a body has to be present; it shouold be stated given the
>> importance of this fact from table 1.
>=20
> I think the reason for this is that I envisioned a PUBLISH carrying
> state in some other place than the body. However, I don't know if
> such a case would ever exist, and other people have also commented on
> this. I think it is safe to assume that the state is always carried
> in the body, as opposed to "typically".

Yes.


>> * the steps in section 6 omit handling for the case where there was
>> no SIP-If-Match header field. step 6 says to check for it, and all
>> the following steps assume it was there. I would suggest adding a
>> bullet to step 6 which says, if there was no SIP-If-Match header
>> field, the ESC creates a new etag, and associates it with as-yet
>> empty event state. Processing continues in the steps below. This
>> way, you always exit step 6 with an etag.
>=20
> The attempt was to make etag generation part of step 9. This way, an
> entity-tag is generated for each successful publication and returned
> in 2xx, irrespective of whether the publication was an initial
> publication, a refresh, or a modification. Indeed, a 2xx to a removal
> could be treated the same, although such an entity-tag is useless at
> that point.
>=20
> So I think it is ok to exit step 6 without an entity tag, as long as
> step 9 is clear that entity-tags are generated for all successful
> publications.
>=20

Step 8 requires an entity tag to be defined, however.


>> * sectiton 11.1:
>>=20
>>> Authentication issues are discussed in SIP [2]. The exact
>> methods for
>>> creation and manipulation of the ESC authorization policies are=20
>>> outside the scope of this document.
>>=20
>> you will get dinged on this by Bellovin, guaranteed. He wants to
>> see statements about baseline mandtory to implement mechahnisms.=20
>> Its fine to say that this mechanism is digest, as mandated by
>> rfc3261.
>>=20
>>> To prevent replay attacks, implementations SHOULD require=20
>>> authentication with anti-replay protection.
>> Authentication issues are
>>> discussed in SIP [2].
>>=20
>> as above.... does this mean digest with next-nonce or are you
>> talking sips?
>>=20
>> and section 11.4:
>>=20
>>> To prevent such attacks, implementations SHOULD, at a minimum,=20
>>> provide integrity protection across the To, From, Event,=20
>>> SIP-If-Match, Route, and Expires headers and the bodies
>> of PUBLISH
>>> requests.
>>=20
>> as above, he will ask for a mechanism. This requiurement makes the
>>  choice sips.
>>=20
>> Same in 11.5.
>=20
> I see what you are saying. But is it really so that each new method
> that gets added to SIP needs to explicitly mandate certain security
> mechanisms? At least going beyond the security considerations in RFC
> 3261 seems unnecessary, since the properties of PUBLISH are very
> similar to, say, REGISTER.

I have made this same comment in past discussions of specs going through =

iesg. However, the security ADs believe strongly in reptition of=20
security requirements in specifications. Trust me, you WILL get dinged=20
on this. It is much easier to just add text stating the same normative=20
requiremetns as in rfc3261, then to get a discuss during iesg =
evaluation.

-Jonathan R.

--=20
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


--__--__--

Message: 4
Date: Wed, 04 Feb 2004 01:03:16 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
To: Dean Willis <dean.willis@softarmor.com>
CC: Orit Levin <oritl@microsoft.com>, sip@ietf.org
Subject: Re: [Sip] draft-ietf-sip-gruu-00

inline.

Dean Willis wrote:

> Orit Levin wrote:
>=20
>> I have the following remarks regarding the GRUU draft:
>>
>> 1. The draft states that one of the GRUU properties is  "   o It MUST =

>> NOT be possible, based on inspection of the URI, to
>> determine the associated Contact URI or Address of Record."
>>
>> It MUST be possible to achieve such a property for a GRUU (if the
>> Registrar wants to enforce such a policy for its domain), but it is =
not
>> part of the requirements for the GRUU concept to always enforce such =
a
>> policy.
>=20
>=20
> I have to admit I concur with this position. There might be very =
useful=20
> situations where the anonymity aspect of the GRUU would be =
undesirable.
>=20
> I can even envision models where the GRUU is cryptographically=20
> validatable as representing an instance associated with a specific =
AOR,=20
> including variations where the validation can or cannot be done =
without=20
> prior knowledge of the associated AOR.
>=20
> As far as I can tell, reversibility does not impact the given =
rationale=20
> for the requirements, that:
>=20
>    With these rules, it is possible, though not required, to construct =
a
>    GRUU without requiring the maintenance of any additional state. To =
do
>    that, the URI would be constructed in the following fashion:
>=20
> So I believe I would support making this a SHOULD strength =
requirement.=20
> Conceivably some discussion of the security implications of NOT =
meeting=20
> this requirement would be appropriate.

I am ok to lessen the requirement.

The question becomes, do we want a way to allow the client to signal=20
that a randomized gruu is desired, or is it up to the server? I would=20
much prefer to leave it up to the server, in which case the only thing=20
that we need to change is the requirement, above. I changed the text to=20
read:

<t>In many cases, it will be desirable to construct the GRUU in such a
way that it will not be possible, based on inspection of the URI, to
determine the associated Contact URI or Address of Record. Whether or
not a GRUU should be constructed with this property is a local policy
decision of the administrator.
</t>


Orit writes:
> 2. It is expressed throughout the draft that=20
> "This extension allows a UA to obtain a GRUU, and to use a GRUU. These
> two mechanisms are separate, in that a UA can obtain a GRUU in any way
> it likes, and use the mechanisms in this specification to use them.
> Similarly, a UA can obtain a GRUU but never use it."
>=20
> For consistency, I suggest in section "6.2 Using the GRUU" change the
> first sentence from
>=20
>    "A UA first obtains a GRUU using the procedures of Section 6.1."
> to=20
>    "A UA first obtains a GRUU using the procedures of Section 6.1 or =
by
> other means".

OK, with the addition of "outside the scope of this specification.".

-Jonathan R.



--=20
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


--__--__--

Message: 5
From: Matthew.Gardiner@aculab.com
To: sip@ietf.org
Date: Wed, 4 Feb 2004 10:04:49 -0000=20
Subject: [Sip] Call transfer using SIP

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Could somebody please confirm that if I am writing a call control
application offering call hold and transfer then the definitive document =
to
follow is draft-ietf-sipping-service-examples-05.txt?

Matthew Gardiner
Software Engineer
Aculab=20
Tel: +44 (0) 1908 273 911
Fax: +44 (0) 1908 273 801
Email: mailto:matthew.gardiner@aculab.com
Website: <http://www.aculab.com>




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<P><FONT SIZE=3D3D2 FACE=3D3D"Arial">Could somebody please confirm that =
if =3D
I am writing a call control application offering call hold and transfer =
=3D
then the definitive document to follow is =3D
draft-ietf-sipping-service-examples-05.txt?</FONT></P>

<P><FONT SIZE=3D3D2 FACE=3D3D"Arial">Matthew Gardiner</FONT>
<BR><FONT SIZE=3D3D2 FACE=3D3D"Arial">Software Engineer<BR>
</FONT><FONT SIZE=3D3D2 FACE=3D3D"Arial">Aculab</FONT><FONT SIZE=3D3D2 =
=3D
FACE=3D3D"Times New Roman"><BR>
</FONT><FONT SIZE=3D3D2 FACE=3D3D"Arial">Tel: +44 (0) 1908 273 =3D
911</FONT><BR>
<FONT SIZE=3D3D2 FACE=3D3D"Arial">Fax: +44 (0) 1908 273 801</FONT>
<BR><FONT SIZE=3D3D2 FACE=3D3D"Arial">Email: <A =3D
HREF=3D3D"mailto:matthew.gardiner@aculab.com">mailto:matthew.gardiner@acu=
l=3D
ab.com</A></FONT>
<BR><FONT SIZE=3D3D2 FACE=3D3D"Arial">Website:<U> </U></FONT><U><FONT =
=3D
COLOR=3D3D"#0000FF" SIZE=3D3D2 FACE=3D3D"Arial">&lt;<A =3D
HREF=3D3D"http://www.aculab.com" =3D
TARGET=3D3D"_blank">http://www.aculab.com</A>&gt;</FONT></U>
</P>
<BR>
<BR>
<BR>

<P><B><FONT SIZE=3D3D1 FACE=3D3D"Arial">For Aculab's privacy policy and =
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--__--__--

Message: 6
Date: Wed, 04 Feb 2004 10:07:14 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
To: Matthew.Gardiner@aculab.com
CC: sip@ietf.org
Subject: Re: [Sip] Call transfer using SIP



Matthew.Gardiner@aculab.com wrote:
> Could somebody please confirm that if I am writing a call control=20
> application offering call hold and transfer then the definitive =
document=20
> to follow is draft-ietf-sipping-service-examples-05.txt?

No. The *definitive* document is RFC3261 and related documents. The=20
details of call control are derivative from that in somewhat the same=20
way the much of physics is derivative from F=3DMA.

Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) an=20
*example*. It is not normative.

However, that draft is certainly helpful and educational. You would do=20
well to understand what is in there before completing your task.

	Paul



--__--__--

Message: 7
From: "Conroy, Lawrence (SMTP)" <lwc@roke.co.uk>
To: Paul Kyzivat <pkyzivat@cisco.com>
Cc: Matthew.Gardiner@aculab.com, sip@ietf.org
Subject: Re: [Sip] Call transfer using SIP
Date: Wed, 4 Feb 2004 16:09:52 +0000

Hi folks,
   It's an interesting point.
Paul's comment seems to be "there are many different ways of skinning a=20
cat, and the
only normative definition is of the knife".
The IETF defines the details of knife behaviour, and it's up to=20
everyone else to sort
out how to use it for their purpose.

My understanding is that the IETF (and the SIP WG in particular) will=20
not produce normative
definitions of these (or any other) teleservices, only definitions of=20
protocol elements that
might be used for these (or other) purposes.
SIPPING will consider such services and decide on the protocol elements=20
they think are needed
(that don't exist already) and then SIP grinds through the details.

Thus the only teleservice examples are in the service-examples document=20
(collected together
as a convenience), these are non-normative, and there will never BE=20
normative teleservice
definitions produced by these groups. Is this correct?

all the best,
   Lawrence

On 4 Feb 2004, at 3:07 pm, Paul Kyzivat wrote:

>
>
> Matthew.Gardiner@aculab.com wrote:
>> Could somebody please confirm that if I am writing a call control=20
>> application offering call hold and transfer then the definitive=20
>> document to follow is draft-ietf-sipping-service-examples-05.txt?
>
> No. The *definitive* document is RFC3261 and related documents. The=20
> details of call control are derivative from that in somewhat the same=20
> way the much of physics is derivative from F=3DMA.
>
> Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) an=20
> *example*. It is not normative.
>
> However, that draft is certainly helpful and educational. You would do =

> well to understand what is in there before completing your task.
>
> 	Paul
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


--__--__--

Message: 8
From: Matthew.Gardiner@aculab.com
To: lwc@roke.co.uk, pkyzivat@cisco.com
Cc: sip@ietf.org
Subject: RE: [Sip] Call transfer using SIP
Date: Wed, 4 Feb 2004 16:24:01 -0000=20

This message is in MIME format. Since your mail reader does not =
understand
this format, some or all of this message may not be legible.

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This vagueness makes it very difficult to engineer a telephony solution
which interoperates with other vendors equipment. In the telephony arena
getting call transfer right is of fundamental importance. A simple =
document
titled "Call transfer in SIP" would suffice in de-mystifying this area.

-----Original Message-----
From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
Sent: 04 February 2004 16:10
To: Paul Kyzivat
Cc: Matthew.Gardiner@aculab.com; sip@ietf.org
Subject: Re: [Sip] Call transfer using SIP


Hi folks,
   It's an interesting point.
Paul's comment seems to be "there are many different ways of skinning a=20
cat, and the
only normative definition is of the knife".
The IETF defines the details of knife behaviour, and it's up to=20
everyone else to sort
out how to use it for their purpose.

My understanding is that the IETF (and the SIP WG in particular) will=20
not produce normative
definitions of these (or any other) teleservices, only definitions of=20
protocol elements that
might be used for these (or other) purposes.
SIPPING will consider such services and decide on the protocol elements=20
they think are needed
(that don't exist already) and then SIP grinds through the details.

Thus the only teleservice examples are in the service-examples document=20
(collected together
as a convenience), these are non-normative, and there will never BE=20
normative teleservice
definitions produced by these groups. Is this correct?

all the best,
   Lawrence

On 4 Feb 2004, at 3:07 pm, Paul Kyzivat wrote:

>
>
> Matthew.Gardiner@aculab.com wrote:
>> Could somebody please confirm that if I am writing a call control=20
>> application offering call hold and transfer then the definitive=20
>> document to follow is draft-ietf-sipping-service-examples-05.txt?
>
> No. The *definitive* document is RFC3261 and related documents. The=20
> details of call control are derivative from that in somewhat the same=20
> way the much of physics is derivative from F=3DMA.
>
> Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) an=20
> *example*. It is not normative.
>
> However, that draft is certainly helpful and educational. You would do =

> well to understand what is in there before completing your task.
>
> 	Paul
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

--=20
Registered Office: Roke Manor Research Ltd, Siemens House, Oldbury,
Bracknell,
Berkshire. RG12 8FZ

The information contained in this e-mail and any attachments is =
confidential
to
Roke Manor Research Ltd and must not be passed to any third party =
without
permission. This communication is for information only and shall not =
create
or
change any contractual relationship.


For Aculab's privacy policy and E-Mail disclaimer, Click Here :
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<P><FONT SIZE=3D3D2>This vagueness makes it very difficult to engineer a =
=3D
telephony solution which interoperates with other vendors equipment. In =
=3D
the telephony arena getting call transfer right is of fundamental =3D
importance. A simple document titled &quot;Call transfer in SIP&quot; =
=3D
would suffice in de-mystifying this area.</FONT></P>

<P><FONT SIZE=3D3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D3D2>From: Conroy, Lawrence (SMTP) [<A =3D
HREF=3D3D"mailto:lwc@roke.co.uk">mailto:lwc@roke.co.uk</A>]</FONT>
<BR><FONT SIZE=3D3D2>Sent: 04 February 2004 16:10</FONT>
<BR><FONT SIZE=3D3D2>To: Paul Kyzivat</FONT>
<BR><FONT SIZE=3D3D2>Cc: Matthew.Gardiner@aculab.com; =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D3D2>Subject: Re: [Sip] Call transfer using SIP</FONT>
</P>
<BR>

<P><FONT SIZE=3D3D2>Hi folks,</FONT>
<BR><FONT SIZE=3D3D2>&nbsp;&nbsp; It's an interesting point.</FONT>
<BR><FONT SIZE=3D3D2>Paul's comment seems to be &quot;there are many =3D
different ways of skinning a </FONT>
<BR><FONT SIZE=3D3D2>cat, and the</FONT>
<BR><FONT SIZE=3D3D2>only normative definition is of the =3D
knife&quot;.</FONT>
<BR><FONT SIZE=3D3D2>The IETF defines the details of knife behaviour, =
and =3D
it's up to </FONT>
<BR><FONT SIZE=3D3D2>everyone else to sort</FONT>
<BR><FONT SIZE=3D3D2>out how to use it for their purpose.</FONT>
</P>

<P><FONT SIZE=3D3D2>My understanding is that the IETF (and the SIP WG in =
=3D
particular) will </FONT>
<BR><FONT SIZE=3D3D2>not produce normative</FONT>
<BR><FONT SIZE=3D3D2>definitions of these (or any other) teleservices, =
=3D
only definitions of </FONT>
<BR><FONT SIZE=3D3D2>protocol elements that</FONT>
<BR><FONT SIZE=3D3D2>might be used for these (or other) purposes.</FONT>
<BR><FONT SIZE=3D3D2>SIPPING will consider such services and decide on =
=3D
the protocol elements </FONT>
<BR><FONT SIZE=3D3D2>they think are needed</FONT>
<BR><FONT SIZE=3D3D2>(that don't exist already) and then SIP grinds =3D
through the details.</FONT>
</P>

<P><FONT SIZE=3D3D2>Thus the only teleservice examples are in the =3D
service-examples document </FONT>
<BR><FONT SIZE=3D3D2>(collected together</FONT>
<BR><FONT SIZE=3D3D2>as a convenience), these are non-normative, and =3D
there will never BE </FONT>
<BR><FONT SIZE=3D3D2>normative teleservice</FONT>
<BR><FONT SIZE=3D3D2>definitions produced by these groups. Is this =3D
correct?</FONT>
</P>

<P><FONT SIZE=3D3D2>all the best,</FONT>
<BR><FONT SIZE=3D3D2>&nbsp;&nbsp; Lawrence</FONT>
</P>

<P><FONT SIZE=3D3D2>On 4 Feb 2004, at 3:07 pm, Paul Kyzivat =
wrote:</FONT>
</P>

<P><FONT SIZE=3D3D2>&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; Matthew.Gardiner@aculab.com wrote:</FONT>
<BR><FONT SIZE=3D3D2>&gt;&gt; Could somebody please confirm that if I am =
=3D
writing a call control </FONT>
<BR><FONT SIZE=3D3D2>&gt;&gt; application offering call hold and =
transfer =3D
then the definitive </FONT>
<BR><FONT SIZE=3D3D2>&gt;&gt; document to follow is =3D
draft-ietf-sipping-service-examples-05.txt?</FONT>
<BR><FONT SIZE=3D3D2>&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; No. The *definitive* document is RFC3261 and =
=3D
related documents. The </FONT>
<BR><FONT SIZE=3D3D2>&gt; details of call control are derivative from =
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Message: 9
Subject: RE: [Sip] Call transfer using SIP
Date: Wed, 4 Feb 2004 16:35:00 -0000
From: "Chris Boulton" <cboulton@ubiquity.net>
To: "Conroy, Lawrence (SMTP)" <lwc@roke.co.uk>,
        "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: <Matthew.Gardiner@aculab.com>, <sip@ietf.org>

Lawrence - remember that SIP is a transport protocol which allows an
objective to be reached in many differing ways.  There is no correct
behavior in many scenarios as long as they fall between the guidelines
set out.  It would be wrong to define normative behavior for such
scenarios and so a push in the right direction towards preferred methods
is the best approach.

Chris.


>-----Original Message-----
>From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
>Sent: 04 February 2004 16:10
>To: Paul Kyzivat
>Cc: Matthew.Gardiner@aculab.com; sip@ietf.org
>Subject: Re: [Sip] Call transfer using SIP
>
>Hi folks,
>   It's an interesting point.
>Paul's comment seems to be "there are many different ways of skinning a
>cat, and the
>only normative definition is of the knife".
>The IETF defines the details of knife behaviour, and it's up to
>everyone else to sort
>out how to use it for their purpose.
>
>My understanding is that the IETF (and the SIP WG in particular) will
>not produce normative
>definitions of these (or any other) teleservices, only definitions of
>protocol elements that
>might be used for these (or other) purposes.
>SIPPING will consider such services and decide on the protocol elements
>they think are needed
>(that don't exist already) and then SIP grinds through the details.
>
>Thus the only teleservice examples are in the service-examples document
>(collected together
>as a convenience), these are non-normative, and there will never BE
>normative teleservice
>definitions produced by these groups. Is this correct?
>
>all the best,
>   Lawrence
>
>On 4 Feb 2004, at 3:07 pm, Paul Kyzivat wrote:
>
>>
>>
>> Matthew.Gardiner@aculab.com wrote:
>>> Could somebody please confirm that if I am writing a call control
>>> application offering call hold and transfer then the definitive
>>> document to follow is draft-ietf-sipping-service-examples-05.txt?
>>
>> No. The *definitive* document is RFC3261 and related documents. The
>> details of call control are derivative from that in somewhat the same
>> way the much of physics is derivative from F=3D3DMA.
>>
>> Draft-ietf-sipping-service-examples-05.txt is (not surprisingly) an
>> *example*. It is not normative.
>>
>> However, that draft is certainly helpful and educational. You would
do
>> well to understand what is in there before completing your task.
>>
>> 	Paul
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


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From exim@www1.ietf.org  Fri Feb  6 06:14:34 2004
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Date: Fri, 06 Feb 2004 11:09:36 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Optionality for History-Info
To: "'Mary Barnes'" <mary.barnes@nortelnetworks.com>,
        "'sip@ietf.org'" <sip@ietf.org>
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Mary,

I too am happy with option 4.

Regards,

John (john.elwell@siemens.com)


> -----Original Message-----
> From: Mary Barnes [mailto:mary.barnes@nortelnetworks.com] 
> Sent: 05 February 2004 14:39
> To: 'sip@ietf.org'
> Subject: [Sip] Optionality for History-Info
> 
> 
> I'm in the process of updating History Info
> (http://www.ietf.org/internet-drafts/draft-ietf-sip-history-in
> fo-01.txt)
> based upon a few comments I've received and merging the 
> requirements into
> the solution document.  However, I really need more feedback from the
> working group to address the primary point raised during the 
> discussion in
> Minneapolis on the optionality.  Initial analysis way long 
> ago resulted in
> the decision that Local Policy was the best approach to the 
> optionality,
> since it didn't seem reasonable to use Require for an 
> optional header and
> lack of the header should not prevent a basic request from 
> being processed.
> However, a concern has been raised that Local Policy isn't 
> sufficient as it
> requires coordination between clients and proxies (i.e. 
> there's no aspect of
> the protocol that allows a client to define or discover that 
> the Proxy will
> add the header).  Supported, of course, is used to let the 
> Proxy know that
> the information, if available, should be included in the responses.  
> 
> In taking a few steps back and re-evaluating all the possibilities for
> making the optionality more under the control of the protocol 
> and I came up
> with the following list of possibilities. It may not be 
> complete, nor are
> the list of pros and cons intended to be exhaustive, but I 
> think it serves
> as a useful basis for discussion.
> 
> 1) Use Require 
> Pros: 
> - Puts the use of the header explicitly under control of the client
> initiating the request. 
> 
> Cons: 
> - Seems extremely restrictive and functionally limiting, 
> although it would
> depend upon the application at the end client that wants to 
> make use of the
> information as to how detrimental this might be. 
> - Doesn't make sense since the applicability of History-Info isn't
> restricted to the initiator of the request. 
> 
> 
> 2) Make use of the Request-Disposition header field as 
> defined in the caller
> preferences draft to explicit request History-Info be included:
> history-directive
> 
> Pros: 
> - Consistent with the intent of that header in that it's a 
> mechanism for a
> client to request how a server/proxy should handle a request 
> 
> Cons:
> - that field isn't extensible 
> - doesn't really accomplish more than what can be implied by 
> the Supported
> header (i.e. if the client wants the information in a 
> response, then if a
> server is capable of adding the header, it SHOULD). 
> 
> 3) Make use of or define something similar to the Session 
> specific policy,
> whereby the client "authorizes" a proxy to capture 
> History-Info and that
> it's only captured in that case.   
> 
> Pros:
> - Explicit.
> 
> Cons:
> - Seems like a lot of overhead, with little advantage. 
> 
> 4) Follow the approach of existing optional SIP headers added by
> intermediaries (e.g. Path and Reason), which is that it's 
> purely a local
> policy decision and doesn't require any coordination with the UA as to
> whether it should or shouldn't be added. 
> 
> Pros: 
> - It's fairly straightforward and consistent with the intent 
> of optional
> headers.
> 
> Cons: 
> - Does require some guidelines to ensure that applications 
> that might use
> the optional information are robust enough to handle the lack of
> information. 
> 
> Feedback on these proposals is appreciated and hopefully this 
> discussion
> will allow us to reach a final, constructive conclusion.  
> 
> Obviously, my preference is the 4th option and perhaps one 
> reason why we've
> gotten so stuck on this point is that the guidelines for 
> applications on the
> usage of the header are perhaps not obvious.  So, it might be 
> useful to add
> an additional section to the document (or perhaps include additional
> information for each of the scenarios in the appendix) on application
> guidelines around the usage of the information and default 
> behavior if the
> information is not available.
> 
> Regards, 
> Mary H. Barnes
> mary.barnes@nortelnetworks.com
> 972-684-5432
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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Thread-Topic: SIP for controlling user to website interactions via http
Thread-Index: AcPt4QT83bHRo9PkQburHzVJa0pPMQ==
From: "Khare, Anir" <Aniruddha.Khare@team.telstra.com>
To: <sip@ietf.org>
Cc: "Khare, Anir" <Aniruddha.Khare@team.telstra.com>
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Subject: [Sip] SIP for controlling user to website interactions via http
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I have a general query:

Stateless HTTP protocol is currently used beyond its original purpose =
i.e. ad-hoc access of information on web without any access-control =
restrictions etc. Variety of techniques are used like cookies, http =
redirects for mimicking user session with web site as well as for =
identity management and SSO (Liberty).

SIP seems to be a fit for streamlining majority of these issues around =
user-web site http interactions. i.e. a scenario could be described as: =
user opens a SIP session with web site, negotiates preferences etc. =
using SDP. Then the session between user-website can consist of http, =
RTP and other streaming protocols as necessary.  SIP can be mediated by =
the service provider/telcos (similar to telcos providing SIP based =
signalling infrastructure for establishment of VoIP/MoIP services)  for =
establishing trust relationships between the user and the web-site, for =
charging  as well as for providing QoS for the transport of traffic in =
the SIP session (using appropriate QoS signalling).

Is SIP directly or indirectly used in this space? If not, was it =
considered before and rejected? If rejected, what were the issues?

Is there any other session control protocol under development for this =
space? When Http used for SOAP in web services interactions, does the =
notion of session gets formalised via any of the related web services =
protocols?=20

Regards,
Anir Khare
Telstra Research Labs, Australia
+61-3-6323 2659



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        Brian.Rosen@marconi.com, sip@ietf.org
Subject: Re: [Sip] Session timer question
From: Hiroaki Hata <hata@sphere.ad.jp>
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Jonathan
I thought that a response to re-INVITE was a part of spec of session timer.
That is independent on session timer specification. I understood the reason
why the session timer works even if a peer doesn't support it.
--
Hiro

> 
> 
> Hiroaki Hata wrote:
> 
> > Hi
> > Even if UAC does not support timer and ignores sesssion expires header
> > in a 2xx response, a session could be refreshed in case that UAC can response
> > a 2xx to re-INVITE from UAS. If UAC is not able to recongize a difference between
> > INVITE and re-INVITE and responds 4xx to the request of the refresh, a session
> > would be terminated. It depends just upon the implementation of UAC, doesn't it?
> 
> Why would the caller reject the re-INVITE?
> 
> > 
> > PS.The table 2 Jonathan mentioned below is the Figure 3 in draft-13 but it is still 
> > called table 3 in text page20.
> 
> Thanks for pointing this out.
> 
> -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] SIP for controlling user to website interactions via http
From: Scott Lawrence <slawrence@pingtel.com>
To: "Khare, Anir" <Aniruddha.Khare@team.telstra.com>
Cc: sip@ietf.org
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On Sat, 2004-02-07 at 20:15, Khare, Anir wrote:
> I have a general query:
> 
> Stateless HTTP protocol is currently used beyond its original purpose i.e. ad-hoc access of information on web without any access-control restrictions etc. Variety of techniques are used like cookies, http redirects for mimicking user session with web site as well as for identity management and SSO (Liberty).
> 
> SIP seems to be a fit for streamlining majority of these issues around user-web site http interactions. i.e. a scenario could be described as: user opens a SIP session with web site, negotiates preferences etc. using SDP. Then the session between user-website can consist of http, RTP and other streaming protocols as necessary.  SIP can be mediated by the service provider/telcos (similar to telcos providing SIP based signalling infrastructure for establishment of VoIP/MoIP services)  for establishing trust relationships between the user and the web-site, for charging  as well as for providing QoS for the transport of traffic in the SIP session (using appropriate QoS signalling).

SIP is no more suited to providing a session for those purposes than
HTTP already is.

-- 
Scott Lawrence        
  Pingtel Corp.   



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Moving to sip - no response on sip-implementors ....

-----Original Message-----
From: sip-implementors-bounces@cs.columbia.edu
[mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of
Arunachalam Venkatraman
Sent: Friday, February 06, 2004 10:49 AM
To: sip-implementors@cs.columbia.edu
Subject: [Sip-implementors] Contact header on UPDATE

RFC3311 - UPDATE
Why is the Contact header mandatory on an UPDATE request?

Why is the Contact header mandatory on a 2xx response to an UPDATE
request?

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Subject: Re: [Sip] Contact header on UPDATE
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UPDATE is a target refresh request, which means it updates the remote 
target (i.e., the Contact). That means, like INVITE, it needs to contain 
a contact.

Thanks,
Jonathan R.

arunvenk wrote:

> Moving to sip - no response on sip-implementors ....
> 
> -----Original Message-----
> From: sip-implementors-bounces@cs.columbia.edu
> [mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of
> Arunachalam Venkatraman
> Sent: Friday, February 06, 2004 10:49 AM
> To: sip-implementors@cs.columbia.edu
> Subject: [Sip-implementors] Contact header on UPDATE
> 
> RFC3311 - UPDATE
> Why is the Contact header mandatory on an UPDATE request?
> 
> Why is the Contact header mandatory on a 2xx response to an UPDATE
> request?
> 
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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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Subject: RE: [Sip] Contact header on UPDATE
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Jonathan
Since INVITE establishes the target information, it is clear why it is
mandatory on the INVITE.
In subsequent rerquests withina dialog, the Contact need not be mandatory.
If the remote traget is to be changed, then the Contact must be specified,
otherwise not. What is broken if UPDATE does not have a Contact? What
functionality is lost?

My understanding for making Contact mandatory on a re-INVITE, was to avoid
making re-INVITE any different from an INVITE.

It is not a terribly onerous thing to provide Conatct on an UPDATE (and its
200), but I wish to understand what is achieved by doing it and what is lost
by not doing it.

Regards,
Venkat

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Tuesday, February 10, 2004 2:08 PM
To: arunvenk
Cc: sip@ietf.org
Subject: Re: [Sip] Contact header on UPDATE


UPDATE is a target refresh request, which means it updates the remote
target (i.e., the Contact). That means, like INVITE, it needs to contain
a contact.

Thanks,
Jonathan R.

arunvenk wrote:

> Moving to sip - no response on sip-implementors ....
>
> -----Original Message-----
> From: sip-implementors-bounces@cs.columbia.edu
> [mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of
> Arunachalam Venkatraman
> Sent: Friday, February 06, 2004 10:49 AM
> To: sip-implementors@cs.columbia.edu
> Subject: [Sip-implementors] Contact header on UPDATE
>
> RFC3311 - UPDATE
> Why is the Contact header mandatory on an UPDATE request?
>
> Why is the Contact header mandatory on a 2xx response to an UPDATE
> request?
>
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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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Subject: Re: [Sip] Session timer question
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Hiroaki Hata wrote:

> Jonathan
> I thought that a response to re-INVITE was a part of spec of session timer.

re-INVITEs are part of rfc3261. Thus, an rfc3261 compliant UA will 
respond to a reinvite, and it shouldn't reject it for no reason.

-Jonathan R.

> 
>>
>>Hiroaki Hata wrote:
>>
>>
>>>Hi
>>>Even if UAC does not support timer and ignores sesssion expires header
>>>in a 2xx response, a session could be refreshed in case that UAC can response
>>>a 2xx to re-INVITE from UAS. If UAC is not able to recongize a difference between
>>>INVITE and re-INVITE and responds 4xx to the request of the refresh, a session
>>>would be terminated. It depends just upon the implementation of UAC, doesn't it?
>>
>>Why would the caller reject the re-INVITE?
>>
>>
>>>PS.The table 2 Jonathan mentioned below is the Figure 3 in draft-13 but it is still 
>>>called table 3 in text page20.
>>
>>Thanks for pointing this out.
>>
>>-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
>>
> 
> 

-- 
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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This list is for NEW development of the core SIP Protocol
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Hello,

 

            I have a question about CallId. Suppose two UA's generate
callid's putting theire hostnames and send to firewall/NAT.

            UA1:INVITE

                  CallID: call1@192.10.10.10 <mailto:call1@192.10.10.10> 

            UA2: INVITE

                     CallId: call1@192.10.10.12 <mailto:call1@192.10.10.12> 

            

            Because those callid information contain IP addresses, the
firewall/NAT should translate these IP addresses by putting it's own IP in
callId. The callId's are changed to call1@firewall_ip, call1@firewall_ip.
When responses come back from their respective UAS's the firewall/NAT
translates the it's IP address in callId's to their original IP's. My
question is should I change the string "call1" to something generated by
firewall or Is it OK to leave like that. Even though they have same CallId,
the proxy should discriminate the calls by from-tag, callid, and to-tag.

According to Spec, any UA should have a different means of generating
callid's with those of other UA's. 

 

Anything to do correctly to the above procedure?

 

Thanks in advance,

Anil

This email contains material that is confidential. The content of this email
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Hello,</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I
have a question about CallId. Suppose two UA's generate callid's
putting theire hostnames and send to firewall/NAT.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA1:INVITE</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CallID: <a href="mailto:call1@192.10.10.10">call1@192.10.10.10</a></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UA2:
INVITE</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CallId: <a href="mailto:call1@192.10.10.12">call1@192.10.10.12</a></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Because
those callid information contain IP addresses, the firewall/NAT should
translate these IP addresses by putting it's own IP in callId. The callId's
are changed to call1@firewall_ip, call1@firewall_ip. When responses come back
from their respective UAS's the firewall/NAT translates the it's IP
address in callId's to their original IP's. My question is should I
change the string "call1" to something generated by firewall or Is
it OK to leave like that. Even though they have same CallId, the proxy should discriminate
the calls by from-tag, callid, and to-tag.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>According to Spec, any UA should have a different means of generating
callid's with those of other UA's. </span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Anything to do correctly to the above procedure?</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Thanks in advance,</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Anil</span></font></p>

<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>This
email contains material that is confidential. The content of this email is for
the sole use of the intended recipient(s). Any review or distribution by
persons other than the intended recipient(s) without the express permission of
NetScreen Technologies, Inc. is strictly prohibited. If you are not the
intended recipient, please contact the sender and delete/destroy all copies of
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accuracy or completeness of third party materials or information.</span></font></p>

<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
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From: "Christian Stredicke" <stredicke@snom.de>
To: "'Anil Bollineni'" <ABollineni@netscreen.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] CallId question
Date: Tue, 10 Feb 2004 22:15:00 +0100
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:-) That's the reason why we started to AVOID IP addresses in the call =
ID!!!

=20

Well I guess the firewall which is changing the Call-ID is a half-way
implementation of an application layer firewall. What it is doing is =
simply
junk and it's a miracle if it's working with any SIP compliant devices.

=20

The "call1" part should be significantly random to avoid using the same
call-ID. That's why most of the user agents chose a call ID that is
random-generated.

=20

Christian

=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Anil
Bollineni
Sent: Tuesday, February 10, 2004 9:27 PM
To: 'sip@ietf.org'
Subject: [Sip] CallId question

=20

Hello,

=20

            I have a question about CallId. Suppose two UA's generate
callid's putting theire hostnames and send to firewall/NAT.

            UA1:INVITE

                  CallID: call1@192.10.10.10

            UA2: INVITE

                     CallId: call1@192.10.10.12

           =20

            Because those callid information contain IP addresses, the
firewall/NAT should translate these IP addresses by putting it's own IP =
in
callId. The callId's are changed to call1@firewall_ip, =
call1@firewall_ip.
When responses come back from their respective UAS's the firewall/NAT
translates the it's IP address in callId's to their original IP's. My
question is should I change the string "call1" to something generated by
firewall or Is it OK to leave like that. Even though they have same =
CallId,
the proxy should discriminate the calls by from-tag, callid, and to-tag.

According to Spec, any UA should have a different means of generating
callid's with those of other UA's.=20

=20

Anything to do correctly to the above procedure?

=20

Thanks in advance,

Anil

This email contains material that is confidential. The content of this =
email
is for the sole use of the intended recipient(s). Any review or =
distribution
by persons other than the intended recipient(s) without the express
permission of NetScreen Technologies, Inc. is strictly prohibited. If =
you
are not the intended recipient, please contact the sender and =
delete/destroy
all copies of this email and any related attachments. NetScreen does not
guarantee the accuracy or completeness of third party materials or
information.

=20


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<body lang=3DDE link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<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'>:-) =
That&#8217;s the
reason why we started to AVOID IP addresses in the call =
ID!!!</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'>Well I guess =
the
firewall which is changing the Call-ID is a half-way implementation of =
an
application layer firewall. What it is doing is simply junk and =
it&#8217;s a miracle
if it&#8217;s working with any SIP compliant devices.</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'>The =
&#8220;call1&#8221;
part should be significantly random to avoid using the same call-ID. =
That&#8217;s
why most of the user agents chose a call ID that is =
random-generated.</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'>-----</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> sip-admin@ietf.org
[mailto:sip-admin@ietf.org] <b><span style=3D'font-weight:bold'>On =
Behalf Of </span></b>Anil
Bollineni<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, February =
10, 2004
9:27 PM<br>
<b><span style=3D'font-weight:bold'>To:</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'>Subject:</span></b> [Sip] CallId =
question</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=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'>Hello,</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;
I have a question about CallId. Suppose two UA's generate callid's =
putting
theire hostnames and send to firewall/NAT.</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;
UA1:INVITE</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CallID: <a =
href=3D"mailto:call1@192.10.10.10">call1@192.10.10.10</a></span></font></=
p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;
UA2: INVITE</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CallId: <a
href=3D"mailto:call1@192.10.10.12">call1@192.10.10.12</a></span></font></=
p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;
Because those callid information contain IP addresses, the firewall/NAT =
should
translate these IP addresses by putting it's own IP in callId. The =
callId's are
changed to call1@firewall_ip, call1@firewall_ip. When responses come =
back from
their respective UAS's the firewall/NAT translates the it's IP address =
in
callId's to their original IP's. My question is should I change the =
string
&quot;call1&quot; to something generated by firewall or Is it OK to =
leave like
that. Even though they have same CallId, the proxy should discriminate =
the
calls by from-tag, callid, and to-tag.</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'>According to Spec, any UA should have a =
different
means of generating callid's with those of other UA's. =
</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'>Anything to do correctly to the above =
procedure?</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'>Thanks in advance,</span></font></p>

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

<p><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
Arial'>This email contains material that is confidential. The content of =
this
email is for the sole use of the intended recipient(s). Any review or
distribution by persons other than the intended recipient(s) without the
express permission of NetScreen Technologies, Inc. is strictly =
prohibited. If
you are not the intended recipient, please contact the sender and
delete/destroy all copies of this email and any related attachments. =
NetScreen
does not guarantee the accuracy or completeness of third party materials =
or information.</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
lang=3DEN-US
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

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Date: Tue, 10 Feb 2004 16:25:46 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Dean Willis <dwillis@dynamicsoft.com>
CC: sip@ietf.org
Subject: Re: [Sip] Questions on draft-ietf-sip-session-timer-13 and refresh
 timers
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inline.

Dean Willis wrote:

> I'me reviewing some IESG comments on this draft, and I've found
> something I don't understand.
> 
> The original comment from Ted was something like "Why do we recommend
> refresh at 1/2 the session timer in sections 7 and 8, but require a BYE
> at a recommended 1/3St-10s in section 10?"
> 
> I understand why we require refresh, and why we suggested refreshing at
> 1/2 of the session timer value. What I don't understand is the BYE logic
> in Section 10.
> 
> 
>>From Section 10:
> 
>    If no 2xx response to a session refresh request is received before
>    the session expiration, the UA SHOULD send a BYE request to terminate
>    the session. It SHOULD send this BYE slightly before session
>    expiration. The minimum of ten seconds and one third the session
>    interval is RECOMMENDED.
> 
>    For example, if the session interval is 120 seconds, one third of
>    this is 40 seconds. Since the minimum of 10 seconds and 40 seconds
>    is 10 seconds, the BYE would be sent 10 seconds before the session
>    expires.
> 
> 
> I read this as:
> 
> A UA has issues a session refresh request, and is waiting around for
> that transaction to complete. Unfortunately, it either 1) didn't issue
> the request sufficiently early for the transaction to complete before
> the session is going to expire, so it needs to abandon the refreshing
> transaction and terminate the session before the session expires, 
> or 2) the session refresh request failed, but we're going to continue to
> use this session for a while before tearing it down before it expires.
> 
> It seems more reasonable that the UA should issue the session refresh
> request far enough in advance of session expiration that the session
> refresh request will complete (or fail with timeout) before the session
> expires, and that if the session refresh request fails, the UA should
> immediately terminate the session with a BYE.

I am inclined to agree with this. I honestly cannot remember why we 
chose {1/3,10s}. Indeed, this rule actually contradicts RFC3261! From 
section 12.2.1.2:

12.2.1.2 Processing the Responses

    The UAC will receive responses to the request from the transaction
    layer.  If the client transaction returns a timeout, this is treated
    as a 408 (Request Timeout) response.

    The behavior of a UAC that receives a 3xx response for a request sent
    within a dialog is the same as if the request had been sent outside a
    dialog.  This behavior is described in Section 8.1.3.4.

       Note, however, that when the UAC tries alternative locations, it
       still uses the route set for the dialog to build the Route header
       of the request.

    When a UAC receives a 2xx response to a target refresh request, it
    MUST replace the dialog's remote target URI with the URI from the
    Contact header field in that response, if present.





Rosenberg, et. al.          Standards Track                    [Page 75]

RFC 3261            SIP: Session Initiation Protocol           June 2002


    If the response for a request within a dialog is a 481
    (Call/Transaction Does Not Exist) or a 408 (Request Timeout), the UAC
    SHOULD terminate the dialog.  A UAC SHOULD also terminate a dialog if
    no response at all is received for the request (the client
    transaction would inform the TU about the timeout.)

In other words, it recommends exactly the behavior you are describing - 
on transaction timeout, terminate the dialog.

> 
> This implies that the lowest allowable threshold for the negotiated
> session timer should be larger than the largest timeout value for a
> session refresh request (which could be an INVITE transaction). It also
> may place requirements on UAs that have implemented non-standard timer
> values to calculate this lowest allowable threshold for the negotiated
> session timer value and include this calculated value in their
> negotiation by initializing the Min-SE value.

I think the spec should state that no entity can request a session timer 
below 1 minute.

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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Date: Tue, 10 Feb 2004 15:34:06 -0800
Subject: Re: [Sip] CallId question
From: Cullen Jennings <fluffy@cisco.com>
To: Anil Bollineni <ABollineni@netscreen.com>, <sip@ietf.org>
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The NAT/FW must not go changing this stuff. If it does, a S/MIME signed
sipfrag will fail. There is no need for the NAT to go change call IDs



On 2/10/04 12:27 PM, "Anil Bollineni" <ABollineni@netscreen.com> wrote:

> Hello,
>  
>             I have a question about CallId. Suppose two UA's generate callid's
> putting theire hostnames and send to firewall/NAT.
>             UA1:INVITE
>                  CallID: call1@192.10.10.10
>             UA2: INVITE
>                     CallId: call1@192.10.10.12
>             
>             Because those callid information contain IP addresses, the
> firewall/NAT should translate these IP addresses by putting it's own IP in
> callId. The callId's are changed to call1@firewall_ip, call1@firewall_ip. When
> responses come back from their respective UAS's the firewall/NAT translates
> the it's IP address in callId's to their original IP's. My question is should
> I change the string "call1" to something generated by firewall or Is it OK to
> leave like that. Even though they have same CallId, the proxy should
> discriminate the calls by from-tag, callid, and to-tag.
> According to Spec, any UA should have a different means of generating callid's
> with those of other UA's.
>  
> Anything to do correctly to the above procedure?
>  
> Thanks in advance,
> Anil
> This email contains material that is confidential. The content of this email
> is for the sole use of the intended recipient(s). Any review or distribution
> by persons other than the intended recipient(s) without the express permission
> of NetScreen Technologies, Inc. is strictly prohibited. If you are not the
> intended recipient, please contact the sender and delete/destroy all copies of
> this email and any related attachments. NetScreen does not guarantee the
> accuracy or completeness of third party materials or information.
> 
>  
> 



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<HTML>
<HEAD>
<TITLE>Re: [Sip] CallId question</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><BR>
The NAT/FW must not go changing this stuff. If it does, a S/MIME signed sip=
frag will fail. There is no need for the NAT to go change call IDs<BR>
<BR>
<BR>
<BR>
On 2/10/04 12:27 PM, &quot;Anil Bollineni&quot; &lt;ABollineni@netscreen.co=
m&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Arial"=
>Hello,<BR>
&nbsp;<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;I h=
ave a question about CallId. Suppose two UA's generate callid's putting thei=
re hostnames and send to firewall/NAT.<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;UA1=
:INVITE<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;CallID: call1@192.10.10.10<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;UA2=
: INVITE<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CallId: call1@192.10.10.12<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR=
>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Bec=
ause those callid information contain IP addresses, the firewall/NAT should =
translate these IP addresses by putting it's own IP in callId. The callId's =
are changed to call1@firewall_ip, call1@firewall_ip. When responses come bac=
k from their respective UAS's the firewall/NAT translates the it's IP addres=
s in callId's to their original IP's. My question is should I change the str=
ing &quot;call1&quot; to something generated by firewall or Is it OK to leav=
e like that. Even though they have same CallId, the proxy should discriminat=
e the calls by from-tag, callid, and to-tag.<BR>
According to Spec, any UA should have a different means of generating calli=
d's with those of other UA's. <BR>
&nbsp;<BR>
Anything to do correctly to the above procedure?<BR>
&nbsp;<BR>
Thanks in advance,<BR>
Anil<BR>
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 the intended recipient, please contact the sender and delete/destroy all co=
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From exim@www1.ietf.org  Wed Feb 11 08:06:48 2004
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From: "Jesske, R" <R.Jesske@t-com.net>
To: ali.el-moussawi@hp.com, sip@ietf.org
Cc: "Poetzl, Joachim" <Joachim.Poetzl@t-com.net>,
        "Martinez-Rebordosa, Anna" <Anna.Martinez-Rebordosa@t-com.net>,
        "Schuette, Wolfgang" <Wolfgang.Schuette@t-com.net>,
        "Benz, Robert" <Robert.Benz@t-com.net>,
        "Pinker, Gerold" <Gerold.Pinker@t-com.net>
Subject: AW: [Sip] Some details in RFC 3398
Date: Wed, 11 Feb 2004 14:03:43 +0100
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Hi Ali,
I think this is a wrong description of ISUP Procedures on the outgoing ISUP side. If you are looking in Q.732.2-5 you will find the correct description of call forwarding scenarios. 
You will never find a situation were a ACM and a CPG will be send in backward direction at once, because all information will be given in the early ACM.
The "Generic notification indicator" will be set to "Call is a diverting call" that's all. No CPG is now needed to indicate the same.

With regard to the timer expiry please look in Q.1912.5 there you will find detail information. (e.g. T7 --> 484 Address Incomplete, T9-->480 Temporarily Unavailable...)

I hope this helps.

Best Regards

Roland


-----Ursprungliche Nachricht-----
Von: sip-admin@ietf.org [mailto:sip-admin@ietf.org]Im Auftrag von El Moussawi, Ali
Gesendet: Mittwoch, 4. Februar 2004 14:45
An: sip@ietf.org
Betreff: [Sip] Some details in RFC 3398


Hi all,
I have two tiny questions concerning RFC3398.
In  7.1.6 , what do you think the cause code should be in the REL sent to the PSTN network after interwork tinmer expires ?  
On the other hand, in 8.2.3, it says that upon receipt of a SIP 181 response, the gateway should send an early ACM to the PSTN side, plus a CPG with event code 6. What is the use of CPG since the other side will already know that its call is beying handled when it receives a early ACM ? Is the CPG mandatory ?
Thanks a lot 

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From exim@www1.ietf.org  Wed Feb 11 08:16:38 2004
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From: "Jesske, R" <R.Jesske@t-com.net>
To: mary.barnes@nortelnetworks.com, sip@ietf.org
Subject: AW: [Sip] Optionality for History-Info
Date: Wed, 11 Feb 2004 14:12:57 +0100
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Mary,
I'm also happy with option 4.

Best Regards

Roland

-----Ursprungliche Nachricht-----
Von: sip-admin@ietf.org [mailto:sip-admin@ietf.org]Im Auftrag von Mary
Barnes
Gesendet: Donnerstag, 5. Februar 2004 15:39
An: 'sip@ietf.org'
Betreff: [Sip] Optionality for History-Info


I'm in the process of updating History Info
(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-01.txt)
based upon a few comments I've received and merging the requirements into
the solution document.  However, I really need more feedback from the
working group to address the primary point raised during the discussion in
Minneapolis on the optionality.  Initial analysis way long ago resulted in
the decision that Local Policy was the best approach to the optionality,
since it didn't seem reasonable to use Require for an optional header and
lack of the header should not prevent a basic request from being processed.
However, a concern has been raised that Local Policy isn't sufficient as it
requires coordination between clients and proxies (i.e. there's no aspect of
the protocol that allows a client to define or discover that the Proxy will
add the header).  Supported, of course, is used to let the Proxy know that
the information, if available, should be included in the responses.  

In taking a few steps back and re-evaluating all the possibilities for
making the optionality more under the control of the protocol and I came up
with the following list of possibilities. It may not be complete, nor are
the list of pros and cons intended to be exhaustive, but I think it serves
as a useful basis for discussion.

1) Use Require 
Pros: 
- Puts the use of the header explicitly under control of the client
initiating the request. 

Cons: 
- Seems extremely restrictive and functionally limiting, although it would
depend upon the application at the end client that wants to make use of the
information as to how detrimental this might be. 
- Doesn't make sense since the applicability of History-Info isn't
restricted to the initiator of the request. 


2) Make use of the Request-Disposition header field as defined in the caller
preferences draft to explicit request History-Info be included:
history-directive

Pros: 
- Consistent with the intent of that header in that it's a mechanism for a
client to request how a server/proxy should handle a request 

Cons:
- that field isn't extensible 
- doesn't really accomplish more than what can be implied by the Supported
header (i.e. if the client wants the information in a response, then if a
server is capable of adding the header, it SHOULD). 

3) Make use of or define something similar to the Session specific policy,
whereby the client "authorizes" a proxy to capture History-Info and that
it's only captured in that case.   

Pros:
- Explicit.

Cons:
- Seems like a lot of overhead, with little advantage. 

4) Follow the approach of existing optional SIP headers added by
intermediaries (e.g. Path and Reason), which is that it's purely a local
policy decision and doesn't require any coordination with the UA as to
whether it should or shouldn't be added. 

Pros: 
- It's fairly straightforward and consistent with the intent of optional
headers.

Cons: 
- Does require some guidelines to ensure that applications that might use
the optional information are robust enough to handle the lack of
information. 

Feedback on these proposals is appreciated and hopefully this discussion
will allow us to reach a final, constructive conclusion.  

Obviously, my preference is the 4th option and perhaps one reason why we've
gotten so stuck on this point is that the guidelines for applications on the
usage of the header are perhaps not obvious.  So, it might be useful to add
an additional section to the document (or perhaps include additional
information for each of the scenarios in the appendix) on application
guidelines around the usage of the information and default behavior if the
information is not available.

Regards, 
Mary H. Barnes
mary.barnes@nortelnetworks.com
972-684-5432



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From exim@www1.ietf.org  Wed Feb 11 08:25:43 2004
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From: "Jesske, R" <R.Jesske@t-com.net>
To: mary.barnes@nortelnetworks.com, sebastien.prouvost@francetelecom.com
Cc: sip@ietf.org
Subject: AW: RE : [Sip] draft-ietf-sip-history-info-01.txt
Date: Wed, 11 Feb 2004 14:22:39 +0100
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Hi Mary,

Yes your last paragraph is the correct interpretation of my email.

I think a  specific privacy tag enables the domain more flexibility with regard to the information shown to the next domain or the user.

E.g. a forwarding entity does not allow the presentation of the URI but the originating do. With a general privacy tag all information will not be forwarded to the UAS, but if you have an explicit tag for the history info only the history info will be not shown and the originating URI will be shown.

Also information that is only needed for domain/local purposes will be hidden but not the information of the originating of the SIP session.

From my understanding a specific privacy tag will help in such kind of use.

Best Regards

Roland

-----Ursprungliche Nachricht-----
Von: Mary Barnes [mailto:mary.barnes@nortelnetworks.com]
Gesendet: Mittwoch, 4. Februar 2004 23:13
An: 'PROUVOST S?bastien FTRD/DAC/ISS'; Jesske, Roland
Cc: sip@ietf.org
Betreff: RE: RE : [Sip] draft-ietf-sip-history-info-01.txt




Sebastien and Roland,

I apologize for the delay in responding.  

You are correct in that one privacy aspect currently addressed for
History-Info is around the UA's ability to require privacy for elements of a
message that can somehow divulge information about the end client's
identity. In some ways, History-Info could be considered in the category of
Record Route,  Via, etc.  However, HI is an optional header and per the
recommendation in RFC 3323, the decision was made that if Header privacy was
requested, then History-Info SHOULD not be included.   In addition, if
through local policy or some other local information available wrt the
retargeted URIs, there is knowledge that there should be privacy applied to
these URIs, then as discussed in RFC 3323, the intermediary processing that
request can also add the Privacy header to ensure that History-Info is also
not captured by subsequent intermediaries.  This latter point perhaps is not
so clearly or explicitly spelled out in the current document in the privacy
sections, but there is discussion in the detailed proxy processing behaviour
in 2.3.3 that the local policy can define whether the History-Info captured
would leave a specific domain (this is of course for optionality, as well,
which is another issue that I'll address in a separate email).      

It seems that what's being discussed in these emails is introducing a
specific privacy tag against those entries that local policy defines should
not be forwarded outside a specific domain,  so that they can be removed
(and don't need to be kept persistent by a privacy service).  Right now,
there's an asumption that these entries are somehow known by the proxy, but
perhaps an explicit tag would be useful.  OR are you talking about extending
the current model in RFC 3323 to specifically address the use of a privacy
service for History-Info headers?  

Regards,
Mary H. Barnes
mary.barnes@nortelnetworks.com
972-684-5432


-----Original Message-----
From: PROUVOST Sebastien FTRD/DAC/ISS
[mailto:sebastien.prouvost@francetelecom.com]
Sent: Thursday, January 22, 2004 4:32 AM
To: Jesske, R; sip@ietf.org; Barnes, Mary [NGC:B622:EXCH]; Watson, Mark
[MOP:EP10:EXCH]; fluffy@cisco.com
Subject: RE : [Sip] draft-ietf-sip-history-info-01.txt


I fully support this proposal. 

The privacy regarding the history-info header relates to the privacy policy
accorded to the target of the request (before it is retargeted). Therefore
it shall not depend on the privacy policy accorded to the calling UA. The
history-info header shall have its own privacy value. 
Moreover in case the request is retargeted more than once (particularly if
the targets represent different users), it shall be possible to request
privacy for a target independently of the others.  

Regards,
Sebastien Prouvost
France Telecom R&D/DAC/CAR

-----Message d'origine-----
De : Jesske, R [mailto:R.Jesske@t-com.net] 
Envoye : vendredi 16 janvier 2004 10:29
A : sip@ietf.org
Cc : Poetzl, Joachim; Alexeitsev, D
Objet : [Sip] draft-ietf-sip-history-info-01.txt


Dear all,
in 1.3 Ensuring the Privacy of History_Info it is proposed to use a
priv-value of Session or Header level privacy. I would like to propose a own
privacy level for the history info, because the Information shows also
information about the network. With regard to the provider operating a
network some of the providers don't like to transmit information about call
history. Or a redirecting party don't like to show the redirecting address.
Therefore it should be possible to delete the history info at domain
boundaries based on the privacy-value.

I propose the priv-value   =   "history_info"  for the whole History-Info
header.

In addition a option would be useful where only parts (e.g Indes1.1) of the
history info are restricted. That could be done with a privacy statement
within the History-Info .

Best Regards

Roland


Deutsche Telekom AG
T Com Zentrale
Roland Jesske, T332-2
Section T33; Signalling, Gateways and Switching Systems 
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940 
Fax:      +49 6151 83-4577 
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From exim@www1.ietf.org  Wed Feb 11 08:30:40 2004
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Is it ok to send a 302 redirect on a register to redirect the UA to another SIP registrar?

/Olle


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From: "Elwell, John" <john.elwell@siemens.com>
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Subject: [Sip] New I-D on state update
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Please be aware of a new I-D on state update (I did not see any notice on
the list):

http://www.ietf.org/internet-drafts/draft-elwell-sip-state-update-00.txt
<http://www.ietf.org/internet-drafts/draft-elwell-sip-state-update-00.txt> 

Abstract: This document examines the need for updating state information,
such as remote party identity, during a SIP dialog. It explores existing
mechanisms that might be appropriate and identifies matters that need to be
fixed to make this possible. 

John Elwell (john.elwell@siemens.com)


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<P><FONT SIZE=2 FACE="Arial">Please be aware of a new I-D on state update (I did not see any notice on the list):</FONT>
</P>

<P><A HREF="http://www.ietf.org/internet-drafts/draft-elwell-sip-state-update-00.txt"><U><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">http://www.ietf.org/internet-drafts/draft-elwell-sip-state-update-00.txt</FONT></U></A>
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<P><FONT SIZE=2 FACE="Arial">Abstract: This document examines the need for updating state information, such as remote party identity, during a SIP dialog. It explores existing mechanisms that might be appropriate and identifies matters that need to be fixed to make this possible. </FONT></P>

<P><FONT SIZE=2 FACE="Arial">John Elwell (john.elwell@siemens.com)</FONT>
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From exim@www1.ietf.org  Wed Feb 11 10:09:43 2004
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Cullen Jennings wrote:

>
> The NAT/FW must not go changing this stuff. If it does, a S/MIME 
> signed sipfrag will fail. There is no need for the NAT to go change 
> call IDs

Yes, NAT does not change the content of SIP messages (Call-ID, 
Via-Header, or SDP content...), since it maps IP/ports at the transport 
level (TCP/UDP...). That's even why it is necessary to use additional  
functions (STUN/TURN and more..) to make SIP work with NAT...

>
>
>
> On 2/10/04 12:27 PM, "Anil Bollineni" <ABollineni@netscreen.com> wrote:
>
>     Hello,
>      
>                 I have a question about CallId. Suppose two UA's
>     generate callid's putting theire hostnames and send to firewall/NAT.
>                 UA1:INVITE
>                      CallID: call1@192.10.10.10
>                 UA2: INVITE
>                         CallId: call1@192.10.10.12
>                 
>                 Because those callid information contain IP addresses,
>     the firewall/NAT should translate these IP addresses by putting
>     it's own IP in callId. The callId's are changed to
>     call1@firewall_ip, call1@firewall_ip. When responses come back
>     from their respective UAS's the firewall/NAT translates the it's
>     IP address in callId's to their original IP's. My question is
>     should I change the string "call1" to something generated by
>     firewall or Is it OK to leave like that. Even though they have
>     same CallId, the proxy should discriminate the calls by from-tag,
>     callid, and to-tag.
>     According to Spec, any UA should have a different means of
>     generating callid's with those of other UA's.
>      
>     Anything to do correctly to the above procedure?
>




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Olle E. Johansson wrote:

> Is it ok to send a 302 redirect on a register to redirect the UA to 
> another SIP registrar?
>
> /Olle 


In section 10.3 of RFC 3261, it is written:

"registrar MAY redirect REGISTER requests as appropriate. One common 
usage would be for a registrar listening on a multicast interface to 
redirect multicast REGISTER requests to its own unicast interface with a 
302 (Moved Temporarily) response."

So, I would consider you MAY do it.

Thomas  


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From exim@www1.ietf.org  Wed Feb 11 11:30:44 2004
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Date: Wed, 11 Feb 2004 11:24:51 -0500
From: "Dusan Mudric" <dmudric@nortelnetworks.com>
Organization: Nortel Networks
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: Scott Lawrence <slawrence@pingtel.com>, Paul Kyzivat <pkyzivat@cisco.com>,
        ksrini@motorola.com, sip@ietf.org
Subject: Re: [Sip] Registering Multiple 'Contacts'
References: <Pine.LNX.4.21.0401121159440.23050-100000@bulb.corp.mot.com>	<4002E0B9.8030203@cisco.com> <vheku1f0ks.fsf@sukothai.pingtel.com> <400FEFD1.60004@dynamicsoft.com>
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Hi:

Let me compare given example with registration renewal. It my clarify
some doubts.

Registration renewal is done for the same Contact URI's, within the same
call leg, using the same Call-ID and incremented CSeq number in the
refresh REGISTER request. This just extends registration time but
doesn't change or add Contact URI. At refresh, Contact(s)'s CSeq number
is updated with a new one, as well as {action, expires, qvalue}
parameters. It is unlikely that Contact parameters are going to be
changed in the renewal requests.

For the same Contact URI in a refresh REGISTER request, if Call-ID is
different, the Contact's Call-ID is updated with a new one, if
expiration time is not zero.

There is no replace of Contact URI's. Only remove (expires=0 for the
Contact) and add.

Is this right?

Thanks,
Dusan.

Jonathan Rosenberg wrote:
> 
> Scott Lawrence wrote:
> 
> > Srinivasan Krishnamoorthy wrote:
> >
> >
> >>>  Section 10.2.4 of RFC3261 'Refreshing Bindings' says:
> >>>  "The UA then issues a REGISTER request for each of its bindings
> >>>before the expiration interval has elapsed. It MAY combine   several
> >>>updates into one REGISTER request."
> >>>  The above seems to indicate that the UAC may send:   (forgive the
> >>>Syntax)
> >>>        REGISTER         To: user@operator.net
> >>>        Contact: sip:user1@10.0.0.1;expires=3600
> >>>                200 OK
> >>>        REGISTER         To: user@operator.net
> >>>        Contact: sip:user1@192.0.0.1;expires=3600
> >>>                200 OK
> >>>  This, according to RFC3261, will register both Contact1 and Contact2
> >>>for user@operator.net. (The server will then use the q parameter
> >>>  of the contact for priority among contacts)
> >
> >
> > Paul Kyzivat <pkyzivat@cisco.com> writes:
> >
> >
> >>Yes.
> >
> >
> > My reading is that the results should differ depending on the Call-Id
> > used in those two REGISTER requests:
> >
> >   - If they have different Call-Id values, then the server should
> >     register both contacts.
> >
> >   - If they have the same Call-Id value, then the second REGISTER
> >     should cause the server to replace the first contact with the
> >     second, so that there would be only
> 
> No, this is not the case. The callid/cseq are used as an ordering
> tool. In the abovoe case, if the Call-ID are the same, both contacts
> are added, and both are marked as having that particular callid, but
> with different cseq. if the Call-id are different, both contacts are
> added, but with different call-id and cseq.
> 
> -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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Subject: [Sip] IPR notice on GRUU filed
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-------- Original Message --------
Subject: IPR Notice
Date: Tue, 10 Feb 2004 18:33:37 -0500 (EST)
From: IESG Secretary <iesg-secretary@ietf.org>
To: jdrosen@dynamicsoft.com
CC: mankin@psg.com, <rohan@cisco.com>, <dean.willis@softarmor.com>, 
<ietf-secretariat@ietf.org>

Dear Mr. Rosenberg,

An IPR Notice has been posted that pertains to your Internet-Draft
entitled "Obtaining and Using Globally Routable User Agent (UA) URIs
(GRUU) in the Session Iniation Protocol (SIP)"
(http://www.ietf.org/internet-drafts/draft-ietf-sip-gruu-00.txt).

The title of the IPR Notice is "Microsoft Corporation's Statement About
IPR Claimed in draft-ietf-sip-gruu"
(http://www.ietf.org/ietf/IPR/microsoft-ipr-draft-ietf-sip-gruu.txt).

The IETF Secretariat



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Hi all,
 If a UAC generates a request in Dialog (say an ACK for 200 it received),
 
 If the 200 had a Record-Route with loose routing and also a Contact 
header where should it route the request ? Can it route it directly to 
the IP/FQDN specified in the uri of Contact ? Does any header take 
precedence over other when generating a subsequent request in a Dialog ?
 Is the significance of Contact header end-to-end or between UAs. For 
the example given in the RFC 3261 (alice and bob), can the intermediate 
proxies modify the value of the Contact header in the response sent by 
Alice to Bob ?

 Thank you,

 Chintan
 ipNetFusion Inc,
 Dallas, Tx - 75080



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http://www.ietf.org/internet-drafts/draft-sterman-aaa-sip-01.txt

contains a revised version of the draft. It has a Security Considerations
section now and I've added support for Authentication-Info. Please
send your comments to to the radiusext mailing list, radiusext@ops.ietf.org.


Wolfgang Beck

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To: Matthew.Gardiner@aculab.com
CC: cthakker@ipnetfusion.com, sip@ietf.org
Subject: Re: [Sip] Record-Route and Contact
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Matthew.Gardiner@aculab.com wrote:
> 
> It's my understanding that Record-Route headers take precedence over
> the IP/FQDN in the contact header of the response, as they
> (Record-Route headers) are inserted by specifically by the proxies in
> order to influence the route that subsequent messaging takes. 

Assumption is that proxy is in a Record-Route URI domain. Otherwise some
other means can be used to route a request.

> They influence, not only the route taken by the ACK, furthermore that taken
> by any subsequent requests.

Except REGISTER, which must ignore a Record-Route.

> 
> -----Original Message-----
> From: cthakker [mailto:cthakker@ipnetfusion.com]
> Sent: 11 February 2004 21:05
> To: sip@ietf.org
> Subject: [Sip] Record-Route and Contact
> 
> Hi all,
>  If a UAC generates a request in Dialog (say an ACK for 200 it
> received),
> 
>  If the 200 had a Record-Route with loose routing and also a Contact
> header where should it route the request ? Can it route it directly to
> 
> the IP/FQDN specified in the uri of Contact ? Does any header take
> precedence over other when generating a subsequent request in a Dialog
> ?
>  Is the significance of Contact header end-to-end or between UAs. For
> the example given in the RFC 3261 (alice and bob), can the
> intermediate
> proxies modify the value of the Contact header in the response sent by
> 
> Alice to Bob ?
> 
>  Thank you,
> 
>  Chintan
>  ipNetFusion Inc,
>  Dallas, Tx - 75080
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> For Aculab's privacy policy and E-Mail disclaimer, Click Here :
> http://www.aculab.com/company/legal_notice.htm.

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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] Contact header on UPDATE
Date: Fri, 13 Feb 2004 04:00:56 +0530
Message-ID: <10C4348A1BA43A4FA8D5B767E052CEADEDB396@blr-ec-msg04.wipro.com>
Thread-Topic: [Sip] Contact header on UPDATE
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Comments inline...

Thanks & Regards,
Nataraju A.B.=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
Arunachalam Venkatraman
Sent: Wednesday, February 11, 2004 1:49 AM
To: Jonathan Rosenberg
Cc: sip@ietf.org
Subject: RE: [Sip] Contact header on UPDATE

Jonathan
Since INVITE establishes the target information, it is clear why it is
mandatory on the INVITE.
In subsequent rerquests withina dialog, the Contact need not be
mandatory.
If the remote traget is to be changed, then the Contact must be
specified,
otherwise not. What is broken if UPDATE does not have a Contact? What
functionality is lost?

My understanding for making Contact mandatory on a re-INVITE, was to
avoid
making re-INVITE any different from an INVITE.

It is not a terribly onerous thing to provide Conatct on an UPDATE (and
its
200), but I wish to understand what is achieved by doing it and what is
lost
by not doing it.

[ABN] here you are using UPDATE with an intention of updating your
contact information. I can narrate your scenario as sending an envelope
without addressee contact information. In this case for sure envelope
does not reach the intended destination.=20
Here I meant that you are sending an UPDATE to update your contact
information with the remote party, and UPADTE without contact
information what does it serves for you -- Nothing... hence contact is
mandatory for UPDAET request...

Regards,
Venkat

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Tuesday, February 10, 2004 2:08 PM
To: arunvenk
Cc: sip@ietf.org
Subject: Re: [Sip] Contact header on UPDATE


UPDATE is a target refresh request, which means it updates the remote
target (i.e., the Contact). That means, like INVITE, it needs to contain
a contact.

Thanks,
Jonathan R.

arunvenk wrote:

> Moving to sip - no response on sip-implementors ....
>
> -----Original Message-----
> From: sip-implementors-bounces@cs.columbia.edu
> [mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of
> Arunachalam Venkatraman
> Sent: Friday, February 06, 2004 10:49 AM
> To: sip-implementors@cs.columbia.edu
> Subject: [Sip-implementors] Contact header on UPDATE
>
> RFC3311 - UPDATE
> Why is the Contact header mandatory on an UPDATE request?
>
> Why is the Contact header mandatory on a 2xx response to an UPDATE
> request?
>
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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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This list is for NEW development of the core SIP 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] Record-Route and Contact
Date: Fri, 13 Feb 2004 04:15:41 +0530
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=20

=20

Thanks & Regards,
Nataraju A.B.=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
Matthew.Gardiner@aculab.com
Sent: Thursday, February 12, 2004 1:09 PM
To: cthakker@ipnetfusion.com
Cc: sip@ietf.org
Subject: RE: [Sip] Record-Route and Contact

=20

It's my understanding that Record-Route headers take precedence over the
IP/FQDN in the contact header of the response, as they (Record-Route
headers) are inserted by specifically by the proxies in order to
influence the route that subsequent messaging takes. They influence, not
only the route taken by the ACK, furthermore that taken by any
subsequent requests.

[ABN] once 200 for INVITE is received it forms the Route header for
further requests sent in the same dialog. The logic to form Route header
is been clearly mentioned in the Rfc3261 and the contact information in
200-INVITE forms the Req-URI for subsequent requests.=20

-----Original Message-----=20
From: cthakker [mailto:cthakker@ipnetfusion.com]=20
Sent: 11 February 2004 21:05=20
To: sip@ietf.org=20
Subject: [Sip] Record-Route and Contact=20

=20

Hi all,=20
 If a UAC generates a request in Dialog (say an ACK for 200 it
received),=20
 =20
 If the 200 had a Record-Route with loose routing and also a Contact=20
header where should it route the request ? Can it route it directly to=20
the IP/FQDN specified in the uri of Contact ? Does any header take=20
precedence over other when generating a subsequent request in a Dialog ?

 Is the significance of Contact header end-to-end or between UAs. For=20
the example given in the RFC 3261 (alice and bob), can the intermediate=20
proxies modify the value of the Contact header in the response sent by=20
Alice to Bob ?=20

 Thank you,=20

 Chintan=20
 ipNetFusion Inc,=20
 Dallas, Tx - 75080=20

=20

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

=20

For Aculab's privacy policy and E-Mail disclaimer, Click Here :
http://www.aculab.com/company/legal_notice.htm.=20

=20


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<p><font size=3D2 color=3Dblue face=3D"Courier New"><span =
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font-family:"Courier New";color:blue'>Thanks &amp; Regards,<br>
Nataraju A.B. </span></font></p>

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<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'>-----Original =
Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> sip-admin@ietf.org
[mailto:sip-admin@ietf.org] <b><span style=3D'font-weight:bold'>On =
Behalf Of </span></b>Matthew.Gardiner@aculab.com<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> </span></font><font =
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>Thursday,
 February 12, 2004</span></font><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> </span></font><font
 size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>1:09 PM</span></font><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'><br>
<b><span style=3D'font-weight:bold'>To:</span></b> =
cthakker@ipnetfusion.com<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [Sip] =
Record-Route
and Contact</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 style=3D'margin-left:.5in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>It's my understanding that Record-Route =
headers take
precedence over the IP/FQDN in the contact header of the response, as =
they
(Record-Route headers) are inserted by specifically by the proxies in =
order to
influence the route that subsequent messaging takes. They influence, not =
only
the route taken by the ACK, furthermore that taken by any subsequent =
requests.</span></font></p>

<p style=3D'margin-left:.5in'><font size=3D2 color=3Dblue =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>[ABN] =
once 200
for INVITE is received it forms the Route header for further requests =
sent in
the same dialog. The logic to form Route header is been clearly =
mentioned in the
Rfc3261 and the contact information in 200-INVITE forms the Req-URI for
subsequent requests. </span></font></p>

<p style=3D'margin-left:.5in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>-----Original Message-----</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>From: cthakker [<a
href=3D"mailto:cthakker@ipnetfusion.com">mailto:cthakker@ipnetfusion.com<=
/a>]</span></font>
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>Sent: =
</span></font><font size=3D2><span style=3D'font-size:10.0pt'>11 =
February 2004</span></font><font size=3D2><span
style=3D'font-size:10.0pt'> </span></font><font size=3D2><span =
style=3D'font-size:
 10.0pt'>21:05</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>To: =
sip@ietf.org</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>Subject: [Sip] =
Record-Route and
Contact</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 style=3D'margin-left:.5in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>Hi all,</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;If a UAC generates =
a request
in Dialog (say an ACK for 200 it received),</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;</span></font> =
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;If the 200 had a =
Record-Route
with loose routing and also a Contact </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>header where should it =
route the
request ? Can it route it directly to </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>the IP/FQDN specified in =
the uri of
Contact ? Does any header take </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>precedence over other =
when
generating a subsequent request in a Dialog ?</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;Is the =
significance of
Contact header end-to-end or between UAs. For </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>the example given in the =
RFC 3261
(alice and bob), can the intermediate </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>proxies modify the value =
of the
Contact header in the response sent by </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>Alice to Bob =
?</span></font> </p>

<p style=3D'margin-left:.5in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>&nbsp;Thank you,</span></font> </p>

<p style=3D'margin-left:.5in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>&nbsp;Chintan</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;ipNetFusion =
Inc,</span></font>
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;Dallas, Tx - =
75080</span></font>
</p>

<p class=3DMsoNormal =
style=3D'margin-right:0in;margin-bottom:12.0pt;margin-left:
.5in'><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<p style=3D'margin-left:.5in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>______________________________________________=
_</span></font>
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>Sip mailing list&nbsp; =
<a
href=3D"https://www1.ietf.org/mailman/listinfo/sip" =
target=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</a></span></=
font>
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>This list is for NEW =
development of
the core SIP Protocol</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>Use
sip-implementors@cs.columbia.edu for questions on current =
sip</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>Use sipping@ietf.org for =
new
developments on the application of sip</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 style=3D'margin-left:.5in'><b><font size=3D1 face=3D"Times New =
Roman"><span
style=3D'font-size:7.5pt;font-weight:bold'>For Aculab's privacy policy =
and E-Mail
disclaimer, Click Here : <a
href=3D"http://www.aculab.com/company/legal_notice.htm" =
target=3D"_blank">http://www.aculab.com/company/legal_notice.htm</a>.
</span></font></b></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>

</body>

</html>

------_=_NextPart_001_01C3F1B9.F1565D9E--

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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] RE: [Sip-implementors] Via: processing
Date: Fri, 13 Feb 2004 04:36:59 +0530
Message-ID: <10C4348A1BA43A4FA8D5B767E052CEADEDB398@blr-ec-msg04.wipro.com>
Thread-Topic: [Sip] RE: [Sip-implementors] Via: processing
Thread-Index: AcPnT4Kg0BWKLk73SbWHQuKZwWUX4AKbAfvg
To: <Paul.D.Smith@dataconnection.com>, <brett@broadsoft.com>
Cc: <sip@ietf.org>
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I guess putting FQDN or IP addresses in the Via header driven by the
tradeoff between overhead in DNS query for FQDN name but provides
robustness or direct usage of IP address which lead to undeliverable in
case intermediate entities failure.=20

It's my feeling, please correct me if I am wrong....

Thanks & Regards,
Nataraju A.B.=20


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Paul
D.Smith
Sent: Friday, January 30, 2004 9:56 PM
To: 'brett@broadsoft.com'
Cc: SIP WG
Subject: [Sip] RE: [Sip-implementors] Via: processing

Brett, Arlie,

RFC3261 clearly allows a response to be "rerouted" if it can be
determined
that the response did not reach the first-attempt target.  For example,
if
the request is received over TCP, and the connection has gone and cannot
be
reestablished by the time the response is ready to be sent back, RFC3261
states the following

--- from RFC3261, section 18.2.2, 1st bullet.

If that connection attempt fails, the server SHOULD use the procedures
in
[4] for servers in order to determine the IP address and port to open
the
connection and send the response to.

--- end of cut

RFC3263 then gives an algorithm for attempting to determine alternate IP
addresses to which the response might be sent.  Clearly having an FQDN
in
the sent-by will give a chance of success which having an IP address
will
not, especially if the sent-by and received IP addresses are the same!

Paul DS.

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: Brett Tate [mailto:brett@broadsoft.com]
Sent: 29 January 2004 00:27
To: SIP (E-mail)
Subject: RE: [Sip-implementors] Via: processing


> I'm looking at the spec for Via: processing. =20
> RFC 3261 clearly shows that a Via: line can=20
> contain DNS names, as well as textual
> representations of network addresses.
>=20
> How many products actually implement support=20
> for resolving DNS names provided in Via: lines? =20

Not very many support DNS resolution of a
Via entry.  However many support the concept
of adding the Via's received parameter when
it does not match the Via's sent-by address.

> This seems...  well, it seems unnecessary,
> potentially insecure (allowing for easy DDoS=20
> and potentially amplification attacks), and=20
> just a general nuisance.  Is this kind of
> thing ever actually used for a good purpose? =20

Some people have toyed with the idea of
advancing response addresses based upon
rfc3263 during failure situations.

> Does any product do anything *but* just put=20
> real network addresses in Via: lines in
> requests?

Yes.

_______________________________________________
Sip-implementors mailing list
Sip-implementors@cs.columbia.edu
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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] RE: [Sip-implementors] Via: processing
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        <brett@broadsoft.com>
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Hi
   actually putting IP address in Via is not recommended when the SIP
message has to traverse through NAT. So it is better to have FQDN
instead of IP.=20

Ranjit





-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
Nataraju Alilaghatta (WT01 - TELECOM & INTER-NETWORKING SOLUTIONS)
Sent: Friday, February 13, 2004 4:37 AM
To: Paul.D.Smith@dataconnection.com; brett@broadsoft.com
Cc: sip@ietf.org
Subject: RE: [Sip] RE: [Sip-implementors] Via: processing


I guess putting FQDN or IP addresses in the Via header driven by the
tradeoff between overhead in DNS query for FQDN name but provides
robustness or direct usage of IP address which lead to undeliverable in
case intermediate entities failure.=20

It's my feeling, please correct me if I am wrong....

Thanks & Regards,
Nataraju A.B.=20


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Paul
D.Smith
Sent: Friday, January 30, 2004 9:56 PM
To: 'brett@broadsoft.com'
Cc: SIP WG
Subject: [Sip] RE: [Sip-implementors] Via: processing

Brett, Arlie,

RFC3261 clearly allows a response to be "rerouted" if it can be
determined that the response did not reach the first-attempt target.
For example, if the request is received over TCP, and the connection has
gone and cannot be reestablished by the time the response is ready to be
sent back, RFC3261 states the following

--- from RFC3261, section 18.2.2, 1st bullet.

If that connection attempt fails, the server SHOULD use the procedures
in [4] for servers in order to determine the IP address and port to open
the connection and send the response to.

--- end of cut

RFC3263 then gives an algorithm for attempting to determine alternate IP
addresses to which the response might be sent.  Clearly having an FQDN
in the sent-by will give a chance of success which having an IP address
will not, especially if the sent-by and received IP addresses are the
same!

Paul DS.

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: Brett Tate [mailto:brett@broadsoft.com]
Sent: 29 January 2004 00:27
To: SIP (E-mail)
Subject: RE: [Sip-implementors] Via: processing


> I'm looking at the spec for Via: processing.
> RFC 3261 clearly shows that a Via: line can=20
> contain DNS names, as well as textual
> representations of network addresses.
>=20
> How many products actually implement support
> for resolving DNS names provided in Via: lines? =20

Not very many support DNS resolution of a
Via entry.  However many support the concept
of adding the Via's received parameter when
it does not match the Via's sent-by address.

> This seems...  well, it seems unnecessary,
> potentially insecure (allowing for easy DDoS
> and potentially amplification attacks), and=20
> just a general nuisance.  Is this kind of
> thing ever actually used for a good purpose? =20

Some people have toyed with the idea of
advancing response addresses based upon
rfc3263 during failure situations.

> Does any product do anything *but* just put
> real network addresses in Via: lines in
> requests?

Yes.

_______________________________________________
Sip-implementors mailing list
Sip-implementors@cs.columbia.edu
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

_______________________________________________
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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To: nataraju.alilaghatta@wipro.com
Cc: sip@ietf.org, cthakker@ipnetfusion.com
Subject: RE: [Sip] Record-Route and Contact
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I was under the impression that the Route header is to be used in order that
UAs can influence the route taken (subsequent to initial INVITE) to include
particular proxy/proxies. To me, that's what 20.34 (of 3261) implies.
 
Are you saying that somewhere in 3261 it states that the Contact info in the
200-INVITE be used to build a "Route:" header for later insertion into, for
example, an ACK request?   

-----Original Message-----
From: nataraju.alilaghatta@wipro.com [mailto:nataraju.alilaghatta@wipro.com]
Sent: 12 February 2004 22:46
To: Matthew.Gardiner@aculab.com; cthakker@ipnetfusion.com
Cc: sip@ietf.org
Subject: RE: [Sip] Record-Route and Contact



 

 

Thanks & Regards,
Nataraju A.B. 

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
Matthew.Gardiner@aculab.com
Sent: Thursday, February 12, 2004 1:09 PM
To: cthakker@ipnetfusion.com
Cc: sip@ietf.org
Subject: RE: [Sip] Record-Route and Contact

 

It's my understanding that Record-Route headers take precedence over the
IP/FQDN in the contact header of the response, as they (Record-Route
headers) are inserted by specifically by the proxies in order to influence
the route that subsequent messaging takes. They influence, not only the
route taken by the ACK, furthermore that taken by any subsequent requests.

[ABN] once 200 for INVITE is received it forms the Route header for further
requests sent in the same dialog. The logic to form Route header is been
clearly mentioned in the Rfc3261 and the contact information in 200-INVITE
forms the Req-URI for subsequent requests. 

-----Original Message----- 
From: cthakker [ mailto:cthakker@ipnetfusion.com
<mailto:cthakker@ipnetfusion.com> ] 
Sent: 11 February 2004 21:05 
To: sip@ietf.org 
Subject: [Sip] Record-Route and Contact 

 

Hi all, 
 If a UAC generates a request in Dialog (say an ACK for 200 it received), 
  
 If the 200 had a Record-Route with loose routing and also a Contact 
header where should it route the request ? Can it route it directly to 
the IP/FQDN specified in the uri of Contact ? Does any header take 
precedence over other when generating a subsequent request in a Dialog ? 
 Is the significance of Contact header end-to-end or between UAs. For 
the example given in the RFC 3261 (alice and bob), can the intermediate 
proxies modify the value of the Contact header in the response sent by 
Alice to Bob ? 

 Thank you, 

 Chintan 
 ipNetFusion Inc, 
 Dallas, Tx - 75080 

 

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<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=034075007-13022004>I was 
under the impression that the Route header is to be used in order that UAs 
can&nbsp;influence the route taken (subsequent to initial INVITE) 
to&nbsp;include particular proxy/proxies. To me, that's what 20.34 (of 3261) 
implies.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=034075007-13022004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=034075007-13022004>Are 
you saying that somewhere in 3261 it states that the Contact info in the 
200-INVITE be used to build a "Route:" header for later insertion into,&nbsp;for 
example, an ACK request?&nbsp;&nbsp;&nbsp;</SPAN></FONT></DIV>
<BLOCKQUOTE>
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> 
  nataraju.alilaghatta@wipro.com 
  [mailto:nataraju.alilaghatta@wipro.com]<BR><B>Sent:</B> 12 February 2004 
  22:46<BR><B>To:</B> Matthew.Gardiner@aculab.com; 
  cthakker@ipnetfusion.com<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> RE: 
  [Sip] Record-Route and Contact<BR><BR></DIV></FONT>
  <DIV class=Section1>
  <P class=MsoNormal><FONT color=blue face="Courier New" size=2><SPAN 
  style="COLOR: blue; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp;</SPAN></FONT></P>
  <P class=MsoNormal><FONT color=blue face="Courier New" size=2><SPAN 
  style="COLOR: blue; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp;</SPAN></FONT></P>
  <DIV>
  <P><FONT color=blue face="Courier New" size=2><SPAN 
  style="COLOR: blue; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Thanks &amp; 
  Regards,<BR>Nataraju A.B. </SPAN></FONT></P></DIV>
  <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face=Tahoma size=2><SPAN 
  style="FONT-FAMILY: Tahoma; FONT-SIZE: 10pt">-----Original 
  Message-----<BR><B><SPAN style="FONT-WEIGHT: bold">From:</SPAN></B> 
  sip-admin@ietf.org [mailto:sip-admin@ietf.org] <B><SPAN 
  style="FONT-WEIGHT: bold">On Behalf Of 
  </SPAN></B>Matthew.Gardiner@aculab.com<BR><B><SPAN 
  style="FONT-WEIGHT: bold">Sent:</SPAN></B> </SPAN></FONT><FONT face=Tahoma 
  size=2><SPAN style="FONT-FAMILY: Tahoma; FONT-SIZE: 10pt">Thursday, February 
  12, 2004</SPAN></FONT><FONT face=Tahoma size=2><SPAN 
  style="FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"> </SPAN></FONT><FONT face=Tahoma 
  size=2><SPAN style="FONT-FAMILY: Tahoma; FONT-SIZE: 10pt">1:09 
  PM</SPAN></FONT><FONT face=Tahoma size=2><SPAN 
  style="FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"><BR><B><SPAN 
  style="FONT-WEIGHT: bold">To:</SPAN></B> cthakker@ipnetfusion.com<BR><B><SPAN 
  style="FONT-WEIGHT: bold">Cc:</SPAN></B> sip@ietf.org<BR><B><SPAN 
  style="FONT-WEIGHT: bold">Subject:</SPAN></B> RE: [Sip] Record-Route and 
  Contact</SPAN></FONT></P>
  <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" 
  size=3><SPAN style="FONT-SIZE: 12pt">&nbsp;</SPAN></FONT></P>
  <P style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" size=2><SPAN 
  style="FONT-SIZE: 10pt">It's my understanding that Record-Route headers take 
  precedence over the IP/FQDN in the contact header of the response, as they 
  (Record-Route headers) are inserted by specifically by the proxies in order to 
  influence the route that subsequent messaging takes. They influence, not only 
  the route taken by the ACK, furthermore that taken by any subsequent 
  requests.</SPAN></FONT></P>
  <P style="MARGIN-LEFT: 0.5in"><FONT color=blue face="Courier New" size=2><SPAN 
  style="COLOR: blue; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">[ABN] once 
  200 for INVITE is received it forms the Route header for further requests sent 
  in the same dialog. The logic to form Route header is been clearly mentioned 
  in the Rfc3261 and the contact information in 200-INVITE forms the Req-URI for 
  subsequent requests. </SPAN></FONT></P>
  <P style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" size=2><SPAN 
  style="FONT-SIZE: 10pt">-----Original Message-----</SPAN></FONT> <BR><FONT 
  size=2><SPAN style="FONT-SIZE: 10pt">From: cthakker [<A 
  href="mailto:cthakker@ipnetfusion.com">mailto:cthakker@ipnetfusion.com</A>]</SPAN></FONT> 
  <BR><FONT size=2><SPAN style="FONT-SIZE: 10pt">Sent: </SPAN></FONT><FONT 
  size=2><SPAN style="FONT-SIZE: 10pt">11 February 2004</SPAN></FONT><FONT 
  size=2><SPAN style="FONT-SIZE: 10pt"> </SPAN></FONT><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">21:05</SPAN></FONT> <BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">To: sip@ietf.org</SPAN></FONT> <BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">Subject: [Sip] Record-Route and Contact</SPAN></FONT> 
  </P>
  <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" 
  size=3><SPAN style="FONT-SIZE: 12pt">&nbsp;</SPAN></FONT></P>
  <P style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" size=2><SPAN 
  style="FONT-SIZE: 10pt">Hi all,</SPAN></FONT> <BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">&nbsp;If a UAC generates a request in Dialog (say an 
  ACK for 200 it received),</SPAN></FONT> <BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">&nbsp;</SPAN></FONT> <BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">&nbsp;If the 200 had a Record-Route with loose routing 
  and also a Contact </SPAN></FONT><BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">header where should it route the request ? Can it 
  route it directly to </SPAN></FONT><BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">the IP/FQDN specified in the uri of Contact ? Does any 
  header take </SPAN></FONT><BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">precedence over other when generating a subsequent 
  request in a Dialog ?</SPAN></FONT> <BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">&nbsp;Is the significance of Contact header end-to-end 
  or between UAs. For </SPAN></FONT><BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">the example given in the RFC 3261 (alice and bob), can 
  the intermediate </SPAN></FONT><BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">proxies modify the value of the Contact header in the 
  response sent by </SPAN></FONT><BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">Alice to Bob ?</SPAN></FONT> </P>
  <P style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" size=2><SPAN 
  style="FONT-SIZE: 10pt">&nbsp;Thank you,</SPAN></FONT> </P>
  <P style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" size=2><SPAN 
  style="FONT-SIZE: 10pt">&nbsp;Chintan</SPAN></FONT> <BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">&nbsp;ipNetFusion Inc,</SPAN></FONT> <BR><FONT 
  size=2><SPAN style="FONT-SIZE: 10pt">&nbsp;Dallas, Tx - 75080</SPAN></FONT> 
  </P>
  <P class=MsoNormal 
  style="MARGIN-BOTTOM: 12pt; MARGIN-LEFT: 0.5in; MARGIN-RIGHT: 0in"><FONT 
  face="Times New Roman" size=3><SPAN 
  style="FONT-SIZE: 12pt">&nbsp;</SPAN></FONT></P>
  <P style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" size=2><SPAN 
  style="FONT-SIZE: 10pt">_______________________________________________</SPAN></FONT> 
  <BR><FONT size=2><SPAN style="FONT-SIZE: 10pt">Sip mailing list&nbsp; <A 
  href="https://www1.ietf.org/mailman/listinfo/sip" 
  target=_blank>https://www1.ietf.org/mailman/listinfo/sip</A></SPAN></FONT> 
  <BR><FONT size=2><SPAN style="FONT-SIZE: 10pt">This list is for NEW 
  development of the core SIP Protocol</SPAN></FONT> <BR><FONT size=2><SPAN 
  style="FONT-SIZE: 10pt">Use sip-implementors@cs.columbia.edu for questions on 
  current sip</SPAN></FONT> <BR><FONT size=2><SPAN style="FONT-SIZE: 10pt">Use 
  sipping@ietf.org for new developments on the application of sip</SPAN></FONT> 
  </P>
  <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" 
  size=3><SPAN style="FONT-SIZE: 12pt">&nbsp;</SPAN></FONT></P>
  <P style="MARGIN-LEFT: 0.5in"><B><FONT face="Times New Roman" size=1><SPAN 
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  E-Mail disclaimer, Click Here : <A 
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  <P class=MsoNormal style="MARGIN-LEFT: 0.5in"><FONT face="Times New Roman" 
  size=3><SPAN style="FONT-SIZE: 12pt">&nbsp;</SPAN></FONT></P></DIV>
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Subject: RE: [Sip] Record-Route and Contact
Date: Fri, 13 Feb 2004 14:08:53 +0530
Message-ID: <10C4348A1BA43A4FA8D5B767E052CEADF31212@blr-ec-msg04.wipro.com>
Thread-Topic: [Sip] Record-Route and Contact
Thread-Index: AcPyCE8FovK5WHe6RrGr2qgkEbRDBQAASV8w
To: <Matthew.Gardiner@aculab.com>
Cc: <sip@ietf.org>, <cthakker@ipnetfusion.com>
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=20

Thanks & Regards,
Nataraju A.B.=20

-----Original Message-----
From: Matthew.Gardiner@aculab.com [mailto:Matthew.Gardiner@aculab.com]=20
Sent: Friday, February 13, 2004 1:27 PM
To: Nataraju Alilaghatta (WT01 - TELECOM & INTER-NETWORKING SOLUTIONS)
Cc: sip@ietf.org; cthakker@ipnetfusion.com
Subject: RE: [Sip] Record-Route and Contact

=20

I was under the impression that the Route header is to be used in order
that UAs can influence the route taken (subsequent to initial INVITE) to
include particular proxy/proxies. To me, that's what 20.34 (of 3261)
implies.

=20

 [ABN] You are right...

=20

Are you saying that somewhere in 3261 it states that the Contact info in
the 200-INVITE be used to build a "Route:" header for later insertion
into, for example, an ACK request?  =20

     =20

[ABN] No... The contact serves the initial Req-URI for subsequent
requests within dialog but then again while the finding next hop's
address Route header (if there is one) takes precedence over Req-Uri.
The top route header might modify the Req-URI if top route header is a
strict router.

=20

=20

-----Original Message-----
From: nataraju.alilaghatta@wipro.com
[mailto:nataraju.alilaghatta@wipro.com]
Sent: 12 February 2004 22:46
To: Matthew.Gardiner@aculab.com; cthakker@ipnetfusion.com
Cc: sip@ietf.org
Subject: RE: [Sip] Record-Route and Contact

=20

=20

Thanks & Regards,
Nataraju A.B.=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
Matthew.Gardiner@aculab.com
Sent: Thursday, February 12, 2004 1:09 PM
To: cthakker@ipnetfusion.com
Cc: sip@ietf.org
Subject: RE: [Sip] Record-Route and Contact

=20

It's my understanding that Record-Route headers take precedence over the
IP/FQDN in the contact header of the response, as they (Record-Route
headers) are inserted by specifically by the proxies in order to
influence the route that subsequent messaging takes. They influence, not
only the route taken by the ACK, furthermore that taken by any
subsequent requests.

[ABN] once 200 for INVITE is received it forms the Route header for
further requests sent in the same dialog. The logic to form Route header
is been clearly mentioned in the Rfc3261 and the contact information in
200-INVITE forms the Req-URI for subsequent requests.=20

-----Original Message-----=20
From: cthakker [mailto:cthakker@ipnetfusion.com]=20
Sent: 11 February 2004 21:05=20
To: sip@ietf.org=20
Subject: [Sip] Record-Route and Contact=20

=20

Hi all,=20
 If a UAC generates a request in Dialog (say an ACK for 200 it
received),=20
 =20
 If the 200 had a Record-Route with loose routing and also a Contact=20
header where should it route the request ? Can it route it directly to=20
the IP/FQDN specified in the uri of Contact ? Does any header take=20
precedence over other when generating a subsequent request in a Dialog ?

 Is the significance of Contact header end-to-end or between UAs. For=20
the example given in the RFC 3261 (alice and bob), can the intermediate=20
proxies modify the value of the Contact header in the response sent by=20
Alice to Bob ?=20

 Thank you,=20

 Chintan=20
 ipNetFusion Inc,=20
 Dallas, Tx - 75080=20

=20

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

=20

For Aculab's privacy policy and E-Mail disclaimer, Click Here :
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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>&nbsp;</span></font></p>

<div>

<p><font size=3D2 color=3Dblue face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New";color:blue'>Thanks &amp; Regards,<br>
Nataraju A.B. </span></font></p>

</div>

<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> </span></font><font =
size=3D2
 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>Matthew.Gardiner@aculab.com=
</span></font><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> =
[mailto:</span></font><font
 size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>Matthew.Gardiner@aculab.com=
</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'>Sent:</span></b> </span></font><font =
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>Friday,
 February 13, 2004</span></font><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> </span></font><font
 size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>1:27 PM</span></font><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'><br>
<b><span style=3D'font-weight:bold'>To:</span></b> </span></font><font =
size=3D2
 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>Nataraju
 Alilaghatta</span></font><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma'> (WT01 - TELECOM &amp; INTER-NETWORKING =
SOLUTIONS)<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sip@ietf.org; =
</span></font><font
 size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>cthakker@ipnetfusion.com</s=
pan></font><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [Sip] =
Record-Route
and Contact</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=3Dblue face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>I was under the
impression that the Route header is to be used in order that UAs
can&nbsp;influence the route taken (subsequent to initial INVITE)
to&nbsp;include particular proxy/proxies. To me, that's what 20.34 (of =
3261)
implies.</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>&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;<b><i><font color=3Dblue><span =
style=3D'color:blue;
font-weight:bold;font-style:italic'>[ABN] </span></font></i></b><font
color=3Dblue><span style=3D'color:blue'>You are =
right&#8230;</span></font></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>&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'>Are you saying =
that
somewhere in 3261 it states that the Contact info in the 200-INVITE be =
used to
build a &quot;Route:&quot; header for later insertion into,&nbsp;for =
example,
an ACK request?&nbsp;&nbsp;&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue
face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>[ABN] </span></font><font size=3D2 color=3Dred =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:red'>No...</span></font><font
size=3D2 color=3Dblue face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:
"Courier New";color:blue'> The contact serves the initial Req-URI for
subsequent requests within dialog but then again while the finding next
hop&#8217;s address Route header (if there is one) takes precedence over =
Req-Uri.
The top route header might modify the Req-URI if top route header is a =
strict
router.</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue
face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue
face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>&nbsp;</span></font></p>

</div>

<blockquote style=3D'margin-top:5.0pt;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>
nataraju.alilaghatta@wipro.com =
[mailto:nataraju.alilaghatta@wipro.com]<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> </span></font><font =
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>12 February
 2004</span></font><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma'> </span></font><font size=3D2 face=3DTahoma><span
 style=3D'font-size:10.0pt;font-family:Tahoma'>22:46</span></font><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'><br>
<b><span style=3D'font-weight:bold'>To:</span></b> </span></font><font =
size=3D2
 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>Matthew.Gardiner@aculab.com=
</span></font><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>; </span></font><font
 size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>cthakker@ipnetfusion.com</s=
pan></font><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'><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] =
Record-Route
and Contact</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue
face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
color=3Dblue
face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>&nbsp;</span></font></p>

<div>

<p style=3D'margin-left:.5in'><font size=3D2 color=3Dblue =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>Thanks =
&amp;
Regards,<br>
Nataraju A.B. </span></font></p>

</div>

<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></span></font><font
 size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>Matthew.Gardiner@aculab.com=
</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'>Sent:</span></b> Thursday, February =
12, 2004
1:09 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> </span></font><font =
size=3D2
 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>cthakker@ipnetfusion.com</s=
pan></font><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'><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] =
Record-Route
and Contact</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>

<p style=3D'margin-left:1.0in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>It's my understanding that Record-Route =
headers take
precedence over the IP/FQDN in the contact header of the response, as =
they
(Record-Route headers) are inserted by specifically by the proxies in =
order to
influence the route that subsequent messaging takes. They influence, not =
only
the route taken by the ACK, furthermore that taken by any subsequent =
requests.</span></font></p>

<p style=3D'margin-left:1.0in'><font size=3D2 color=3Dblue =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>[ABN] =
once 200 for
INVITE is received it forms the Route header for further requests sent =
in the
same dialog. The logic to form Route header is been clearly mentioned in =
the
Rfc3261 and the contact information in 200-INVITE forms the Req-URI for
subsequent requests. </span></font></p>

<p style=3D'margin-left:1.0in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>-----Original Message-----</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>From: cthakker [<a
href=3D"mailto:cthakker@ipnetfusion.com">mailto:cthakker@ipnetfusion.com<=
/a>]</span></font>
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>Sent: 11 February 2004 =
21:05</span></font>
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>To: =
sip@ietf.org</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>Subject: [Sip] =
Record-Route and
Contact</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>

<p style=3D'margin-left:1.0in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>Hi all,</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;If a UAC generates =
a request
in Dialog (say an ACK for 200 it received),</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;</span></font> =
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;If the 200 had a =
Record-Route
with loose routing and also a Contact </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>header where should it =
route the
request ? Can it route it directly to </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>the IP/FQDN specified in =
the uri of
Contact ? Does any header take </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>precedence over other =
when
generating a subsequent request in a Dialog ?</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;Is the =
significance of
Contact header end-to-end or between UAs. For </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>the example given in the =
RFC 3261
(alice and bob), can the intermediate </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>proxies modify the value =
of the
Contact header in the response sent by </span></font><br>
<font size=3D2><span style=3D'font-size:10.0pt'>Alice to Bob =
?</span></font> </p>

<p style=3D'margin-left:1.0in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>&nbsp;Thank you,</span></font> </p>

<p style=3D'margin-left:1.0in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>&nbsp;Chintan</span></font> <br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;ipNetFusion =
Inc,</span></font>
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>&nbsp;Dallas, Tx - =
75080</span></font>
</p>

<p class=3DMsoNormal =
style=3D'margin-right:0in;margin-bottom:12.0pt;margin-left:
1.0in'><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<p style=3D'margin-left:1.0in'><font size=3D2 face=3D"Times New =
Roman"><span
style=3D'font-size:10.0pt'>______________________________________________=
_</span></font>
<br>
<font size=3D2><span style=3D'font-size:10.0pt'>Sip mailing list&nbsp; =
<a
href=3D"https://www1.ietf.org/mailman/listinfo/sip" =
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From exim@www1.ietf.org  Fri Feb 13 10:41:54 2004
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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] RE: [Sip-implementors] Via: processing
Date: Fri, 13 Feb 2004 10:41:40 -0500
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> I guess putting FQDN or IP addresses in 
> the Via header driven by the tradeoff between 
> overhead in DNS query for FQDN name but provides
> robustness or direct usage of IP address which 
> lead to undeliverable in case intermediate 
> entities failure. 
> 
> It's my feeling, please correct me if I am wrong....

The Via's sent-by FQDN only needs to be
resolved during failure situations since
the locally added "received" parameter 
would be tried before the sent-by.

Thus providing an FQDN within the Via's
sent-by does not really cause any
overhead beyond the extra "received"
parameter getting added to proxied Via 
entries.


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From exim@www1.ietf.org  Fri Feb 13 11:12:41 2004
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Subject: [Sip] draft-ietf-sip-publish-02.txt comments
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Folks,

Current version of SIP-PUBLISH  as  specified in 
draft-ietf-sip-publish-02.txt leaves out
composer policy as out-of-scope.  Can this lead to interoperability 
issues? If so, I suggest
the draft defines some basic composer policy structure.

Regards,
Alex.


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From exim@www1.ietf.org  Fri Feb 13 13:26:07 2004
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        nataraju.alilaghatta@wipro.com
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Subject: RE: [Sip] Record-Route and Contact
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I believe Contact info is used to build a request URI and same set of URI's
in Record-Route are used to build a Route header. If there is no Contact,
the URI's in Record Route are used to build request URI.

 

Regards,

Anil

 

This email contains material that is confidential. The content of this email
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-----Original Message-----
From: Matthew.Gardiner@aculab.com [mailto:Matthew.Gardiner@aculab.com] 
Sent: Thursday, February 12, 2004 11:57 PM
To: nataraju.alilaghatta@wipro.com
Cc: sip@ietf.org; cthakker@ipnetfusion.com
Subject: RE: [Sip] Record-Route and Contact

 

I was under the impression that the Route header is to be used in order that
UAs can influence the route taken (subsequent to initial INVITE) to include
particular proxy/proxies. To me, that's what 20.34 (of 3261) implies.

 

Are you saying that somewhere in 3261 it states that the Contact info in the
200-INVITE be used to build a "Route:" header for later insertion into, for
example, an ACK request?   

-----Original Message-----
From: nataraju.alilaghatta@wipro.com [mailto:nataraju.alilaghatta@wipro.com]
Sent: 12 February 2004 22:46
To: Matthew.Gardiner@aculab.com; cthakker@ipnetfusion.com
Cc: sip@ietf.org
Subject: RE: [Sip] Record-Route and Contact

 

 

Thanks & Regards,
Nataraju A.B. 

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
Matthew.Gardiner@aculab.com
Sent: Thursday, February 12, 2004 1:09 PM
To: cthakker@ipnetfusion.com
Cc: sip@ietf.org
Subject: RE: [Sip] Record-Route and Contact

 

It's my understanding that Record-Route headers take precedence over the
IP/FQDN in the contact header of the response, as they (Record-Route
headers) are inserted by specifically by the proxies in order to influence
the route that subsequent messaging takes. They influence, not only the
route taken by the ACK, furthermore that taken by any subsequent requests.

[ABN] once 200 for INVITE is received it forms the Route header for further
requests sent in the same dialog. The logic to form Route header is been
clearly mentioned in the Rfc3261 and the contact information in 200-INVITE
forms the Req-URI for subsequent requests. 

-----Original Message----- 
From: cthakker [mailto:cthakker@ipnetfusion.com
<mailto:cthakker@ipnetfusion.com> ] 
Sent: 11 February 2004 21:05 
To: sip@ietf.org 
Subject: [Sip] Record-Route and Contact 

 

Hi all, 
 If a UAC generates a request in Dialog (say an ACK for 200 it received), 
  
 If the 200 had a Record-Route with loose routing and also a Contact 
header where should it route the request ? Can it route it directly to 
the IP/FQDN specified in the uri of Contact ? Does any header take 
precedence over other when generating a subsequent request in a Dialog ? 
 Is the significance of Contact header end-to-end or between UAs. For 
the example given in the RFC 3261 (alice and bob), can the intermediate 
proxies modify the value of the Contact header in the response sent by 
Alice to Bob ? 

 Thank you, 

 Chintan 
 ipNetFusion Inc, 
 Dallas, Tx - 75080 

 

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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I believe Contact info is used to build a
request URI and same set of URI's in Record-Route are used to build a
Route header. If there is no Contact, the URI's in </span></font><font
  size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
  color:navy'>Record Route</span></font><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'> are used to build
request URI.</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Regards,</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Anil</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<div>

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<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b> Matthew.Gardiner@aculab.com
[mailto:Matthew.Gardiner@aculab.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, February 12, 2004
11:57 PM<br>
<b><span style='font-weight:bold'>To:</span></b> nataraju.alilaghatta@wipro.com<br>
<b><span style='font-weight:bold'>Cc:</span></b> sip@ietf.org;
cthakker@ipnetfusion.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: [Sip] Record-Route
and Contact</span></font></p>

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

<div>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=blue face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:blue'>I was under the
impression that the Route header is to be used in order that UAs
can&nbsp;influence the route taken (subsequent to initial INVITE)
to&nbsp;include particular proxy/proxies. To me, that's what 20.34 (of 3261)
implies.</span></font></p>

</div>

<div>

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

</div>

<div>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=blue face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:blue'>Are you saying that
somewhere in 3261 it states that the Contact info in the 200-INVITE be used to
build a &quot;Route:&quot; header for later insertion into,&nbsp;for example,
an ACK request?&nbsp;&nbsp;&nbsp;</span></font></p>

</div>

<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>

<p class=MsoNormal style='margin-right:0in;margin-bottom:12.0pt;margin-left:
.5in'><font size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>-----Original
Message-----<br>
<b><span style='font-weight:bold'>From:</span></b>
nataraju.alilaghatta@wipro.com [mailto:nataraju.alilaghatta@wipro.com]<br>
<b><span style='font-weight:bold'>Sent:</span></b> 12 February 2004 22:46<br>
<b><span style='font-weight:bold'>To:</span></b> Matthew.Gardiner@aculab.com;
cthakker@ipnetfusion.com<br>
<b><span style='font-weight:bold'>Cc:</span></b> sip@ietf.org<br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: [Sip] Record-Route
and Contact</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=blue
face="Courier New"><span style='font-size:10.0pt;font-family:"Courier New";
color:blue'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=blue
face="Courier New"><span style='font-size:10.0pt;font-family:"Courier New";
color:blue'>&nbsp;</span></font></p>

<div>

<p style='margin-left:.5in'><font size=2 color=blue face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New";color:blue'>Thanks &amp;
Regards,<br>
Nataraju A.B. </span></font></p>

</div>

<p class=MsoNormal style='margin-left:1.0in'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b> sip-admin@ietf.org
[mailto:sip-admin@ietf.org] <b><span style='font-weight:bold'>On Behalf Of </span></b>Matthew.Gardiner@aculab.com<br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, February 12, 2004
1:09 PM<br>
<b><span style='font-weight:bold'>To:</span></b> cthakker@ipnetfusion.com<br>
<b><span style='font-weight:bold'>Cc:</span></b> sip@ietf.org<br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: [Sip] Record-Route
and Contact</span></font></p>

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

<p style='margin-left:1.0in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>It's my understanding that Record-Route headers take
precedence over the IP/FQDN in the contact header of the response, as they
(Record-Route headers) are inserted by specifically by the proxies in order to
influence the route that subsequent messaging takes. They influence, not only
the route taken by the ACK, furthermore that taken by any subsequent requests.</span></font></p>

<p style='margin-left:1.0in'><font size=2 color=blue face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New";color:blue'>[ABN] once 200
for INVITE is received it forms the Route header for further requests sent in
the same dialog. The logic to form Route header is been clearly mentioned in
the Rfc3261 and the contact information in 200-INVITE forms the Req-URI for
subsequent requests. </span></font></p>

<p style='margin-left:1.0in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>-----Original Message-----</span></font> <br>
<font size=2><span style='font-size:10.0pt'>From: cthakker [<a
href="mailto:cthakker@ipnetfusion.com">mailto:cthakker@ipnetfusion.com</a>]</span></font>
<br>
<font size=2><span style='font-size:10.0pt'>Sent: 11 February 2004 21:05</span></font>
<br>
<font size=2><span style='font-size:10.0pt'>To: sip@ietf.org</span></font> <br>
<font size=2><span style='font-size:10.0pt'>Subject: [Sip] Record-Route and
Contact</span></font> </p>

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

<p style='margin-left:1.0in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>Hi all,</span></font> <br>
<font size=2><span style='font-size:10.0pt'>&nbsp;If a UAC generates a request
in Dialog (say an ACK for 200 it received),</span></font> <br>
<font size=2><span style='font-size:10.0pt'>&nbsp;</span></font> <br>
<font size=2><span style='font-size:10.0pt'>&nbsp;If the 200 had a Record-Route
with loose routing and also a Contact </span></font><br>
<font size=2><span style='font-size:10.0pt'>header where should it route the
request ? Can it route it directly to </span></font><br>
<font size=2><span style='font-size:10.0pt'>the IP/FQDN specified in the uri of
Contact ? Does any header take </span></font><br>
<font size=2><span style='font-size:10.0pt'>precedence over other when
generating a subsequent request in a Dialog ?</span></font> <br>
<font size=2><span style='font-size:10.0pt'>&nbsp;Is the significance of
Contact header end-to-end or between UAs. For </span></font><br>
<font size=2><span style='font-size:10.0pt'>the example given in the RFC 3261
(alice and bob), can the intermediate </span></font><br>
<font size=2><span style='font-size:10.0pt'>proxies modify the value of the
Contact header in the response sent by </span></font><br>
<font size=2><span style='font-size:10.0pt'>Alice to Bob ?</span></font> </p>

<p style='margin-left:1.0in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>&nbsp;Thank you,</span></font> </p>

<p style='margin-left:1.0in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>&nbsp;Chintan</span></font> <br>
<font size=2><span style='font-size:10.0pt'>&nbsp;ipNetFusion Inc,</span></font>
<br>
<font size=2><span style='font-size:10.0pt'>&nbsp;Dallas, Tx - 75080</span></font>
</p>

<p class=MsoNormal style='margin-right:0in;margin-bottom:12.0pt;margin-left:
1.0in'><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>&nbsp;</span></font></p>

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<p class=MsoNormal style='margin-left:1.0in'><font size=3 face="Times New Roman"><span
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Dear All,

 

In draft-ietf-sip-nat-01.txt, there are SIP extensions for NAT traversal.
The received and rport parameter are used to send back the responses
correctly. However if we don't use these extensions, and change the via ip
address and port with that of firewall/port then the SIP responses will come
directly to firewall, in that case the firewall permits the responses. UA1
inside the network sends requests to firewall/NAT.

 

UA1 

INVITE sip:user@domain.com SIP/2.0 ----------------> Firewall/NAT
----------------------------------------------->UA2

Via: SIP/2.0/UDP 10.1.1.1:4540                          INVITE
sip:user@domain.com SIP/2.0                             

Via: SIP/2.0/UDP firewall_ip:5060

 
(Maintain mapping) 

 

the UA2 will send the responses to firewall/NAT and firewall/NAT will write
back the addresses in Via to original IP/port. I understand from the draft,
if there are no SIP ALG functionality in firewall, then the extensions are
used to make SIP traverse through NAT. However if we implement SIP ALG in
firewall then we can make SIP traverse through NAT. Can anybody tell that
this way is per SIP spec.

 

Thank you,

Anil

 

This email contains material that is confidential. The content of this email
is for the sole use of the intended recipient(s). Any review or distribution
by persons other than the intended recipient(s) without the express
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are not the intended recipient, please contact the sender and delete/destroy
all copies of this email and any related attachments. NetScreen does not
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<div class=3DSection1>

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

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>In draft-ietf-sip-nat-01.txt, there are SIP =
extensions for
NAT traversal. The received and rport parameter are used to send back =
the
responses correctly. However if we don't use these extensions, and =
change
the via ip address and port with that of firewall/port then the SIP =
responses
will come directly to firewall, in that case the firewall permits the
responses. UA1 inside the network sends requests to =
firewall/NAT.</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>UA1 </span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>INVITE sip:user@domain.com SIP/2.0 =
--------------</span></font><font
size=3D2 face=3DWingdings><span =
style=3D'font-size:10.0pt;font-family:Wingdings'>&agrave;</span></font><=
font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>
Firewall/NAT =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; ---------------------------------------------</span></font><font
size=3D2 face=3DWingdings><span =
style=3D'font-size:10.0pt;font-family:Wingdings'>&agrave;</span></font><=
font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>UA2</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Via: SIP/2.0/UDP =
10.1.1.1:4540&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; INVITE
sip:user@domain.com SIP/2.0 =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; </span></font></p>

<p class=3DMsoNormal style=3D'margin-left:2.5in;text-indent:.5in'><font =
size=3D2
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>Via: =
SIP/2.0/UDP firewall_ip:5060</span></font></p>

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

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>the UA2 will send the responses to firewall/NAT and
firewall/NAT will write back the addresses in Via to original IP/port. =
I
understand from the draft, if there are no SIP ALG functionality in =
firewall, then
the extensions are used to make SIP traverse through NAT. However if we
implement SIP ALG in firewall then we can make SIP traverse through =
NAT. Can
anybody tell that this way is per SIP spec.</span></font></p>

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

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

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

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

<p><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>This
email contains material that is confidential. The content of this email =
is for
the sole use of the intended recipient(s). Any review or distribution =
by
persons other than the intended recipient(s) without the express =
permission of
NetScreen Technologies, Inc. is strictly prohibited. If you are not the
intended recipient, please contact the sender and delete/destroy all =
copies of
this email and any related attachments. NetScreen does not guarantee =
the
accuracy or completeness of third party materials or =
information.</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>

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Date: Fri, 13 Feb 2004 22:58:18 +0200
From: Aki Niemi <aki.niemi@nokia.com>
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Subject: [Sip] Re: draft-ietf-sip-publish-02.txt comments
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Hi,

Inline.

ext Alex Audu wrote:
> Folks,
> 
> Current version of SIP-PUBLISH  as  specified in 
> draft-ietf-sip-publish-02.txt leaves out
> composer policy as out-of-scope.  Can this lead to interoperability 
> issues? If so, I suggest
> the draft defines some basic composer policy structure.

You're right in that the current draft leaves composer policy (both 
defining it or setting/changing it) out-of-scope. As to whether it leads 
to interop problems, probably not as far as PUBLISH is concerned.

So any interop problems would be in terms of PIDF and what the PUA is 
expecting the watchers to see when it publishes certain tuples. So this 
is very presence specific.

And for presence, it isn't really clear yet as to how the composer would 
go about resolving conflicts among the tuples coming from different 
sources. By not doing any sort of merging or selection among the tuples 
might work, so the default policy could be that a composer simply passes 
all tuples to watchers intact.

But I don't know how helpful, or indeed what the proper way to state 
this default policy would be. Any suggestions?

Cheers,
Aki



> Regards,
> Alex.
> 


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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 exim@www1.ietf.org  Fri Feb 13 17:00:43 2004
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Date: Fri, 13 Feb 2004 15:58:34 -0600
From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sip] Agenda Requests for SIP at IETF 59
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I've started layong out the SIP WG supplemental web site for IETF 59 at 
the usual place:

http://www.softarmor.com/sipwg/meets/ietf59/index.html

This page includes a link for "Agenda Requests", which I shall try to 
keep up-to-date as requests are made.

Remember -- we should use meeting time NOT to introduce new ideas, but 
to discuss issues that we just haven't been able to work out on the 
mailing list. If you have something that you think needs to be talked 
about in the meeting, please send me a request using the html template 
given below. Please include a link to the relevant draft, so that we can 
put together a "reading list".

Here's the template, which I blatantly copied from Gonzalo's note to 
SIPPING:

    <tr>
      <td><a href="mailto:Gonzalo.Camarillo@ericsson.com">
Gonzalo Camarillo</a></td>
      <td>Exploders</td>
      <td><center>5</center></td>
      <td>
<a 
href="http://www.ietf.org/internet-drafts/draft-camarillo-sipping-exploders-02.txt"> 

draft-camarillo-sipping-exploders-02.txt</a></td>
    </tr>


If the HTML here confuses you, fake it and I'll try and sort it out.

Thanks,

--
Dean


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



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Subject: [Sip] FW: Changing Via In NAT
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Content-Type: text/plain

Hi Guys,

 

  Can anybody have some comments about the discussion below.

 

Thanks in Advance,

Anil

 

This email contains material that is confidential. The content of this email
is for the sole use of the intended recipient(s). Any review or distribution
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-----Original Message-----
From: Anil Bollineni 
Sent: Friday, February 13, 2004 11:08 AM
To: 'sip@ietf.org'
Subject: Changing Via In NAT

 

Dear All,

 

In draft-ietf-sip-nat-01.txt, there are SIP extensions for NAT traversal.
The received and rport parameter are used to send back the responses
correctly. However if we don't use these extensions, and change the via ip
address and port with that of firewall/port then the SIP responses will come
directly to firewall, in that case the firewall permits the responses. UA1
inside the network sends requests to firewall/NAT.

 

UA1 

INVITE sip:user@domain.com SIP/2.0 ----------------> Firewall/NAT
----------------------------------------------->UA2

Via: SIP/2.0/UDP 10.1.1.1:4540                          INVITE
sip:user@domain.com SIP/2.0                             

Via: SIP/2.0/UDP firewall_ip:5060

 
(Maintain mapping) 

 

the UA2 will send the responses to firewall/NAT and firewall/NAT will write
back the addresses in Via to original IP/port. I understand from the draft,
if there are no SIP ALG functionality in firewall, then the extensions are
used to make SIP traverse through NAT. However if we implement SIP ALG in
firewall then we can make SIP traverse through NAT. Can anybody tell that
this way is per SIP spec.

 

Thank you,

Anil

 

This email contains material that is confidential. The content of this email
is for the sole use of the intended recipient(s). Any review or distribution
by persons other than the intended recipient(s) without the express
permission of NetScreen Technologies, Inc. is strictly prohibited. If you
are not the intended recipient, please contact the sender and delete/destroy
all copies of this email and any related attachments. NetScreen does not
guarantee the accuracy or completeness of third party materials or
information.

 


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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Hi Guys,</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp; Can anybody have some comments
about the discussion below.</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Thanks in Advance,</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Anil</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
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information.</span></font></p>

</div>

<p class=MsoNormal><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b> </span></font><font size=2
 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>Anil Bollineni</span></font><font
size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> <br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, February 13, 2004
11:08 AM<br>
<b><span style='font-weight:bold'>To:</span></b> 'sip@ietf.org'<br>
<b><span style='font-weight:bold'>Subject:</span></b> Changing Via In NAT</span></font></p>

<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Dear All,</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>In draft-ietf-sip-nat-01.txt, there are SIP extensions for
NAT traversal. The received and rport parameter are used to send back the
responses correctly. However if we don't use these extensions, and change
the via ip address and port with that of firewall/port then the SIP responses
will come directly to firewall, in that case the firewall permits the
responses. UA1 inside the network sends requests to firewall/NAT.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>UA1 </span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>INVITE sip:user@domain.com SIP/2.0 --------------</span></font><font
size=2 face=Wingdings><span style='font-size:10.0pt;font-family:Wingdings'>&agrave;</span></font><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>
Firewall/NAT
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
---------------------------------------------</span></font><font size=2
face=Wingdings><span style='font-size:10.0pt;font-family:Wingdings'>&agrave;</span></font><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>UA2</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Via: SIP/2.0/UDP
10.1.1.1:4540&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
INVITE sip:user@domain.com SIP/2.0 &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></font></p>

<p class=MsoNormal style='margin-left:2.5in;text-indent:.5in'><font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>Via: SIP/2.0/UDP
firewall_ip:5060</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
(Maintain mapping) </span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>the UA2 will send the responses to firewall/NAT and
firewall/NAT will write back the addresses in Via to original IP/port. I
understand from the draft, if there are no SIP ALG functionality in firewall,
then the extensions are used to make SIP traverse through NAT. However if we
implement SIP ALG in firewall then we can make SIP traverse through NAT. Can
anybody tell that this way is per SIP spec.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Thank you,</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Anil</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>This
email contains material that is confidential. The content of this email is for
the sole use of the intended recipient(s). Any review or distribution by
persons other than the intended recipient(s) without the express permission of
NetScreen Technologies, Inc. is strictly prohibited. If you are not the
intended recipient, please contact the sender and delete/destroy all copies of
this email and any related attachments. NetScreen does not guarantee the
accuracy or completeness of third party materials or information.</span></font></p>

<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

</body>

</html>

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Date: Fri, 13 Feb 2004 08:48:40 -0800
From: Cullen Jennings <fluffy@cisco.com>
To: <sip@ietf.org>
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I like this draft, just one little NIT, I think it would be better to call
the tag conf-policy-info instead of just policy-info

Cullen



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From: Cullen Jennings <fluffy@cisco.com>
To: <sip@ietf.org>
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I submitted a draft on binary encoding for SIP. This improves
interoperability, improves speed, decreases sizes of messages, and generally
makes life better for SIP which will no doubt eventually help with World
Hunger. I need feedback, please let me know:

a) Yah, lets do it

b) No this is a stupid ideas and here are the reasons why

c) There has not been many list comments, this draft involves some very
complex issues. I need a year or two to develop an opinion on the single
sentence inside it.

d) I agree with Cowboy Willis

Until it arrives in the archives, you can find it at

http://www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.html

or, if you prefer ASCII encodings of everything,

www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.txt

Thanks, Cullen







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From: Raphael Tryster <raphael@tdsoft.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>, sip@ietf.org
Subject: RE: [Sip] Binary Encoding for SIP
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Hello Cullen,

Given that this is my first venture into the SIP email forum, and the first
response to your posting, it is only fitting that I should vote for option
c:).

During the coming year or two, I hope to develop an opinion on whether the
issues of backward compatibility and ease of debugging, which seem to make
the binary option rather unpopular in Megaco, apply to SIP as well.  If they
do, it could have an adverse affect on World Hunger, as those with the right
tools to debug binary messages will eat more, at the expense of the rest.

Raphael Tryster

	-----Original Message-----
	From:	Cullen Jennings [SMTP:fluffy@cisco.com]
	Sent:	Sunday, 15 February 2004 9:03
	To:	sip@ietf.org
	Subject:	[Sip] Binary Encoding for SIP

	    
	I submitted a draft on binary encoding for SIP. This improves
	interoperability, improves speed, decreases sizes of messages, and
generally
	makes life better for SIP which will no doubt eventually help with
World
	Hunger. I need feedback, please let me know:

	a) Yah, lets do it

	b) No this is a stupid ideas and here are the reasons why

	c) There has not been many list comments, this draft involves some
very
	complex issues. I need a year or two to develop an opinion on the
single
	sentence inside it.

	d) I agree with Cowboy Willis

	Until it arrives in the archives, you can find it at

	
http://www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.html

	or, if you prefer ASCII encodings of everything,

	www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.txt

	Thanks, 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

------_=_NextPart_001_01C3F39B.A609DAD0
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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i">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version 5.5.2653.1=
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<TITLE>RE: [Sip] Binary Encoding for SIP</TITLE>
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<P><FONT SIZE=3D2 FACE=3D"Arial">Hello Cullen,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Given that this is my first venture into=
 the SIP email forum, and the first response to your posting, it is only =
fitting that I should vote for option c:).</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">During the coming year or two, I hope to=
 develop an opinion on whether the issues of backward compatibility and e=
ase of debugging, which seem to make the binary option rather unpopular i=
n Megaco, apply to SIP as well.&nbsp; If they do, it could have an advers=
e affect on World Hunger, as those with the right tools to debug binary m=
essages will eat more, at the expense of the rest.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Tahoma">Raphael Tryster</FONT>
</P>
<UL>
<P><A NAME=3D"_MailData"><FONT SIZE=3D2 FACE=3D"Arial">-----Original Mess=
age-----</FONT></A>
<BR><B><FONT SIZE=3D2 FACE=3D"Arial">From:&nbsp;&nbsp; Cullen Jennings [S=
MTP:fluffy@cisco.com]</FONT></B>
<BR><B><FONT SIZE=3D2 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT S=
IZE=3D2 FACE=3D"Arial">Sunday, 15 February 2004 9:03</FONT>
<BR><B><FONT SIZE=3D2 FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></=
B> <FONT SIZE=3D2 FACE=3D"Arial">sip@ietf.org</FONT>
<BR><B><FONT SIZE=3D2 FACE=3D"Arial">Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D2 FACE=3D"Arial">[Sip] Binary Enco=
ding for SIP</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">I submitted a draft on binary enc=
oding for SIP. This improves</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">interoperability, improves speed,=
 decreases sizes of messages, and generally</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">makes life better for SIP which w=
ill no doubt eventually help with World</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">Hunger. I need feedback, please l=
et me know:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">a) Yah, lets do it</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">b) No this is a stupid ideas and h=
ere are the reasons why</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">c) There has not been many list co=
mments, this draft involves some very</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">complex issues. I need a year or =
two to develop an opinion on the single</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">sentence inside it.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">d) I agree with Cowboy Willis</FON=
T>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Until it arrives in the archives, =
you can find it at</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New"><A HREF=3D"http://www.employees.or=
g/~fluffy/ietf/draft-jennings-sip-mime-01..html" TARGET=3D"_blank">http:/=
/www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.html</A></FONT=
>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">or, if you prefer ASCII encodings =
of everything,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">www.employees.org/~fluffy/ietf/dra=
ft-jennings-sip-mime-01.txt</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Thanks, Cullen</FONT>
</P>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">__________________________________=
_____________</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">Sip mailing list&nbsp; <A HREF=3D=
"https://www1.ietf.org/mailman/listinfo/sip" TARGET=3D"_blank">https://ww=
w1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">This list is for NEW development =
of the core SIP Protocol</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">Use sip-implementors@cs.columbia.=
edu for questions on current sip</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">Use sipping@ietf.org for new deve=
lopments on the application of sip</FONT>
</P>
</UL>
</BODY>
</HTML>

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_______________________________________________
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Cullen Jennings <fluffy@cisco.com>
CC: sip@ietf.org
Subject: Re: [Sip] Binary Encoding for SIP
References: <BC5459B9.31568%fluffy@cisco.com>
In-Reply-To: <BC5459B9.31568%fluffy@cisco.com>
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You should apply for a job at a local tabloid - headlining a discussion 
on the encoding of MIME body parts into the mother of all mailing list 
battles on protocol encoding.

Given the first reaction to the draft: this is only about the SIP body. 
Cullen is not proposing that SIP headers be coded in ASN.1 :-)

I'm for option (a), for the reasons you mention. The more interesting 
question is whether all SIP devices still need to be prepared to receive 
all the various encodings.

Cullen Jennings wrote:

>     
> I submitted a draft on binary encoding for SIP. This improves
> interoperability, improves speed, decreases sizes of messages, and generally
> makes life better for SIP which will no doubt eventually help with World
> Hunger. I need feedback, please let me know:
> 
> a) Yah, lets do it
> 
> b) No this is a stupid ideas and here are the reasons why
> 
> c) There has not been many list comments, this draft involves some very
> complex issues. I need a year or two to develop an opinion on the single
> sentence inside it.
> 
> d) I agree with Cowboy Willis
> 
> Until it arrives in the archives, you can find it at
> 
> http://www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.html
> 
> or, if you prefer ASCII encodings of everything,
> 
> www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.txt
> 
> Thanks, 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



From exim@www1.ietf.org  Sun Feb 15 12:06:48 2004
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To: Cullen Jennings <fluffy@cisco.com>
Cc: sip@ietf.org
Subject: Re: [Sip] Binary Encoding for SIP
References: <BC5459B9.31568%fluffy@cisco.com>
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I thought that binary body was already supported: RFC3261 section 7.4.1, 
third paragraph.

Cullen Jennings wrote:
>     
> I submitted a draft on binary encoding for SIP. This improves
> interoperability, improves speed, decreases sizes of messages, and generally
> makes life better for SIP which will no doubt eventually help with World
> Hunger. I need feedback, please let me know:
> 
> a) Yah, lets do it
> 
> b) No this is a stupid ideas and here are the reasons why
> 
> c) There has not been many list comments, this draft involves some very
> complex issues. I need a year or two to develop an opinion on the single
> sentence inside it.
> 
> d) I agree with Cowboy Willis
> 
> Until it arrives in the archives, you can find it at
> 
> http://www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.html
> 
> or, if you prefer ASCII encodings of everything,
> 
> www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.txt
> 
> Thanks, Cullen
> 
> 

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



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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
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Subject: [Sip] Updated History-Info draft submitted
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I have submitted an updated version of the History-Info draft.  Until it
becomes available in the respository, you can pick it up at:
http://home.comcast.net/~mary.barnes/IETF/draft-ietf-sip-history-info-02.txt

The primary changes are as follows:
o Merged the SIPPING WG requirements draft into this document. 
   Also, removed redirect server from ISSUER-req since the 
   solution identified this as not being required (or desirable).
  
o Added an explicit privacy requirement (PRIV-req-3) for the 
  proxy's role in recognizing and maintaining privacy associated 
  with a Request-URI being captured in History-Info due to local 
  policy. (Note, that the text was already there, it just wasn't 
  highlighted as an explicit requirement).  

o Removed the compact form for the header since unknown headers 
  with multiple entries would not be recognized.

o Added a summary of Application Considerations to address 
  concerns about the optional usage of History-Info. 

There are 2 open issues identified in the draft:
o Issue 1 is around the privacy issue that has been discussed 
  recently on the list:
http://www.ietf.org/mail-archive/working-groups/sip/current/msg09401.html
http://www.ietf.org/mail-archive/working-groups/sip/current/msg09371.html

  It has been suggested that adding an additional field to the 
  History-Info header (or extending the priv-values defined in 
  RFC 3323) would facilitate the implementation of a local privacy 
  policy associated with specific Request-URIs. 

o Issue 2 is around whether there needs to be some bounds on the size 
  or number of History-Info entries.  And if so, what's the mechanism
  for defining the bounds (e.g. limit the depth and breadth of the 
  tree by putting upper limits on the Index? )  

Regards,
Mary H. Barnes
mary.barnes@nortelnetworks.com
972-684-5432



------_=_NextPart_001_01C3F3F1.5B962DA4
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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>Updated History-Info draft submitted</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>I have submitted an updated version of the =
History-Info draft.&nbsp; Until it becomes available in the =
respository, you can pick it up at:</FONT></P>

<P><FONT SIZE=3D2><A =
HREF=3D"http://home.comcast.net/~mary.barnes/IETF/draft-ietf-sip-history=
-info-02.txt" =
TARGET=3D"_blank">http://home.comcast.net/~mary.barnes/IETF/draft-ietf-s=
ip-history-info-02.txt</A></FONT>
</P>

<P><FONT SIZE=3D2>The primary changes are as follows:</FONT>
<BR><FONT SIZE=3D2>o Merged the SIPPING WG requirements draft into this =
document. </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; Also, removed redirect server from =
ISSUER-req since the </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; solution identified this as not being =
required (or desirable).</FONT>
<BR><FONT SIZE=3D2>&nbsp; </FONT>
<BR><FONT SIZE=3D2>o Added an explicit privacy requirement (PRIV-req-3) =
for the </FONT>
<BR><FONT SIZE=3D2>&nbsp; proxy's role in recognizing and maintaining =
privacy associated </FONT>
<BR><FONT SIZE=3D2>&nbsp; with a Request-URI being captured in =
History-Info due to local </FONT>
<BR><FONT SIZE=3D2>&nbsp; policy. (Note, that the text was already =
there, it just wasn't </FONT>
<BR><FONT SIZE=3D2>&nbsp; highlighted as an explicit =
requirement).&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>o Removed the compact form for the header since =
unknown headers </FONT>
<BR><FONT SIZE=3D2>&nbsp; with multiple entries would not be =
recognized.</FONT>
</P>

<P><FONT SIZE=3D2>o Added a summary of Application Considerations to =
address </FONT>
<BR><FONT SIZE=3D2>&nbsp; concerns about the optional usage of =
History-Info. </FONT>
</P>

<P><FONT SIZE=3D2>There are 2 open issues identified in the =
draft:</FONT>
<BR><FONT SIZE=3D2>o Issue 1 is around the privacy issue that has been =
discussed </FONT>
<BR><FONT SIZE=3D2>&nbsp; recently on the list:</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/mail-archive/working-groups/sip/current/msg0=
9401.html" =
TARGET=3D"_blank">http://www.ietf.org/mail-archive/working-groups/sip/cu=
rrent/msg09401.html</A></FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/mail-archive/working-groups/sip/current/msg0=
9371.html" =
TARGET=3D"_blank">http://www.ietf.org/mail-archive/working-groups/sip/cu=
rrent/msg09371.html</A></FONT>
</P>

<P><FONT SIZE=3D2>&nbsp; It has been suggested that adding an =
additional field to the </FONT>
<BR><FONT SIZE=3D2>&nbsp; History-Info header (or extending the =
priv-values defined in </FONT>
<BR><FONT SIZE=3D2>&nbsp; RFC 3323) would facilitate the implementation =
of a local privacy </FONT>
<BR><FONT SIZE=3D2>&nbsp; policy associated with specific Request-URIs. =
</FONT>
</P>

<P><FONT SIZE=3D2>o Issue 2 is around whether there needs to be some =
bounds on the size </FONT>
<BR><FONT SIZE=3D2>&nbsp; or number of History-Info entries.&nbsp; And =
if so, what's the mechanism</FONT>
<BR><FONT SIZE=3D2>&nbsp; for defining the bounds (e.g. limit the depth =
and breadth of the </FONT>
<BR><FONT SIZE=3D2>&nbsp; tree by putting upper limits on the Index? =
)&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>Regards,</FONT>
<BR><FONT SIZE=3D2>Mary H. Barnes</FONT>
<BR><FONT SIZE=3D2>mary.barnes@nortelnetworks.com</FONT>
<BR><FONT SIZE=3D2>972-684-5432</FONT>
</P>
<BR>

</BODY>
</HTML>
------_=_NextPart_001_01C3F3F1.5B962DA4--

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From exim@www1.ietf.org  Sun Feb 15 15:21:43 2004
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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'Eric Burger'" <eburger@snowshore.com>,
        "SIP List (E-mail)"
	 <sip@ietf.org>
Subject: RE: [Sip] Comments on history-info
Date: Sun, 15 Feb 2004 14:18:12 -0600
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Eric,

Although Rohan responded to your "Big question:" part of this email:
http://www1.ietf.org/mail-archive/working-groups/sip/current/msg08998.html
it doesn't appear that I ever responded to the remaining points on the list.
Thank you for reviewing the draft and providing detailed comments.   In
updating the draft, I considered your comments as indicated below [MB].

Regards,
Mary. 

-----Original Message-----
From: Eric Burger [mailto:eburger@snowshore.com]
Sent: Thursday, November 06, 2003 3:44 AM
To: SIP List (E-mail)
Subject: [Sip] Comments on history-info


Big question:
Why don't we use multiple History-Info headers, instead of a (potentially
VERY long) single header?

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).

[MB]: I've clarified this. 

Section 1.2:
How is History-Info any different than Via w.r.t. security?

[MB]: It's quite similar actually, so I've added a statement to that effect
in this section.  I think the only difference being the dependency on the
Index written by the previous hop to generate the index for the current hop.
I do think the general M2M and M2E security solution would also be
applicable to Via if TLS were not deemed to be sufficient (i.e if there were
some reason why you would not want some intermediaries to be able to see the
Via added by an "untrusted" intermediary).  

Nits:
Section 2.5, first paragraph, last sentence:
this would like be a local
this would likely be a local
               ^^
[MB]: Fixed. 

Appendix B, message F8, extra space before last angle-bracket in the
History-Info line.

[MB]: Fixed.

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Date: Mon, 16 Feb 2004 05:06:50 +0800
From: neil quiogue <neil@quiogue.com>
To: Marc Petit-Huguenin <petithug@acm.org>
cc: sip@ietf.org
Subject: Re: [Sip] Binary Encoding for SIP
Message-ID: <140639390.1076908010@[192.168.1.101]>
In-Reply-To: <402FA64B.2060907@acm.org>
References: <BC5459B9.31568%fluffy@cisco.com> <402FA64B.2060907@acm.org>
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I think what Cullen is proposing is to only send one type of MIME body part =

encoding (i.e., binary) instead of having multiple possible MIME encoding=20
as specified on RFC3261/7.1.4.

Regards, Neil


--On dimanche 15 f=E9vrier 2004 09:03 -0800 Marc Petit-Huguenin=20
<petithug@acm.org> wrote:

> I thought that binary body was already supported: RFC3261 section 7.4.1,
> third paragraph.
>
> Cullen Jennings wrote:
>>
>> I submitted a draft on binary encoding for SIP. This improves
>> interoperability, improves speed, decreases sizes of messages, and
>> generally makes life better for SIP which will no doubt eventually help
>> with World Hunger. I need feedback, please let me know:
>>
>> a) Yah, lets do it
>>
>> b) No this is a stupid ideas and here are the reasons why
>>
>> c) There has not been many list comments, this draft involves some very
>> complex issues. I need a year or two to develop an opinion on the single
>> sentence inside it.
>>
>> d) I agree with Cowboy Willis
>>
>> Until it arrives in the archives, you can find it at
>>
>> http://www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.html
>>
>> or, if you prefer ASCII encodings of everything,
>>
>> www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.txt
>>
>> Thanks, Cullen
>>
>>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>





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From: "George Foti (QC/EMC)" <george.foti@ericsson.com>
To: "'neil quiogue'" <neil@quiogue.com>,
        Marc Petit-Huguenin
	 <petithug@acm.org>
Cc: sip@ietf.org
Subject: RE: [Sip] Binary Encoding for SIP
Date: Sun, 15 Feb 2004 16:26:55 -0600
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Is SIGCOMP not good enough.
Rgds/gf

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of neil
quiogue
Sent: Sunday, February 15, 2004 4:07 PM
To: Marc Petit-Huguenin
Cc: sip@ietf.org
Subject: Re: [Sip] Binary Encoding for SIP


I think what Cullen is proposing is to only send one type of MIME body pa=
rt=20
encoding (i.e., binary) instead of having multiple possible MIME encoding=
=20
as specified on RFC3261/7.1.4.

Regards, Neil


--On dimanche 15 f=E9vrier 2004 09:03 -0800 Marc Petit-Huguenin=20
<petithug@acm.org> wrote:

> I thought that binary body was already supported: RFC3261 section 7.4.1=
,
> third paragraph.
>
> Cullen Jennings wrote:
>>
>> I submitted a draft on binary encoding for SIP. This improves
>> interoperability, improves speed, decreases sizes of messages, and
>> generally makes life better for SIP which will no doubt eventually hel=
p
>> with World Hunger. I need feedback, please let me know:
>>
>> a) Yah, lets do it
>>
>> b) No this is a stupid ideas and here are the reasons why
>>
>> c) There has not been many list comments, this draft involves some ver=
y
>> complex issues. I need a year or two to develop an opinion on the sing=
le
>> sentence inside it.
>>
>> d) I agree with Cowboy Willis
>>
>> Until it arrives in the archives, you can find it at
>>
>> http://www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.html
>>
>> or, if you prefer ASCII encodings of everything,
>>
>> www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.txt
>>
>> Thanks, Cullen
>>
>>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>





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Subject: RE: [Sip] Updated History-Info draft submitted
Date: Mon, 16 Feb 2004 09:51:21 +1100
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Will interworking with and maintaining ISUP diversion information when =
making a PSTN to SIP call be part of the scope of any future work for =
the history-info draft?

-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortelnetworks.com]
Sent: Monday, 16 February 2004 5:27 AM
To: 'sip@ietf.org'
Subject: [Sip] Updated History-Info draft submitted



I have submitted an updated version of the History-Info draft.  Until it =
becomes available in the respository, you can pick it up at:

http://home.comcast.net/~mary.barnes/IETF/draft-ietf-sip-history-info-02.=
txt=20

The primary changes are as follows:=20
o Merged the SIPPING WG requirements draft into this document.=20
   Also, removed redirect server from ISSUER-req since the=20
   solution identified this as not being required (or desirable).=20
 =20
o Added an explicit privacy requirement (PRIV-req-3) for the=20
  proxy's role in recognizing and maintaining privacy associated=20
  with a Request-URI being captured in History-Info due to local=20
  policy. (Note, that the text was already there, it just wasn't=20
  highlighted as an explicit requirement). =20

o Removed the compact form for the header since unknown headers=20
  with multiple entries would not be recognized.=20

o Added a summary of Application Considerations to address=20
  concerns about the optional usage of History-Info.=20

There are 2 open issues identified in the draft:=20
o Issue 1 is around the privacy issue that has been discussed=20
  recently on the list:=20
http://www.ietf.org/mail-archive/working-groups/sip/current/msg09401.html=
=20
http://www.ietf.org/mail-archive/working-groups/sip/current/msg09371.html=
=20

  It has been suggested that adding an additional field to the=20
  History-Info header (or extending the priv-values defined in=20
  RFC 3323) would facilitate the implementation of a local privacy=20
  policy associated with specific Request-URIs.=20

o Issue 2 is around whether there needs to be some bounds on the size=20
  or number of History-Info entries.  And if so, what's the mechanism=20
  for defining the bounds (e.g. limit the depth and breadth of the=20
  tree by putting upper limits on the Index? ) =20

Regards,=20
Mary H. Barnes=20
mary.barnes@nortelnetworks.com=20
972-684-5432=20



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

<META content=3D"MSHTML 6.00.2800.1106" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D763354822-15022004>Will&nbsp;interworking with and maintaining =
ISUP=20
diversion information&nbsp;when making a PSTN to SIP call be part of the =
scope=20
of any future work for the history-info draft?</SPAN></FONT></DIV>
<BLOCKQUOTE>
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Mary Barnes=20
  [mailto:mary.barnes@nortelnetworks.com]<BR><B>Sent:</B> Monday, 16 =
February=20
  2004 5:27 AM<BR><B>To:</B> 'sip@ietf.org'<BR><B>Subject:</B> [Sip] =
Updated=20
  History-Info draft submitted<BR><BR></FONT></DIV>
  <P><FONT size=3D2>I have submitted an updated version of the =
History-Info=20
  draft.&nbsp; Until it becomes available in the respository, you can =
pick it up=20
  at:</FONT></P>
  <P><FONT size=3D2><A=20
  =
href=3D"http://home.comcast.net/~mary.barnes/IETF/draft-ietf-sip-history-=
info-02.txt"=20
  =
target=3D_blank>http://home.comcast.net/~mary.barnes/IETF/draft-ietf-sip-=
history-info-02.txt</A></FONT>=20
  </P>
  <P><FONT size=3D2>The primary changes are as follows:</FONT> <BR><FONT =
size=3D2>o=20
  Merged the SIPPING WG requirements draft into this document. =
</FONT><BR><FONT=20
  size=3D2>&nbsp;&nbsp; Also, removed redirect server from ISSUER-req =
since the=20
  </FONT><BR><FONT size=3D2>&nbsp;&nbsp; solution identified this as not =
being=20
  required (or desirable).</FONT> <BR><FONT size=3D2>&nbsp; =
</FONT><BR><FONT=20
  size=3D2>o Added an explicit privacy requirement (PRIV-req-3) for the=20
  </FONT><BR><FONT size=3D2>&nbsp; proxy's role in recognizing and =
maintaining=20
  privacy associated </FONT><BR><FONT size=3D2>&nbsp; with a Request-URI =
being=20
  captured in History-Info due to local </FONT><BR><FONT size=3D2>&nbsp; =
policy.=20
  (Note, that the text was already there, it just wasn't =
</FONT><BR><FONT=20
  size=3D2>&nbsp; highlighted as an explicit requirement).&nbsp; =
</FONT></P>
  <P><FONT size=3D2>o Removed the compact form for the header since =
unknown=20
  headers </FONT><BR><FONT size=3D2>&nbsp; with multiple entries would =
not be=20
  recognized.</FONT> </P>
  <P><FONT size=3D2>o Added a summary of Application Considerations to =
address=20
  </FONT><BR><FONT size=3D2>&nbsp; concerns about the optional usage of=20
  History-Info. </FONT></P>
  <P><FONT size=3D2>There are 2 open issues identified in the =
draft:</FONT>=20
  <BR><FONT size=3D2>o Issue 1 is around the privacy issue that has been =
discussed=20
  </FONT><BR><FONT size=3D2>&nbsp; recently on the list:</FONT> =
<BR><FONT=20
  size=3D2><A=20
  =
href=3D"http://www.ietf.org/mail-archive/working-groups/sip/current/msg09=
401.html"=20
  =
target=3D_blank>http://www.ietf.org/mail-archive/working-groups/sip/curre=
nt/msg09401.html</A></FONT>=20
  <BR><FONT size=3D2><A=20
  =
href=3D"http://www.ietf.org/mail-archive/working-groups/sip/current/msg09=
371.html"=20
  =
target=3D_blank>http://www.ietf.org/mail-archive/working-groups/sip/curre=
nt/msg09371.html</A></FONT>=20
  </P>
  <P><FONT size=3D2>&nbsp; It has been suggested that adding an =
additional field=20
  to the </FONT><BR><FONT size=3D2>&nbsp; History-Info header (or =
extending the=20
  priv-values defined in </FONT><BR><FONT size=3D2>&nbsp; RFC 3323) =
would=20
  facilitate the implementation of a local privacy </FONT><BR><FONT=20
  size=3D2>&nbsp; policy associated with specific Request-URIs. =
</FONT></P>
  <P><FONT size=3D2>o Issue 2 is around whether there needs to be some =
bounds on=20
  the size </FONT><BR><FONT size=3D2>&nbsp; or number of History-Info=20
  entries.&nbsp; And if so, what's the mechanism</FONT> <BR><FONT =
size=3D2>&nbsp;=20
  for defining the bounds (e.g. limit the depth and breadth of the=20
  </FONT><BR><FONT size=3D2>&nbsp; tree by putting upper limits on the =
Index?=20
  )&nbsp; </FONT></P>
  <P><FONT size=3D2>Regards,</FONT> <BR><FONT size=3D2>Mary H. =
Barnes</FONT>=20
  <BR><FONT size=3D2>mary.barnes@nortelnetworks.com</FONT> <BR><FONT=20
  size=3D2>972-684-5432</FONT> </P><BR></BLOCKQUOTE></BODY></HTML>

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



From exim@www1.ietf.org  Sun Feb 15 17:57:37 2004
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Date: Sun, 15 Feb 2004 17:53:19 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: [Sip] draft-ietf-sip-resource-priority-02
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I will shortly (finally...) be submitting 
draft-ietf-sip-resource-priority-02

http://www.cs.columbia.edu/sip/draft/resource-priority/draft-ietf-sip-resource-priority-02.html

Changes are noted below, based on my reading of the January 2004 mailing 
list discussions and the reviews by Paul Kyvizat and Ben Campbell last fall.

(1) Corrected BNF. [Paul, Rohan]

(2) Added namespace registration template and converted initial 
registrations to that template. [multiple]

(3) Now there is a single IANA section with subsections. [Rohan]

(4) Current behavior for DSN, Q.735 specified as non-preemptive. DRSN 
left as implementation-defined. Not sure what other behavior one can 
specify for priorities. [Rohan]

(5) Resource priority for SUBSCRIBE could make sense if a UA has a 
finite capacity for subscriptions and notifications (memory or 
bandwidth). Since NOTIFY establishes no state, only the generic 
request-queueing behavior applies. Text added. [Rohan]

(6) Addressed wording comments (Section 3); was no longer in the draft 
anyway. [Rohan]

(7) Complete request method list. [Rohan] Rather than speculating where 
this might be used, marked as 'o'. [Paul]

(8) Changed 'support' to 'implement' and added some explanatory wording. 
[Rohan]

(9) resource-priority option tag defined in Section 3. [Rohan]

(10) RFC XXXX note added, but not really necessary since the new XML 
version contains &rfc.number; [Rohan]

(11) Removed Appendix A ('Addressing the Requirements') [Ken Karlberg]

(12) Typo and wording corrections [Paul, Ben]

(13) Added distinction to Priority header, but left old text intact. [Paul]

(14) For simplicity, restricted Accept-R-P to UAS; multiple R-P values 
have always been possible. [Paul]

(15) Accept-R-P 420 marked as 'o', with explanation [Paul]. (I'm dubious 
on the reasoning since this means that 'm' cannot appear in any 
extension, thus diminishing the value of the table.)

(16) Multiple namespaces handling made more concrete. [Paul]

(17) Added caveat to OPTIONS handling under overload. [Paul]

(18) OPTIONS handling made MUST. [Ben]

(19) Added discussion on what happens if you modify an R-P value in 
transit. [Ben, others]

Changes discussed, but not made:

- Continues to allow one priority value since the presence of any value 
indicates special treatment, regardless of whether there is another 
value. (In other words, default behavior is a request with no 
Resource-Priority header.)

This may be partially due to confusion in versioning; since the last 
IETF, there were no more 'default' values. (Indeed, the word does not 
appear in the document.) [Mike Pierce]

- 417 generalization to any unknown parameter (since this would not 
allow the user to determine why the request failed) [Alex Audu]

- No mechanism to name all elements in namespace, for simplicity and to 
avoid arguments over which list is referred to by all when a namespace 
gets additional values. [Paul]

Given the stretch of the discussion, it is possible I missed a comment 
or discussion thread. If you respond to this message, please keep each 
item in a separate message as it is very difficult to deal with monster 
messages covering 20 different points.

Henning


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Subject: RE: [Sip] Comments on history-info
Date: Sun, 15 Feb 2004 17:59:59 -0500
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Thread-Topic: [Sip] Comments on history-info
Thread-Index: AcP0APEouqnUCB5tQTSCZ63TeEfehAABicmQ
From: "Eric Burger" <eburger@snowshore.com>
To: "Mary Barnes" <mary.barnes@nortelnetworks.com>,
        "SIP List (E-mail)" <sip@ietf.org>
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Excellent.

Thinking about it a bit more, is there any reason to ALLOW stuffing =
multiple history info info's on a single line?  The idea going back to =
SMTP was that you are only allowed to add history, not change it.  =
Putting multiple history info's in a single history-info header, by =
definition, changes history.

> -----Original Message-----
> From: Mary Barnes [mailto:mary.barnes@nortelnetworks.com]
> Sent: Sunday, February 15, 2004 3:18 PM
> To: Eric Burger; SIP List (E-mail)
> Subject: RE: [Sip] Comments on history-info
>=20
>=20
> Eric,
>=20
> Although Rohan responded to your "Big question:" part of this email:
> http://www1.ietf.org/mail-archive/working-groups/sip/current/m
> sg08998.html
> it doesn't appear that I ever responded to the remaining=20
> points on the list.
> Thank you for reviewing the draft and providing detailed=20
> comments.   In
> updating the draft, I considered your comments as indicated=20
> below [MB].
>=20
> Regards,
> Mary.=20
>=20
> -----Original Message-----
> From: Eric Burger [mailto:eburger@snowshore.com]
> Sent: Thursday, November 06, 2003 3:44 AM
> To: SIP List (E-mail)
> Subject: [Sip] Comments on history-info
>=20
>=20
> Big question:
> Why don't we use multiple History-Info headers, instead of a=20
> (potentially
> VERY long) single header?
>=20
> I would argue for multiple headers on two grounds.
>=20
> 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.
>=20
> Second, the index parameter frees us from worrying about=20
> header order, which
> is a VERY good thing.
>=20
>=20
>=20
> Overview:
> While the motivation for History-Info is for the initiator of=20
> a request to
> know what happened to a request, it is also of great use to=20
> the recipient
> (End of Section 2 overview paragraph).
>=20
> [MB]: I've clarified this.=20
>=20
> Section 1.2:
> How is History-Info any different than Via w.r.t. security?
>=20
> [MB]: It's quite similar actually, so I've added a statement=20
> to that effect
> in this section.  I think the only difference being the=20
> dependency on the
> Index written by the previous hop to generate the index for=20
> the current hop.
> I do think the general M2M and M2E security solution would also be
> applicable to Via if TLS were not deemed to be sufficient=20
> (i.e if there were
> some reason why you would not want some intermediaries to be=20
> able to see the
> Via added by an "untrusted" intermediary). =20
>=20
> Nits:
> Section 2.5, first paragraph, last sentence:
> this would like be a local
> this would likely be a local
>                ^^
> [MB]: Fixed.=20
>=20
> Appendix B, message F8, extra space before last angle-bracket in the
> History-Info line.
>=20
> [MB]: Fixed.
>=20
> _______________________________________________
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>=20
>=20


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From exim@www1.ietf.org  Sun Feb 15 20:44:45 2004
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sip] Updated gruu I-D
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Folks,

I've just submitted an update to the GRUU specification. Until it 
appears in the archives, you can find it at:

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

The differences are:

* removed requirement that the gruu be constructed in such a way that
the contact and/or AOR can be determined from it

* indicate that a gruu can be obtained by the register mechanisms in
the specification, but that other mechanisms are also possible

* The gruu is now associated with the instance ID of the UA. The
instance ID is defined as a callee capability parameter. The
specification now includes a UML diagram showing the relationships
between GRUU, AOR, Contact URI and instance ID.

* Added registration of the sip.instance media feature tag



The biggest change is the third. Indeed, it seems like we have three 
drafts this round that discuss instance identifiers in SIP:

http://www.jdrosen.net/papers/draft-ietf-sip-gruu-01.txt
http://www.ietf.org/internet-drafts/draft-jennings-sipping-instance-id-00.txt
http://www.ietf.org/internet-drafts/draft-stucker-sip-guid-00.txt

all three are similar. Mine uses a contact header field parameter, and 
uses the callee capabilities framework to enable caller preferences 
routing based on instance iDs. Cullen also uses a contact parameter, but 
not within the callee caps framework. Brian defines a header field. I'll 
post a separate not starting a thread on the pros and cons of the 
various approaches.

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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From exim@www1.ietf.org  Sun Feb 15 20:50:48 2004
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Subject: [Sip] SIP instance identifiers
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Folks,

We have three drafts this round that all propose a way to define an 
instance identifier for a UA:


http://www.jdrosen.net/papers/draft-ietf-sip-gruu-01.txt
http://www.ietf.org/internet-drafts/draft-jennings-sipping-instance-id-00.txt
http://www.ietf.org/internet-drafts/draft-stucker-sip-guid-00.txt

all three are similar. Mine uses a contact header field parameter, and 
uses the callee capabilities framework to enable caller preferences 
routing based on instance iDs. Cullen also uses a contact parameter, but 
not within the callee caps framework. Brian defines a header field.

I think we generally understand the requirement here, though it would 
probably be good to obtain use cases to be sure. The design choice is 
where to include this parameter, and this represents the difference 
between the three approaches.

I tend to prefer a Contact URI parameter over a header. My main reason 
for that is REGISTER requests. If a register request has multiple 
contacts, which contact does the instance identifier apply to? As a 
header field, it would be hard to know. If you make it a contact header 
field parameter, its no problem. Those header field parameters could 
also then be used in the Contact in INVITE/200 SUB/NOT, etc.

Then, the question is whether to use a regular Contact header field 
parameter as Cullen has done, or a callee cap parameter as I have done? 
The callee cap approach allows for caller preferences to be applied to 
routing to a specific instance. Now, a good question is whether such a 
thing is needed or even a good idea, if we have GRUU as the ideal way to 
route to a UA instance. I can go either way, but welcome thoughts on the 
topic.

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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Subject: RE: [Sip] draft-khartabil-sip-policy-uri-call-info-purpose-00
Date: Mon, 16 Feb 2004 11:07:13 +0200
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Thanks.

Do you think that it will only be limited to conferences?

/Hisham

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
> Cullen Jennings
> Sent: 13.February.2004 18:49
> To: sip@ietf.org
> Subject: [Sip] draft-khartabil-sip-policy-uri-call-info-purpose-00
>=20
>=20
>    =20
> I like this draft, just one little NIT, I think it would be=20
> better to call
> the tag conf-policy-info instead of just policy-info
>=20
> Cullen
>=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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In a message dated 2/15/2004 5:57:50 PM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


> This may be partially due to confusion in versioning; since the last 
> IETF, there were no more 'default' values. (Indeed, the word does not 
> appear in the document.) [Mike Pierce]
> 

I don't know what the confusion is. Looking at the -01 version contained on 
the IETF-58 CD I just received, there is the following statement in Section 12:

"The registration MUST indicate the default level to be assumed in the 
absence of the priority value ..."

Was there another unumbered update that I missed?

Mike



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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/15/2004 5:57:50 PM Eastern Standard Time, hgs@cs.columbia.edu writes=
:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">This may be partially due t=
o confusion in versioning; since the last=20
<BR>IETF, there were no more 'default' values. (Indeed, the word does not=20
<BR>appear in the document.) [Mike Pierce]
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>I don't know what the confusion is. Looking at the -01 version contained=
 on the IETF-58 CD I just received, there is the following statement in Sect=
ion 12:
<BR>
<BR>"The registration MUST indicate the default level to be assumed in the a=
bsence of the priority value ..."
<BR>
<BR>Was there another unumbered update that I missed?
<BR>
<BR>Mike
<BR>
<BR></FONT></HTML>

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> I don't know what the confusion is. Looking at the -01 version contained 
> on the IETF-58 CD I just received, there is the following statement in 
> Section 12:
> 
> "The registration MUST indicate the default level to be assumed in the 
> absence of the priority value ..."
> 
> Was there another unumbered update that I missed?
> 

There was an early -02 that I only announced to the mailing list, but 
never got around to submitting. In any event, the notion of defaults is 
gone since it makes no sense in the current setup.

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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'Aders, David I'" <David.I.Aders@team.telstra.com>, sip@ietf.org
Cc: "'john.elwell@siemens.com'" <john.elwell@siemens.com>
Subject: RE: [Sip] Updated History-Info draft submitted
Date: Mon, 16 Feb 2004 08:51:53 -0600
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No, this draft will not address ISUP interworking.  The draft provides some
example applications in order to highlight how the History-Info header could
be used, but there's nothing normative about any of those examples.   John
Elwell had put out a draft on how to do diversion for QSIG interworking,
which does make use of History-Info that you might find useful (note the
draft itself has expired and I'll let John address how or whether he plans
to progress that; my understanding is that this work is being done in ECMA).
So, I would think the functionality you're describing might be something the
ITU or another SDO would want to define.  

Regards,
Mary.

-----Original Message-----
From: Aders, David I [mailto:David.I.Aders@team.telstra.com]
Sent: Sunday, February 15, 2004 4:51 PM
To: sip@ietf.org
Subject: RE: [Sip] Updated History-Info draft submitted


Will interworking with and maintaining ISUP diversion information when
making a PSTN to SIP call be part of the scope of any future work for the
history-info draft?
-----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortelnetworks.com]
Sent: Monday, 16 February 2004 5:27 AM
To: 'sip@ietf.org'
Subject: [Sip] Updated History-Info draft submitted


I have submitted an updated version of the History-Info draft.  Until it
becomes available in the respository, you can pick it up at:
http://home.comcast.net/~mary.barnes/IETF/draft-ietf-sip-history-info-02.txt

The primary changes are as follows: 
o Merged the SIPPING WG requirements draft into this document. 
   Also, removed redirect server from ISSUER-req since the 
   solution identified this as not being required (or desirable). 
  
o Added an explicit privacy requirement (PRIV-req-3) for the 
  proxy's role in recognizing and maintaining privacy associated 
  with a Request-URI being captured in History-Info due to local 
  policy. (Note, that the text was already there, it just wasn't 
  highlighted as an explicit requirement).  
o Removed the compact form for the header since unknown headers 
  with multiple entries would not be recognized. 
o Added a summary of Application Considerations to address 
  concerns about the optional usage of History-Info. 
There are 2 open issues identified in the draft: 
o Issue 1 is around the privacy issue that has been discussed 
  recently on the list: 
http://www.ietf.org/mail-archive/working-groups/sip/current/msg09401.html 
http://www.ietf.org/mail-archive/working-groups/sip/current/msg09371.html 
  It has been suggested that adding an additional field to the 
  History-Info header (or extending the priv-values defined in 
  RFC 3323) would facilitate the implementation of a local privacy 
  policy associated with specific Request-URIs. 
o Issue 2 is around whether there needs to be some bounds on the size 
  or number of History-Info entries.  And if so, what's the mechanism 
  for defining the bounds (e.g. limit the depth and breadth of the 
  tree by putting upper limits on the Index? )  
Regards, 
Mary H. Barnes 
mary.barnes@nortelnetworks.com 
972-684-5432 

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>RE: [Sip] Updated History-Info draft submitted</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>No, this draft will not address ISUP =
interworking.&nbsp; The draft provides some example applications in =
order to highlight how the History-Info header could be used, but =
there's nothing normative about any of those examples.&nbsp;&nbsp; John =
Elwell had put out a draft on how to do diversion for QSIG =
interworking, which does make use of History-Info that you might find =
useful (note the draft itself has expired and I'll let John address how =
or whether he plans to progress that; my understanding is that this =
work is being done in ECMA).&nbsp; So, I would think the functionality =
you're describing might be something the ITU or another SDO would want =
to define.&nbsp; </FONT></P>

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

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Aders, David I [<A =
HREF=3D"mailto:David.I.Aders@team.telstra.com">mailto:David.I.Aders@team=
.telstra.com</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Sunday, February 15, 2004 4:51 PM</FONT>
<BR><FONT SIZE=3D2>To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] Updated History-Info draft =
submitted</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Will interworking with and maintaining ISUP diversion =
information when making a PSTN to SIP call be part of the scope of any =
future work for the history-info draft?</FONT></P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Mary Barnes [<A =
HREF=3D"mailto:mary.barnes@nortelnetworks.com">mailto:mary.barnes@nortel=
networks.com</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Monday, 16 February 2004 5:27 AM</FONT>
<BR><FONT SIZE=3D2>To: 'sip@ietf.org'</FONT>
<BR><FONT SIZE=3D2>Subject: [Sip] Updated History-Info draft =
submitted</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>I have submitted an updated version of the =
History-Info draft.&nbsp; Until it becomes available in the =
respository, you can pick it up at:</FONT></P>

<P><FONT SIZE=3D2><A =
HREF=3D"http://home.comcast.net/~mary.barnes/IETF/draft-ietf-sip-history=
-info-02.txt" =
TARGET=3D"_blank">http://home.comcast.net/~mary.barnes/IETF/draft-ietf-s=
ip-history-info-02.txt</A> </FONT>
<BR><FONT SIZE=3D2>The primary changes are as follows: </FONT>
<BR><FONT SIZE=3D2>o Merged the SIPPING WG requirements draft into this =
document. </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; Also, removed redirect server from =
ISSUER-req since the </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; solution identified this as not being =
required (or desirable). </FONT>
<BR><FONT SIZE=3D2>&nbsp; </FONT>
<BR><FONT SIZE=3D2>o Added an explicit privacy requirement (PRIV-req-3) =
for the </FONT>
<BR><FONT SIZE=3D2>&nbsp; proxy's role in recognizing and maintaining =
privacy associated </FONT>
<BR><FONT SIZE=3D2>&nbsp; with a Request-URI being captured in =
History-Info due to local </FONT>
<BR><FONT SIZE=3D2>&nbsp; policy. (Note, that the text was already =
there, it just wasn't </FONT>
<BR><FONT SIZE=3D2>&nbsp; highlighted as an explicit =
requirement).&nbsp; </FONT>
<BR><FONT SIZE=3D2>o Removed the compact form for the header since =
unknown headers </FONT>
<BR><FONT SIZE=3D2>&nbsp; with multiple entries would not be =
recognized. </FONT>
<BR><FONT SIZE=3D2>o Added a summary of Application Considerations to =
address </FONT>
<BR><FONT SIZE=3D2>&nbsp; concerns about the optional usage of =
History-Info. </FONT>
<BR><FONT SIZE=3D2>There are 2 open issues identified in the draft: =
</FONT>
<BR><FONT SIZE=3D2>o Issue 1 is around the privacy issue that has been =
discussed </FONT>
<BR><FONT SIZE=3D2>&nbsp; recently on the list: </FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/mail-archive/working-groups/sip/current/msg0=
9401.html" =
TARGET=3D"_blank">http://www.ietf.org/mail-archive/working-groups/sip/cu=
rrent/msg09401.html</A> </FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/mail-archive/working-groups/sip/current/msg0=
9371.html" =
TARGET=3D"_blank">http://www.ietf.org/mail-archive/working-groups/sip/cu=
rrent/msg09371.html</A> </FONT>
<BR><FONT SIZE=3D2>&nbsp; It has been suggested that adding an =
additional field to the </FONT>
<BR><FONT SIZE=3D2>&nbsp; History-Info header (or extending the =
priv-values defined in </FONT>
<BR><FONT SIZE=3D2>&nbsp; RFC 3323) would facilitate the implementation =
of a local privacy </FONT>
<BR><FONT SIZE=3D2>&nbsp; policy associated with specific Request-URIs. =
</FONT>
<BR><FONT SIZE=3D2>o Issue 2 is around whether there needs to be some =
bounds on the size </FONT>
<BR><FONT SIZE=3D2>&nbsp; or number of History-Info entries.&nbsp; And =
if so, what's the mechanism </FONT>
<BR><FONT SIZE=3D2>&nbsp; for defining the bounds (e.g. limit the depth =
and breadth of the </FONT>
<BR><FONT SIZE=3D2>&nbsp; tree by putting upper limits on the Index? =
)&nbsp; </FONT>
<BR><FONT SIZE=3D2>Regards, </FONT>
<BR><FONT SIZE=3D2>Mary H. Barnes </FONT>
<BR><FONT SIZE=3D2>mary.barnes@nortelnetworks.com </FONT>
<BR><FONT SIZE=3D2>972-684-5432 </FONT>
</P>

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Use sip-implementors@cs.columbia.edu for questions on current sip
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	 <sip@ietf.org>
Subject: RE: [Sip] Comments on history-info
Date: Mon, 16 Feb 2004 08:57:50 -0600
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Eric,

History-Info is a  bit different than the SMTP (at least as best as I
understand SMTP not being an expert in that area), since the retargeting in
SIP is more complicated that SMTP due to forking, etc.  In order to be able
to accurately reflecting the forking, we've added the index.  The index is
used to generate subsequent History-Info entries.  In addition, there's the
Reason that gets added to the targeted-to URIs to reflect the reason for the
retargeting. So, yes History-Info does change History in the sense that we
embellish the Request-URIs that are "visited" with additional information
that seems to be useful to a variety of end applications.  So, I think what
you're proposing would only provide a subset of the functionality that we're
after.

Mary. 

-----Original Message-----
From: Eric Burger [mailto:eburger@snowshore.com]
Sent: Sunday, February 15, 2004 5:00 PM
To: Barnes, Mary [NGC:B622:EXCH]; SIP List (E-mail)
Subject: RE: [Sip] Comments on history-info


Excellent.

Thinking about it a bit more, is there any reason to ALLOW stuffing multiple
history info info's on a single line?  The idea going back to SMTP was that
you are only allowed to add history, not change it.  Putting multiple
history info's in a single history-info header, by definition, changes
history.

> -----Original Message-----
> From: Mary Barnes [mailto:mary.barnes@nortelnetworks.com]
> Sent: Sunday, February 15, 2004 3:18 PM
> To: Eric Burger; SIP List (E-mail)
> Subject: RE: [Sip] Comments on history-info
> 
> 
> Eric,
> 
> Although Rohan responded to your "Big question:" part of this email:
> http://www1.ietf.org/mail-archive/working-groups/sip/current/m
> sg08998.html
> it doesn't appear that I ever responded to the remaining 
> points on the list.
> Thank you for reviewing the draft and providing detailed 
> comments.   In
> updating the draft, I considered your comments as indicated 
> below [MB].
> 
> Regards,
> Mary. 
> 
> -----Original Message-----
> From: Eric Burger [mailto:eburger@snowshore.com]
> Sent: Thursday, November 06, 2003 3:44 AM
> To: SIP List (E-mail)
> Subject: [Sip] Comments on history-info
> 
> 
> Big question:
> Why don't we use multiple History-Info headers, instead of a 
> (potentially
> VERY long) single header?
> 
> 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).
> 
> [MB]: I've clarified this. 
> 
> Section 1.2:
> How is History-Info any different than Via w.r.t. security?
> 
> [MB]: It's quite similar actually, so I've added a statement 
> to that effect
> in this section.  I think the only difference being the 
> dependency on the
> Index written by the previous hop to generate the index for 
> the current hop.
> I do think the general M2M and M2E security solution would also be
> applicable to Via if TLS were not deemed to be sufficient 
> (i.e if there were
> some reason why you would not want some intermediaries to be 
> able to see the
> Via added by an "untrusted" intermediary).  
> 
> Nits:
> Section 2.5, first paragraph, last sentence:
> this would like be a local
> this would likely be a local
>                ^^
> [MB]: Fixed. 
> 
> Appendix B, message F8, extra space before last angle-bracket in the
> History-Info line.
> 
> [MB]: Fixed.
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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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Last night, I announced 
http://www.cs.columbia.edu/sip/draft/resource-priority/draft-ietf-sip-resource-priority-02.html

Please address your comments to that draft, to avoid confusion and 
fixing old bugs.

schatterjee@hss.hns.com wrote:

> 
> Hi,
> 
> In draft-ietf-sip-resource-priority-01.txt, it looks like there is a 
> printing mistake in the grammar defined for Resource-Priority header . 
> Its given as
>        Resource-Priority  =  "Resource-Priority" HCOLON
>                               (*COMMA Resource-value)
>  
> But I guess, no header can start with a COMMA . It should actually be
>        Resource-Priority  =  "Resource-Priority" HCOLON
>                               Resource-value(*COMMA Resource-value)
> 
> This is the way the grammar is given for Accept-Resource-Priority.
> Can anyone confirm on this pls?
> 
> thanks in advance...

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From: "Brian Stucker" <bstucker@nortelnetworks.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] SIP instance identifiers
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Johnathan,

I'm not opposed to the client ID being transported in the registration
contact,
but seeing as there were two drafts that already discuss that, I didn't see
much point in rehashing it again in my draft. =)

I believe there are some important points that you're leaving out. 

Besides being client generated (which reduces the requirements for
generating
the ID) there are advantages to NOT mandating that the ID be expressed only 
as a parameter in the contact of a registration. 

The idea behind the GUID (and hence the lack of a specific usecase in the
-00 version of the document) is that it can be used for other activities. 
I'm not opposed to communicating the ID in a contact parameter as well as
a regular header, I just don't view transporting it in a contact header
as being always necessary or desirable depending on what you want to do
with it.

Sending it as a header allows for some interesting behavior to occur on 
non-registration requests:

- Can be used to eliminate duplicate subscriptions due to client's state
being wiped out (ie. ignore the callid/tags of an existing subscription
if the client ID matches the client ID of that subscription [and the other
information about the subscription is the same]).
- Can be used to identify the client sending a request across networks
where no other identifier is easily obtainable (NATs and the such), or 
BBUAs are present in the network.
- Can be used to identify the source of state information being pushed
into the network (presence update, etc). This is one area where conveying
the information in a contact seems a little cumbersome. The request may 
not be sent to the registrar, and a contact may otherwise be completely 
unnecessary.

In the cases where registration behavior is controlled, it requires a change
to the registrar in order to capture the appropriate behavior that should be
taken wrt a client ID. Therefore, including the information as a header or
in the contact seems moot to me as either mechanism can easily be made to 
work. Perhaps there's a usecase I'm not considering?

Additionally, I believe that there needs to be some convergence on the
approach
taken by generating the identifier itself. The mechanism that is suggested 
in your draft seems to rely upon a server to generate the unique identifier,

whereas in the other two drafts, it is the client that generates the 
unique identifier. I believe there are advantages to each. 

Regards,

Brian

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Sunday, February 15, 2004 7:48 PM
To: sip@ietf.org
Subject: [Sip] SIP instance identifiers


Folks,

We have three drafts this round that all propose a way to define an 
instance identifier for a UA:


http://www.jdrosen.net/papers/draft-ietf-sip-gruu-01.txt
http://www.ietf.org/internet-drafts/draft-jennings-sipping-instance-id-00.tx
t
http://www.ietf.org/internet-drafts/draft-stucker-sip-guid-00.txt

all three are similar. Mine uses a contact header field parameter, and 
uses the callee capabilities framework to enable caller preferences 
routing based on instance iDs. Cullen also uses a contact parameter, but 
not within the callee caps framework. Brian defines a header field.

I think we generally understand the requirement here, though it would 
probably be good to obtain use cases to be sure. The design choice is 
where to include this parameter, and this represents the difference 
between the three approaches.

I tend to prefer a Contact URI parameter over a header. My main reason 
for that is REGISTER requests. If a register request has multiple 
contacts, which contact does the instance identifier apply to? As a 
header field, it would be hard to know. If you make it a contact header 
field parameter, its no problem. Those header field parameters could 
also then be used in the Contact in INVITE/200 SUB/NOT, etc.

Then, the question is whether to use a regular Contact header field 
parameter as Cullen has done, or a callee cap parameter as I have done? 
The callee cap approach allows for caller preferences to be applied to 
routing to a specific instance. Now, a good question is whether such a 
thing is needed or even a good idea, if we have GRUU as the ideal way to 
route to a UA instance. I can go either way, but welcome thoughts on the 
topic.

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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<P><FONT SIZE=2>Johnathan,</FONT>
</P>

<P><FONT SIZE=2>I'm not opposed to the client ID being transported in the registration contact,</FONT>
<BR><FONT SIZE=2>but seeing as there were two drafts that already discuss that, I didn't see</FONT>
<BR><FONT SIZE=2>much point in rehashing it again in my draft. =)</FONT>
</P>

<P><FONT SIZE=2>I believe there are some important points that you're leaving out. </FONT>
</P>

<P><FONT SIZE=2>Besides being client generated (which reduces the requirements for generating</FONT>
<BR><FONT SIZE=2>the ID) there are advantages to NOT mandating that the ID be expressed only </FONT>
<BR><FONT SIZE=2>as a parameter in the contact of a registration. </FONT>
</P>

<P><FONT SIZE=2>The idea behind the GUID (and hence the lack of a specific usecase in the</FONT>
<BR><FONT SIZE=2>-00 version of the document) is that it can be used for other activities. </FONT>
<BR><FONT SIZE=2>I'm not opposed to communicating the ID in a contact parameter as well as</FONT>
<BR><FONT SIZE=2>a regular header, I just don't view transporting it in a contact header</FONT>
<BR><FONT SIZE=2>as being always necessary or desirable depending on what you want to do</FONT>
<BR><FONT SIZE=2>with it.</FONT>
</P>

<P><FONT SIZE=2>Sending it as a header allows for some interesting behavior to occur on </FONT>
<BR><FONT SIZE=2>non-registration requests:</FONT>
</P>

<P><FONT SIZE=2>- Can be used to eliminate duplicate subscriptions due to client's state</FONT>
<BR><FONT SIZE=2>being wiped out (ie. ignore the callid/tags of an existing subscription</FONT>
<BR><FONT SIZE=2>if the client ID matches the client ID of that subscription [and the other</FONT>
<BR><FONT SIZE=2>information about the subscription is the same]).</FONT>
<BR><FONT SIZE=2>- Can be used to identify the client sending a request across networks</FONT>
<BR><FONT SIZE=2>where no other identifier is easily obtainable (NATs and the such), or </FONT>
<BR><FONT SIZE=2>BBUAs are present in the network.</FONT>
<BR><FONT SIZE=2>- Can be used to identify the source of state information being pushed</FONT>
<BR><FONT SIZE=2>into the network (presence update, etc). This is one area where conveying</FONT>
<BR><FONT SIZE=2>the information in a contact seems a little cumbersome. The request may </FONT>
<BR><FONT SIZE=2>not be sent to the registrar, and a contact may otherwise be completely </FONT>
<BR><FONT SIZE=2>unnecessary.</FONT>
</P>

<P><FONT SIZE=2>In the cases where registration behavior is controlled, it requires a change</FONT>
<BR><FONT SIZE=2>to the registrar in order to capture the appropriate behavior that should be</FONT>
<BR><FONT SIZE=2>taken wrt a client ID. Therefore, including the information as a header or</FONT>
<BR><FONT SIZE=2>in the contact seems moot to me as either mechanism can easily be made to </FONT>
<BR><FONT SIZE=2>work. Perhaps there's a usecase I'm not considering?</FONT>
</P>

<P><FONT SIZE=2>Additionally, I believe that there needs to be some convergence on the approach</FONT>
<BR><FONT SIZE=2>taken by generating the identifier itself. The mechanism that is suggested </FONT>
<BR><FONT SIZE=2>in your draft seems to rely upon a server to generate the unique identifier, </FONT>
<BR><FONT SIZE=2>whereas in the other two drafts, it is the client that generates the </FONT>
<BR><FONT SIZE=2>unique identifier. I believe there are advantages to each. </FONT>
</P>

<P><FONT SIZE=2>Regards,</FONT>
</P>

<P><FONT SIZE=2>Brian</FONT>
</P>

<P><FONT SIZE=2>-----Original Message-----</FONT>
<BR><FONT SIZE=2>From: Jonathan Rosenberg [<A HREF="mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</A>]</FONT>
<BR><FONT SIZE=2>Sent: Sunday, February 15, 2004 7:48 PM</FONT>
<BR><FONT SIZE=2>To: sip@ietf.org</FONT>
<BR><FONT SIZE=2>Subject: [Sip] SIP instance identifiers</FONT>
</P>
<BR>

<P><FONT SIZE=2>Folks,</FONT>
</P>

<P><FONT SIZE=2>We have three drafts this round that all propose a way to define an </FONT>
<BR><FONT SIZE=2>instance identifier for a UA:</FONT>
</P>
<BR>

<P><FONT SIZE=2><A HREF="http://www.jdrosen.net/papers/draft-ietf-sip-gruu-01.txt" TARGET="_blank">http://www.jdrosen.net/papers/draft-ietf-sip-gruu-01.txt</A></FONT>
<BR><FONT SIZE=2><A HREF="http://www.ietf.org/internet-drafts/draft-jennings-sipping-instance-id-00.txt" TARGET="_blank">http://www.ietf.org/internet-drafts/draft-jennings-sipping-instance-id-00.txt</A></FONT>
<BR><FONT SIZE=2><A HREF="http://www.ietf.org/internet-drafts/draft-stucker-sip-guid-00.txt" TARGET="_blank">http://www.ietf.org/internet-drafts/draft-stucker-sip-guid-00.txt</A></FONT>
</P>

<P><FONT SIZE=2>all three are similar. Mine uses a contact header field parameter, and </FONT>
<BR><FONT SIZE=2>uses the callee capabilities framework to enable caller preferences </FONT>
<BR><FONT SIZE=2>routing based on instance iDs. Cullen also uses a contact parameter, but </FONT>
<BR><FONT SIZE=2>not within the callee caps framework. Brian defines a header field.</FONT>
</P>

<P><FONT SIZE=2>I think we generally understand the requirement here, though it would </FONT>
<BR><FONT SIZE=2>probably be good to obtain use cases to be sure. The design choice is </FONT>
<BR><FONT SIZE=2>where to include this parameter, and this represents the difference </FONT>
<BR><FONT SIZE=2>between the three approaches.</FONT>
</P>

<P><FONT SIZE=2>I tend to prefer a Contact URI parameter over a header. My main reason </FONT>
<BR><FONT SIZE=2>for that is REGISTER requests. If a register request has multiple </FONT>
<BR><FONT SIZE=2>contacts, which contact does the instance identifier apply to? As a </FONT>
<BR><FONT SIZE=2>header field, it would be hard to know. If you make it a contact header </FONT>
<BR><FONT SIZE=2>field parameter, its no problem. Those header field parameters could </FONT>
<BR><FONT SIZE=2>also then be used in the Contact in INVITE/200 SUB/NOT, etc.</FONT>
</P>

<P><FONT SIZE=2>Then, the question is whether to use a regular Contact header field </FONT>
<BR><FONT SIZE=2>parameter as Cullen has done, or a callee cap parameter as I have done? </FONT>
<BR><FONT SIZE=2>The callee cap approach allows for caller preferences to be applied to </FONT>
<BR><FONT SIZE=2>routing to a specific instance. Now, a good question is whether such a </FONT>
<BR><FONT SIZE=2>thing is needed or even a good idea, if we have GRUU as the ideal way to </FONT>
<BR><FONT SIZE=2>route to a UA instance. I can go either way, but welcome thoughts on the </FONT>
<BR><FONT SIZE=2>topic.</FONT>
</P>

<P><FONT SIZE=2>Thanks,</FONT>
<BR><FONT SIZE=2>Jonathan R.</FONT>
<BR><FONT SIZE=2>-- </FONT>
<BR><FONT SIZE=2>Jonathan D. Rosenberg, Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 600 Lanidex Plaza</FONT>
<BR><FONT SIZE=2>Chief Technology Officer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Parsippany, NJ 07054-2711</FONT>
<BR><FONT SIZE=2>dynamicsoft</FONT>
<BR><FONT SIZE=2>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FAX:&nbsp;&nbsp; (973) 952-5050</FONT>
<BR><FONT SIZE=2><A HREF="http://www.jdrosen.net" TARGET="_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) 952-5000</FONT>
<BR><FONT SIZE=2><A HREF="http://www.dynamicsoft.com" TARGET="_blank">http://www.dynamicsoft.com</A></FONT>
</P>
<BR>

<P><FONT SIZE=2>_______________________________________________</FONT>
<BR><FONT SIZE=2>Sip mailing list&nbsp; <A HREF="https://www1.ietf.org/mailman/listinfo/sip" TARGET="_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=2>This list is for NEW development of the core SIP Protocol</FONT>
<BR><FONT SIZE=2>Use sip-implementors@cs.columbia.edu for questions on current sip</FONT>
<BR><FONT SIZE=2>Use sipping@ietf.org for new developments on the application of sip</FONT>
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From exim@www1.ietf.org  Mon Feb 16 14:38:50 2004
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In a message dated 2/15/2004 5:57:50 PM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


> (4) Current behavior for DSN, Q.735 specified as non-preemptive. DRSN 
> left as implementation-defined. Not sure what other behavior one can 
> specify for priorities. [Rohan]
> 

It is unclear what this note means, since Section 9.5 does not reflect what 
it says. The -02 draft (Section 9.5) shows "Preemption with rejection" for dsn 
and drsn and "Precedence" for q735. All three of these cases are services 
which are titled "Precedence and Preemption". In fact, q735 (and I.255.3 on which 
it is based) is the same as dsn/drsn, so there should be no difference.

It continues to be unclear what this item ("Preemption or precedence") on the 
template is intended to mean. There is no IANA function related to this type 
of information, and the contents of this part of the template have no effect 
on the operation of any device which uses the R-P header. They will do whatever 
they need to with it depending on the application. For the dsn application, 
this includes "preemption", "precedence", and probably other treatments that 
aren't included in either of those terms.  In fact, this item seems to be 
defined differently in Section 9.4 which says "there are many possible behaviors 
that cannot be exhaustively anticipated" and lists 3 "common" ones.

I would still like to see this item deleted from the template, since it can 
only lead to confusion.

If it not deleted, the item on the template should be titled "Behavior" (as 
in 9.4) and all three namespaces included in the document should indicate 
"Implementation-defined".

Mike

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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/15/2004 5:57:50 PM Eastern Standard Time, hgs@cs.columbia.edu writes=
:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">(4) Current behavior for DS=
N, Q.735 specified as non-preemptive. DRSN=20
<BR>left as implementation-defined. Not sure what other behavior one can=20
<BR>specify for priorities. [Rohan]
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>It is unclear what this note means, since Section 9.5 does not reflect w=
hat it says. The -02 draft (Section 9.5) shows "Preemption with rejection" f=
or dsn and drsn and "Precedence" for q735. All three of these cases are serv=
ices which are titled "Precedence and Preemption". In fact, q735 (and I.255.=
3 on which it is based) is the same as dsn/drsn, so there should be no diffe=
rence.
<BR>
<BR>It continues to be unclear what this item ("Preemption or precedence") o=
n the template is intended to mean. There is no IANA function related to thi=
s type of information, and the contents of this part of the template have no=
 effect on the operation of any device which uses the R-P header. They will=20=
do whatever they need to with it depending on the application. For the dsn a=
pplication, this includes "preemption", "precedence", and probably other tre=
atments that aren't included in either of those terms. &nbsp;In fact, this i=
tem seems to be defined differently in Section 9.4 which says "there are man=
y possible behaviors that cannot be exhaustively anticipated" and lists 3 "c=
ommon" ones.
<BR>
<BR>I would still like to see this item deleted from the template, since it=20=
can only lead to confusion.
<BR>
<BR>If it not deleted, the item on the template should be titled "Behavior"=20=
(as in 9.4) and all three namespaces included in the document should indicat=
e "Implementation-defined".
<BR>
<BR>Mike</FONT></HTML>

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From exim@www1.ietf.org  Mon Feb 16 14:49:38 2004
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The change note pre-dated my last-minute updates; pardon the rush.

My general notion is that IANA registrations should be as useful as 
possible to the implementor, not just bare-bones information needed to 
preserve uniqueness.

Changing to 'Behavior' is fine by me, with implementation-defined 
clearly being a viable option, as there are likely to be priority 
namespaces that simply don't spell out exactly what you're supposed to 
do, as long as priority X gets better treatment than Y, if prio(X) > 
prio(Y).

Mpierce1@aol.com wrote:

> In a message dated 2/15/2004 5:57:50 PM Eastern Standard Time, 
> hgs@cs.columbia.edu writes:
> 
> 
>> (4) Current behavior for DSN, Q.735 specified as non-preemptive. DRSN
>> left as implementation-defined. Not sure what other behavior one can
>> specify for priorities. [Rohan]
> 
> 
> 
> It is unclear what this note means, since Section 9.5 does not reflect 
> what it says. The -02 draft (Section 9.5) shows "Preemption with 
> rejection" for dsn and drsn and "Precedence" for q735. All three of 
> these cases are services which are titled "Precedence and Preemption". 
> In fact, q735 (and I.255.3 on which it is based) is the same as 
> dsn/drsn, so there should be no difference.
> 
> It continues to be unclear what this item ("Preemption or precedence") 
> on the template is intended to mean. There is no IANA function related 
> to this type of information, and the contents of this part of the 
> template have no effect on the operation of any device which uses the 
> R-P header. They will do whatever they need to with it depending on the 
> application. For the dsn application, this includes "preemption", 
> "precedence", and probably other treatments that aren't included in 
> either of those terms.  In fact, this item seems to be defined 
> differently in Section 9.4 which says "there are many possible behaviors 
> that cannot be exhaustively anticipated" and lists 3 "common" ones.
> 
> I would still like to see this item deleted from the template, since it 
> can only lead to confusion.
> 
> If it not deleted, the item on the template should be titled "Behavior" 
> (as in 9.4) and all three namespaces included in the document should 
> indicate "Implementation-defined".
> 
> Mike

_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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From exim@www1.ietf.org  Mon Feb 16 15:12:42 2004
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Subject: Re: [Sip] FW: I-D ACTION:draft-khartabil-sip-auth-analysis-00.txt
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On Fri, 2004-02-06 at 02:48, hisham.khartabil@nokia.com wrote:
> Having tried to implement HTTP Authentication for SIP, we faced 
> a few problems. This draft tries to highlight those problems and 
> in some cases suggests a solution.

Comments on draft-khartabil-sip-auth-analysis-00

> 2. Use of 401 and 407 Responses, www-authenticate and
>  proxy-authenticate headers
>  [...]
>  The use of 401 and 407 must be more normative. Text should read: A
>  UAS that require authentication MUST use 401 in a response and MUST
>  challenge with a www-authenticate header. Registrars and redirect
>  servers MUST use 401 and www-authenticate header. Proxies MUST use
>  407 and proxy-authenticate header.

That would be a clear improvement, I think.

> 3. Rendering to user

It's not clear to me what it is you're objecting to in this section,
except perhaps that the user interface guideline is poorly rendered.

The real requirement, I think, is that if the user is being asked for
credentials, they should be shown the realm value at that time; a fact
that is made clear in RFC 2617 (sec 3.2.1):

  realm
    A string to be displayed to users so they know which username and
    password to use.

> 4. Rejected or not
>
>  Section 22.1 of RFC 3261 says: "A UAC MUST NOT re-attempt requests
>  with the credentials that have just been rejected (though the request
>  may be retried if the nonce was stale)".
>
>  The problem with this is that how do you know if the request has been
>  rejected or it is just the next hop proxy challenging with the same
>  realm?

This case couldn't come up in HTTP, since proxy authentication in HTTP
is hop-by-hop, and 407 challenges are never passed back to the user
agent by a proxy.  However, the reset of the description of realm is
good advice:

                ... This string should contain at least the name of
   the host performing the authentication and might additionally
   indicate the collection of users who might have access. An example
   might be "registered_users@gotham.news.com".

> 5. Use of To-header vs. Remote URI (Request-URI)
>
>  Section 22.2 of RFC 3261 says: "The UAC may require input from the
>  originating user before proceeding.  Once authentication credentials
>  have been supplied (either directly by the user, or discovered in an
>  internal key-ring), UAs SHOULD cache the credentials for a given
>  value of the To header field and "realm" and attempt to re-use these
>  values on the next request for that destination".
>
>  It is more appropriate to cache request-URI, and not rely on what is
>  in the To-header.

Agreed - the To header value is not protected by the hash, so
associating it is not appropriate.  In addition, this paragraph
encourages the existing confusion between these two values
(admittedly, they are often the same, but it is the Request URI that
is used in authentication).

> 7. Caching outbound proxy credentials
>
>  Section 22.3 of RFC 3261 says: "if a UA is configured with the realm
>  of its local outbound proxy, when one exists, then the UA MAY cache
>  credentials for that realm across dialogs".
>
>  It cannot be assumed that the domain name of a configured outbound
>  proxy is its realm. Therefore a terminal configured with only the
>  outbound proxy URI cannot and MUST NOT cache credentials or any proxy
>  challenge across dialogs.
>
>  This introduces the problem of multiple proxies in a chain
>  challenging with the same realm. How does a UAC know that the
>  challenge was from the outbound proxy and not from a downstream proxy
>  with the same realm as the outbound proxy? Can we mandate that the
>  outbound proxy must have a unique realm?

As a practical matter, if you have the same username and password in
both domains it doesn't matter - if you don't, then it's an impossibly
clumsy situation (this problem is part of the reason that we defined
proxy authentication to by hop-by-hop in HTTP in the first place).

> 9. The realm Issue
>
>  Section 22.3 of RFC 3261 says: "When multiple proxies are used in a
>  chain, a Proxy-Authorization header field value MUST NOT be consumed
>  by any proxy whose realm does not match the "realm" parameter
>  specified in that value".
>
>  Another question: Is it possible that multiple proxies with the same
>  realm are placed in a chain? I.e. the first proxy challenges, user
>  provide correct authentication. That proxy authenticates and forwards
>  the request to a down stream proxy. That down stream proxy has the
>  same realm as first proxy. There are a few issues here:

>  examining the nonce is the only solution.

Essentially, I think that putting two proxies in a row with the same
realm is a difficult configuration to diagnose.  If administrators do
this, they are asking for trouble.  But there is another approach I
don't think you examined.

>  Another issue is at the UAC side. If 2 proxies are in a chain and
>  share a realm, how does the UAC know, when the second proxy
>  challenges, that it is not the first proxy re-challenging because the
>  credentials provided to it were wrong?

If the proxy is using the 2617 version of Digest (with the qop
attribute) as opposed to the 1867 version, then the UA can tell
the difference.  The message flow looks like this:

        UA                    P1                  P2
    a    +------REQ 1--------->|                   |
    b    |<----407(1)----------+                   |
    c    +------REQ 2 -------->+----REQ 2--------->|
    e    |<---407(2)-----------+<-----407(2)-------|
         +------REQ 3--------->+----REQ 3--------->|

 a) initial request
 b) challenge from P1, with the qop parameter specified
 c) retried request, with a qop parameter, and therefor also a cnonce
    chosen by the UA; it is passed on (now without authentication) to P2
 d) P2 challenges, also specifying a qop parameter
    P2 challenge is forwarded by P1, with the WWW-Authenticate
    header as provided by P2, but also with an Authentication-Info
    provided by P1.

The UA can recognize that authentication to P1 was successful, because
P1 returned an Authentication-Info header that contained the cnonce
and nc values from message c (REQ 2), and a correct response hash to
show that P1 knows the authentication secret.

> 10. Authentication-Info
>
>  RFC 3261 allows the usage of Authentication-Info header. The BNF in
>  RFC 3261 allows multiple authentication-info headers where RFC 2617
>  allows only one. Is it only the terminating UAS that is allowed to
>  insert this header. If so, why allow multiples to be present. If not,
>  how does the UAC know which proxy added this header? It cannot know
>  since there is no parameter indicating so, not even nonce.

As I noted earlier, 2617 allows only one because proxy authentication
is hop-by-hop, but the cnonce value allows the UA to disambiguate them
(if they are carefully constructed, but then they should be anyway).

All this is even more reason to use qop (which was the intent of 2617
in the first place).

-- 
Scott Lawrence        
  Pingtel Corp.   



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From exim@www1.ietf.org  Mon Feb 16 15:24:40 2004
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In a message dated 2/16/2004 9:41:28 AM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


> There was an early -02 that I only announced to the mailing list, but 
> never got around to submitting. In any event, the notion of defaults is 
> gone since it makes no sense in the current setup.
> 

Okay, guess I missed that change. Deleting registration of the default makes 
sense anyway. It would be part of the procedural details for a specific use of 
the R-P header instead.

Mike


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/16/2004 9:41:28 AM Eastern Standard Time, hgs@cs.columbia.edu writes=
:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">There was an early -02 that=
 I only announced to the mailing list, but=20
<BR>never got around to submitting. In any event, the notion of defaults is=20
<BR>gone since it makes no sense in the current setup.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>Okay, guess I missed that change. Deleting registration of the default m=
akes sense anyway. It would be part of the procedural details for a specific=
 use of the R-P header instead.
<BR>
<BR>Mike
<BR></FONT></HTML>

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

	Title		: Session Initiation Protocol (SIP) Extension for Event State
                              Publication
	Author(s)	: A. Niemi
	Filename	: draft-ietf-sip-publish-03.txt
	Pages		: 40
	Date		: 2004-2-16
	
This document describes an extension to the Session Initiation
Protocol (SIP) for publishing event state used within the framework
for SIP Event Notification. The first application of this extension
is targeted at the publication of presence information.
The mechanism described in this document can be extended to support
publication of any event state, for which there exists an appropriate
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inline.

Brian Stucker wrote:

> Johnathan,
> 
> I'm not opposed to the client ID being transported in the registration 
> contact,
> but seeing as there were two drafts that already discuss that, I didn't see
> much point in rehashing it again in my draft. =)
> 
> I believe there are some important points that you're leaving out.
> 
> Besides being client generated (which reduces the requirements for 
> generating
> the ID) there are advantages to NOT mandating that the ID be expressed only
> as a parameter in the contact of a registration.
> 
> The idea behind the GUID (and hence the lack of a specific usecase in the
> -00 version of the document) is that it can be used for other activities.
> I'm not opposed to communicating the ID in a contact parameter as well as
> a regular header, I just don't view transporting it in a contact header
> as being always necessary or desirable depending on what you want to do
> with it.

I think the key thing we need to do first is figure out what it is we 
want to do with it (i.e., requirements!). I am principally interested in 
its usage with GRUU. In that usage, it is necessary to be carried in the 
contact header.

> 
> Sending it as a header allows for some interesting behavior to occur on
> non-registration requests:
> 
> - Can be used to eliminate duplicate subscriptions due to client's state
> being wiped out (ie. ignore the callid/tags of an existing subscription
> if the client ID matches the client ID of that subscription [and the other
> information about the subscription is the same]).

I'm not sure this is good behavior though. Its perfectly fine to have 
two subscriptions to the same thing from the same UA instance. If the 
state is wiped out and the client sends a second subscription, the first 
will either time out, or if the server sends a NOTIFY, the NOTIFY will 
time out or generatea a 481, both of which terminate the first 
subscription. This behavior seems fine.

> - Can be used to identify the client sending a request across networks
> where no other identifier is easily obtainable (NATs and the such), or
> BBUAs are present in the network.

OK, but whats the use case?


> - Can be used to identify the source of state information being pushed
> into the network (presence update, etc). This is one area where conveying
> the information in a contact seems a little cumbersome. The request may
> not be sent to the registrar, and a contact may otherwise be completely
> unnecessary.

OK, this is a fair point. There does appear to be a need in PUBLISH 
(more in pidf I think) for having a unique identifier of the source of 
the publication.

> 
> In the cases where registration behavior is controlled, it requires a 
> change
> to the registrar in order to capture the appropriate behavior that 
> should be
> taken wrt a client ID. Therefore, including the information as a header or
> in the contact seems moot to me as either mechanism can easily be made to
> work. Perhaps there's a usecase I'm not considering?

If you put it in a header, you would need to include additional 
parameters to correlate it to the specific Contact in a registration 
that it applies to. This can work, but its messy and unnecessary if you 
include the guid in the contact directly.

> 
> Additionally, I believe that there needs to be some convergence on the 
> approach
> taken by generating the identifier itself. The mechanism that is suggested
> in your draft seems to rely upon a server to generate the unique 
> identifier,

No, the instance ID is generated by the client. I just don't say how. I 
think there are specs that give guidance on generating unique IDs. I 
couldnt find an rfc reference, maybe it was somewhere else, but this is 
well-trod territory. I dont think our requirements for unique 
identifiers are different than anyone elses.

> whereas in the other two drafts, it is the client that generates the
> unique identifier. I believe there are advantages to each.

I think the client has to generate it always.

-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
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From exim@www1.ietf.org  Mon Feb 16 16:00:42 2004
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Date: Mon, 16 Feb 2004 15:56:39 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: hisham.khartabil@nokia.com
CC: fluffy@cisco.com, sip@ietf.org
Subject: Re: [Sip] draft-khartabil-sip-policy-uri-call-info-purpose-00
References: <2038BCC78B1AD641891A0D1AE133DBB701797753@esebe019.ntc.nokia.com>
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I think one could provide links to other policy documents, but this is 
best done via other purpose tags. "policy" is pretty general, and I 
think  you want to be as specific as possible here.

-Jonathan R.

hisham.khartabil@nokia.com wrote:

> Thanks.
> 
> Do you think that it will only be limited to conferences?
> 
> /Hisham
> 
> 
>>-----Original Message-----
>>From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
>>Cullen Jennings
>>Sent: 13.February.2004 18:49
>>To: sip@ietf.org
>>Subject: [Sip] draft-khartabil-sip-policy-uri-call-info-purpose-00
>>
>>
>>    
>>I like this draft, just one little NIT, I think it would be 
>>better to call
>>the tag conf-policy-info instead of just policy-info
>>
>>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
> 

-- 
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: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: RE: [Sip] SIP instance identifiers
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inline some more.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Monday, February 16, 2004 2:54 PM
> To: Stucker, Brian [NGC:B617:EXCH]
> Cc: sip@ietf.org
> Subject: Re: [Sip] SIP instance identifiers
> 
> 
> inline.
> 
> Brian Stucker wrote:
> 
> > Johnathan,
> > 
> > I'm not opposed to the client ID being transported in the 
> registration 
> > contact,
> > but seeing as there were two drafts that already discuss 
> that, I didn't see
> > much point in rehashing it again in my draft. =)
> > 
> > I believe there are some important points that you're leaving out.
> > 
> > Besides being client generated (which reduces the requirements for 
> > generating
> > the ID) there are advantages to NOT mandating that the ID 
> be expressed only
> > as a parameter in the contact of a registration.
> > 
> > The idea behind the GUID (and hence the lack of a specific 
> usecase in the
> > -00 version of the document) is that it can be used for 
> other activities.
> > I'm not opposed to communicating the ID in a contact 
> parameter as well as
> > a regular header, I just don't view transporting it in a 
> contact header
> > as being always necessary or desirable depending on what 
> you want to do
> > with it.
> 
> I think the key thing we need to do first is figure out what it is we 
> want to do with it (i.e., requirements!). I am principally 
> interested in 
> its usage with GRUU. In that usage, it is necessary to be 
> carried in the 
> contact header.
> 

I'm open to a number of requirements, but simply constraining the set of
things we ought to try to tackle simple to what's in GRUU seems to miss
the opportunity to kill multiple birds with one stone. I'd like to see
a solution that encapsulates as many usecases as possible, and I'm not
adverse to the idea of carrying the ID in the contact header itself. I
just don't think that's the ONLY place it makes sense to send it.

> > 
> > Sending it as a header allows for some interesting behavior 
> to occur on
> > non-registration requests:
> > 
> > - Can be used to eliminate duplicate subscriptions due to 
> client's state
> > being wiped out (ie. ignore the callid/tags of an existing 
> subscription
> > if the client ID matches the client ID of that subscription 
> [and the other
> > information about the subscription is the same]).
> 
> I'm not sure this is good behavior though. Its perfectly fine to have 
> two subscriptions to the same thing from the same UA instance. If the 
> state is wiped out and the client sends a second 
> subscription, the first 
> will either time out, or if the server sends a NOTIFY, the 
> NOTIFY will 
> time out or generatea a 481, both of which terminate the first 
> subscription. This behavior seems fine.

Sometimes it's OK, sometimes it's not. Let's give ourselves the option
to choose when we want this type of behavior (which I would think would
generally be rare considering we'd be sending an extra NOTIFY and holding
onto a subscription for no apparent reason). Using an ID to uniquely 
identify the client can help to reduce unwanted traffic. Is it the only
way it can be done? No. But it is a handy way to do it if we're not
adverse to including this type of information in requests.

> 
> > - Can be used to identify the client sending a request 
> across networks
> > where no other identifier is easily obtainable (NATs and 
> the such), or
> > BBUAs are present in the network.
> 
> OK, but whats the use case?

Can be used to correlate requests from the same client where IP
addresses may not be available at all nodes, and the context-matching
information may have been jumbled by a BBUA. Can use useful for matching
up calls to other non-call transactions, etc.

> 
> 
> > - Can be used to identify the source of state information 
> being pushed
> > into the network (presence update, etc). This is one area 
> where conveying
> > the information in a contact seems a little cumbersome. The 
> request may
> > not be sent to the registrar, and a contact may otherwise 
> be completely
> > unnecessary.
> 
> OK, this is a fair point. There does appear to be a need in PUBLISH 
> (more in pidf I think) for having a unique identifier of the 
> source of 
> the publication.
> 

Glad we agree on something. =)

> > 
> > In the cases where registration behavior is controlled, it 
> requires a 
> > change
> > to the registrar in order to capture the appropriate behavior that 
> > should be
> > taken wrt a client ID. Therefore, including the information 
> as a header or
> > in the contact seems moot to me as either mechanism can 
> easily be made to
> > work. Perhaps there's a usecase I'm not considering?
> 
> If you put it in a header, you would need to include additional 
> parameters to correlate it to the specific Contact in a registration 
> that it applies to. This can work, but its messy and 
> unnecessary if you 
> include the guid in the contact directly.
> 

This is a good point. In that case it makes sense to put it into the
contacts for the registration, but once again, I'm not trying to exclude
this option in my draft. It was already discussed elsewhere, so there seemed
no need to go back over it.

> > 
> > Additionally, I believe that there needs to be some 
> convergence on the 
> > approach
> > taken by generating the identifier itself. The mechanism 
> that is suggested
> > in your draft seems to rely upon a server to generate the unique 
> > identifier,
> 
> No, the instance ID is generated by the client. I just don't 
> say how. I 
> think there are specs that give guidance on generating unique IDs. I 
> couldnt find an rfc reference, maybe it was somewhere else, 
> but this is 
> well-trod territory. I dont think our requirements for unique 
> identifiers are different than anyone elses.
> 

True enough. I just wanted to put in some ground-rules so we avoid some
of the troubles that you find in poor implementations of callid/tag
generators
for the SIP stack. Sometimes it's helpful to put a baseline out there so
people
are at a minimum cognizant of what they're getting themselves into that may
not otherwise be obvious to the implementor.

> > whereas in the other two drafts, it is the client that generates the
> > unique identifier. I believe there are advantages to each.
> 
> I think the client has to generate it always.

Sometimes it makes sense for the server to generate it, especially if the 
server is a trusted third-party and the ID is used for some sort of
non-repudiation
tactic. 

> 
> -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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<P><FONT SIZE=2>inline some more.</FONT>
</P>

<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: Jonathan Rosenberg [<A HREF="mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</A>]</FONT>
<BR><FONT SIZE=2>&gt; Sent: Monday, February 16, 2004 2:54 PM</FONT>
<BR><FONT SIZE=2>&gt; To: Stucker, Brian [NGC:B617:EXCH]</FONT>
<BR><FONT SIZE=2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=2>&gt; Subject: Re: [Sip] SIP instance identifiers</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; inline.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Brian Stucker wrote:</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; Johnathan,</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; I'm not opposed to the client ID being transported in the </FONT>
<BR><FONT SIZE=2>&gt; registration </FONT>
<BR><FONT SIZE=2>&gt; &gt; contact,</FONT>
<BR><FONT SIZE=2>&gt; &gt; but seeing as there were two drafts that already discuss </FONT>
<BR><FONT SIZE=2>&gt; that, I didn't see</FONT>
<BR><FONT SIZE=2>&gt; &gt; much point in rehashing it again in my draft. =)</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; I believe there are some important points that you're leaving out.</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; Besides being client generated (which reduces the requirements for </FONT>
<BR><FONT SIZE=2>&gt; &gt; generating</FONT>
<BR><FONT SIZE=2>&gt; &gt; the ID) there are advantages to NOT mandating that the ID </FONT>
<BR><FONT SIZE=2>&gt; be expressed only</FONT>
<BR><FONT SIZE=2>&gt; &gt; as a parameter in the contact of a registration.</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; The idea behind the GUID (and hence the lack of a specific </FONT>
<BR><FONT SIZE=2>&gt; usecase in the</FONT>
<BR><FONT SIZE=2>&gt; &gt; -00 version of the document) is that it can be used for </FONT>
<BR><FONT SIZE=2>&gt; other activities.</FONT>
<BR><FONT SIZE=2>&gt; &gt; I'm not opposed to communicating the ID in a contact </FONT>
<BR><FONT SIZE=2>&gt; parameter as well as</FONT>
<BR><FONT SIZE=2>&gt; &gt; a regular header, I just don't view transporting it in a </FONT>
<BR><FONT SIZE=2>&gt; contact header</FONT>
<BR><FONT SIZE=2>&gt; &gt; as being always necessary or desirable depending on what </FONT>
<BR><FONT SIZE=2>&gt; you want to do</FONT>
<BR><FONT SIZE=2>&gt; &gt; with it.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I think the key thing we need to do first is figure out what it is we </FONT>
<BR><FONT SIZE=2>&gt; want to do with it (i.e., requirements!). I am principally </FONT>
<BR><FONT SIZE=2>&gt; interested in </FONT>
<BR><FONT SIZE=2>&gt; its usage with GRUU. In that usage, it is necessary to be </FONT>
<BR><FONT SIZE=2>&gt; carried in the </FONT>
<BR><FONT SIZE=2>&gt; contact header.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

<P><FONT SIZE=2>I'm open to a number of requirements, but simply constraining the set of</FONT>
<BR><FONT SIZE=2>things we ought to try to tackle simple to what's in GRUU seems to miss</FONT>
<BR><FONT SIZE=2>the opportunity to kill multiple birds with one stone. I'd like to see</FONT>
<BR><FONT SIZE=2>a solution that encapsulates as many usecases as possible, and I'm not</FONT>
<BR><FONT SIZE=2>adverse to the idea of carrying the ID in the contact header itself. I</FONT>
<BR><FONT SIZE=2>just don't think that's the ONLY place it makes sense to send it.</FONT>
</P>

<P><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; Sending it as a header allows for some interesting behavior </FONT>
<BR><FONT SIZE=2>&gt; to occur on</FONT>
<BR><FONT SIZE=2>&gt; &gt; non-registration requests:</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; - Can be used to eliminate duplicate subscriptions due to </FONT>
<BR><FONT SIZE=2>&gt; client's state</FONT>
<BR><FONT SIZE=2>&gt; &gt; being wiped out (ie. ignore the callid/tags of an existing </FONT>
<BR><FONT SIZE=2>&gt; subscription</FONT>
<BR><FONT SIZE=2>&gt; &gt; if the client ID matches the client ID of that subscription </FONT>
<BR><FONT SIZE=2>&gt; [and the other</FONT>
<BR><FONT SIZE=2>&gt; &gt; information about the subscription is the same]).</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I'm not sure this is good behavior though. Its perfectly fine to have </FONT>
<BR><FONT SIZE=2>&gt; two subscriptions to the same thing from the same UA instance. If the </FONT>
<BR><FONT SIZE=2>&gt; state is wiped out and the client sends a second </FONT>
<BR><FONT SIZE=2>&gt; subscription, the first </FONT>
<BR><FONT SIZE=2>&gt; will either time out, or if the server sends a NOTIFY, the </FONT>
<BR><FONT SIZE=2>&gt; NOTIFY will </FONT>
<BR><FONT SIZE=2>&gt; time out or generatea a 481, both of which terminate the first </FONT>
<BR><FONT SIZE=2>&gt; subscription. This behavior seems fine.</FONT>
</P>

<P><FONT SIZE=2>Sometimes it's OK, sometimes it's not. Let's give ourselves the option</FONT>
<BR><FONT SIZE=2>to choose when we want this type of behavior (which I would think would</FONT>
<BR><FONT SIZE=2>generally be rare considering we'd be sending an extra NOTIFY and holding</FONT>
<BR><FONT SIZE=2>onto a subscription for no apparent reason). Using an ID to uniquely </FONT>
<BR><FONT SIZE=2>identify the client can help to reduce unwanted traffic. Is it the only</FONT>
<BR><FONT SIZE=2>way it can be done? No. But it is a handy way to do it if we're not</FONT>
<BR><FONT SIZE=2>adverse to including this type of information in requests.</FONT>
</P>

<P><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; - Can be used to identify the client sending a request </FONT>
<BR><FONT SIZE=2>&gt; across networks</FONT>
<BR><FONT SIZE=2>&gt; &gt; where no other identifier is easily obtainable (NATs and </FONT>
<BR><FONT SIZE=2>&gt; the such), or</FONT>
<BR><FONT SIZE=2>&gt; &gt; BBUAs are present in the network.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; OK, but whats the use case?</FONT>
</P>

<P><FONT SIZE=2>Can be used to correlate requests from the same client where IP</FONT>
<BR><FONT SIZE=2>addresses may not be available at all nodes, and the context-matching</FONT>
<BR><FONT SIZE=2>information may have been jumbled by a BBUA. Can use useful for matching</FONT>
<BR><FONT SIZE=2>up calls to other non-call transactions, etc.</FONT>
</P>

<P><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; - Can be used to identify the source of state information </FONT>
<BR><FONT SIZE=2>&gt; being pushed</FONT>
<BR><FONT SIZE=2>&gt; &gt; into the network (presence update, etc). This is one area </FONT>
<BR><FONT SIZE=2>&gt; where conveying</FONT>
<BR><FONT SIZE=2>&gt; &gt; the information in a contact seems a little cumbersome. The </FONT>
<BR><FONT SIZE=2>&gt; request may</FONT>
<BR><FONT SIZE=2>&gt; &gt; not be sent to the registrar, and a contact may otherwise </FONT>
<BR><FONT SIZE=2>&gt; be completely</FONT>
<BR><FONT SIZE=2>&gt; &gt; unnecessary.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; OK, this is a fair point. There does appear to be a need in PUBLISH </FONT>
<BR><FONT SIZE=2>&gt; (more in pidf I think) for having a unique identifier of the </FONT>
<BR><FONT SIZE=2>&gt; source of </FONT>
<BR><FONT SIZE=2>&gt; the publication.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

<P><FONT SIZE=2>Glad we agree on something. =)</FONT>
</P>

<P><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; In the cases where registration behavior is controlled, it </FONT>
<BR><FONT SIZE=2>&gt; requires a </FONT>
<BR><FONT SIZE=2>&gt; &gt; change</FONT>
<BR><FONT SIZE=2>&gt; &gt; to the registrar in order to capture the appropriate behavior that </FONT>
<BR><FONT SIZE=2>&gt; &gt; should be</FONT>
<BR><FONT SIZE=2>&gt; &gt; taken wrt a client ID. Therefore, including the information </FONT>
<BR><FONT SIZE=2>&gt; as a header or</FONT>
<BR><FONT SIZE=2>&gt; &gt; in the contact seems moot to me as either mechanism can </FONT>
<BR><FONT SIZE=2>&gt; easily be made to</FONT>
<BR><FONT SIZE=2>&gt; &gt; work. Perhaps there's a usecase I'm not considering?</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; If you put it in a header, you would need to include additional </FONT>
<BR><FONT SIZE=2>&gt; parameters to correlate it to the specific Contact in a registration </FONT>
<BR><FONT SIZE=2>&gt; that it applies to. This can work, but its messy and </FONT>
<BR><FONT SIZE=2>&gt; unnecessary if you </FONT>
<BR><FONT SIZE=2>&gt; include the guid in the contact directly.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

<P><FONT SIZE=2>This is a good point. In that case it makes sense to put it into the</FONT>
<BR><FONT SIZE=2>contacts for the registration, but once again, I'm not trying to exclude</FONT>
<BR><FONT SIZE=2>this option in my draft. It was already discussed elsewhere, so there seemed</FONT>
<BR><FONT SIZE=2>no need to go back over it.</FONT>
</P>

<P><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; Additionally, I believe that there needs to be some </FONT>
<BR><FONT SIZE=2>&gt; convergence on the </FONT>
<BR><FONT SIZE=2>&gt; &gt; approach</FONT>
<BR><FONT SIZE=2>&gt; &gt; taken by generating the identifier itself. The mechanism </FONT>
<BR><FONT SIZE=2>&gt; that is suggested</FONT>
<BR><FONT SIZE=2>&gt; &gt; in your draft seems to rely upon a server to generate the unique </FONT>
<BR><FONT SIZE=2>&gt; &gt; identifier,</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; No, the instance ID is generated by the client. I just don't </FONT>
<BR><FONT SIZE=2>&gt; say how. I </FONT>
<BR><FONT SIZE=2>&gt; think there are specs that give guidance on generating unique IDs. I </FONT>
<BR><FONT SIZE=2>&gt; couldnt find an rfc reference, maybe it was somewhere else, </FONT>
<BR><FONT SIZE=2>&gt; but this is </FONT>
<BR><FONT SIZE=2>&gt; well-trod territory. I dont think our requirements for unique </FONT>
<BR><FONT SIZE=2>&gt; identifiers are different than anyone elses.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

<P><FONT SIZE=2>True enough. I just wanted to put in some ground-rules so we avoid some</FONT>
<BR><FONT SIZE=2>of the troubles that you find in poor implementations of callid/tag generators</FONT>
<BR><FONT SIZE=2>for the SIP stack. Sometimes it's helpful to put a baseline out there so people</FONT>
<BR><FONT SIZE=2>are at a minimum cognizant of what they're getting themselves into that may</FONT>
<BR><FONT SIZE=2>not otherwise be obvious to the implementor.</FONT>
</P>

<P><FONT SIZE=2>&gt; &gt; whereas in the other two drafts, it is the client that generates the</FONT>
<BR><FONT SIZE=2>&gt; &gt; unique identifier. I believe there are advantages to each.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I think the client has to generate it always.</FONT>
</P>

<P><FONT SIZE=2>Sometimes it makes sense for the server to generate it, especially if the </FONT>
<BR><FONT SIZE=2>server is a trusted third-party and the ID is used for some sort of non-repudiation</FONT>
<BR><FONT SIZE=2>tactic. </FONT>
</P>

<P><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; -Jonathan R.</FONT>
<BR><FONT SIZE=2>&gt; -- </FONT>
<BR><FONT SIZE=2>&gt; Jonathan D. Rosenberg, Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 600 Lanidex Plaza</FONT>
<BR><FONT SIZE=2>&gt; Chief Technology Officer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Parsippany, NJ 07054-2711</FONT>
<BR><FONT SIZE=2>&gt; dynamicsoft</FONT>
<BR><FONT SIZE=2>&gt; jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FAX:&nbsp;&nbsp; (973) 952-5050</FONT>
<BR><FONT SIZE=2>&gt; <A HREF="http://www.jdrosen.net" TARGET="_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) 952-5000</FONT>
<BR><FONT SIZE=2>&gt; <A HREF="http://www.dynamicsoft.com" TARGET="_blank">http://www.dynamicsoft.com</A></FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

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</HTML>
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From exim@www1.ietf.org  Mon Feb 16 18:54:37 2004
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Subject: RE: [Sip] Comments on history-info
Date: Mon, 16 Feb 2004 18:49:59 -0500
Message-ID: <4A3384433CE2AB46A63468CB207E209DB1A194@zoe.office.snowshore.com>
Thread-Topic: [Sip] Comments on history-info
Thread-Index: AcP0nVvv5H5tdxfTRFqHJsqonTp2AwAQXKyA
From: "Eric Burger" <eburger@snowshore.com>
To: "Mary Barnes" <mary.barnes@nortelnetworks.com>,
        "SIP List (E-mail)" <sip@ietf.org>
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I'm not changing any of the functionality.  As Rohan says, it is already =
in the spec.  I'm suggesting we do not allow a proxy to *modify* an =
existing history-info header.  A proxy can (and should) *insert* new =
history-info headers.  It is because we have the index tag that the =
modification restriction should not be a problem.

> -----Original Message-----
> From: Mary Barnes [mailto:mary.barnes@nortelnetworks.com]
> Sent: Monday, February 16, 2004 9:58 AM
> To: Eric Burger; SIP List (E-mail)
> Subject: RE: [Sip] Comments on history-info
>=20
>=20
> Eric,
>=20
> History-Info is a  bit different than the SMTP (at least as best as I
> understand SMTP not being an expert in that area), since the=20
> retargeting in
> SIP is more complicated that SMTP due to forking, etc.  In=20
> order to be able
> to accurately reflecting the forking, we've added the index. =20
> The index is
> used to generate subsequent History-Info entries.  In=20
> addition, there's the
> Reason that gets added to the targeted-to URIs to reflect the=20
> reason for the
> retargeting. So, yes History-Info does change History in the=20
> sense that we
> embellish the Request-URIs that are "visited" with additional=20
> information
> that seems to be useful to a variety of end applications. =20
> So, I think what
> you're proposing would only provide a subset of the=20
> functionality that we're
> after.
>=20
> Mary.=20
>=20
> -----Original Message-----
> From: Eric Burger [mailto:eburger@snowshore.com]
> Sent: Sunday, February 15, 2004 5:00 PM
> To: Barnes, Mary [NGC:B622:EXCH]; SIP List (E-mail)
> Subject: RE: [Sip] Comments on history-info
>=20
>=20
> Excellent.
>=20
> Thinking about it a bit more, is there any reason to ALLOW=20
> stuffing multiple
> history info info's on a single line?  The idea going back to=20
> SMTP was that
> you are only allowed to add history, not change it.  Putting multiple
> history info's in a single history-info header, by definition, changes
> history.
>=20
> > -----Original Message-----
> > From: Mary Barnes [mailto:mary.barnes@nortelnetworks.com]
> > Sent: Sunday, February 15, 2004 3:18 PM
> > To: Eric Burger; SIP List (E-mail)
> > Subject: RE: [Sip] Comments on history-info
> >=20
> >=20
> > Eric,
> >=20
> > Although Rohan responded to your "Big question:" part of this email:
> > http://www1.ietf.org/mail-archive/working-groups/sip/current/m
> > sg08998.html
> > it doesn't appear that I ever responded to the remaining=20
> > points on the list.
> > Thank you for reviewing the draft and providing detailed=20
> > comments.   In
> > updating the draft, I considered your comments as indicated=20
> > below [MB].
> >=20
> > Regards,
> > Mary.=20
> >=20
> > -----Original Message-----
> > From: Eric Burger [mailto:eburger@snowshore.com]
> > Sent: Thursday, November 06, 2003 3:44 AM
> > To: SIP List (E-mail)
> > Subject: [Sip] Comments on history-info
> >=20
> >=20
> > Big question:
> > Why don't we use multiple History-Info headers, instead of a=20
> > (potentially
> > VERY long) single header?
> >=20
> > I would argue for multiple headers on two grounds.
> >=20
> > First, the idea of a proxy modifying history information=20
> doesn't sound
> > smart.  If it screws up, it is hard to figure out who screwed up.
> >=20
> > Second, the index parameter frees us from worrying about=20
> > header order, which
> > is a VERY good thing.
> >=20
> >=20
> >=20
> > Overview:
> > While the motivation for History-Info is for the initiator of=20
> > a request to
> > know what happened to a request, it is also of great use to=20
> > the recipient
> > (End of Section 2 overview paragraph).
> >=20
> > [MB]: I've clarified this.=20
> >=20
> > Section 1.2:
> > How is History-Info any different than Via w.r.t. security?
> >=20
> > [MB]: It's quite similar actually, so I've added a statement=20
> > to that effect
> > in this section.  I think the only difference being the=20
> > dependency on the
> > Index written by the previous hop to generate the index for=20
> > the current hop.
> > I do think the general M2M and M2E security solution would also be
> > applicable to Via if TLS were not deemed to be sufficient=20
> > (i.e if there were
> > some reason why you would not want some intermediaries to be=20
> > able to see the
> > Via added by an "untrusted" intermediary). =20
> >=20
> > Nits:
> > Section 2.5, first paragraph, last sentence:
> > this would like be a local
> > this would likely be a local
> >                ^^
> > [MB]: Fixed.=20
> >=20
> > Appendix B, message F8, extra space before last angle-bracket in the
> > History-Info line.
> >=20
> > [MB]: Fixed.
> >=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
>=20


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Subject: [Sip] Question regarding resource-priority header
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Hi,

In draft-ietf-sip-resource-priority-01.txt, it looks like there is a 
printing mistake in the grammar defined for Resource-Priority header . Its 
given as 
       Resource-Priority  =  "Resource-Priority" HCOLON
                              (*COMMA Resource-value)
 
But I guess, no header can start with a COMMA . It should actually be 
       Resource-Priority  =  "Resource-Priority" HCOLON
                              Resource-value(*COMMA Resource-value)

This is the way the grammar is given for Accept-Resource-Priority. 
Can anyone confirm on this pls? 

thanks in advance... 

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Content-Type: text/html; charset="US-ASCII"


<br><font size=2 face="sans-serif">Hi,</font>
<br>
<br><font size=2 face="Courier New">In draft-ietf-sip-resource-priority-01.txt,
it looks like there is a printing mistake in the grammar defined for Resource-Priority
header . Its given as </font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp;Resource-Priority
&nbsp;= &nbsp;&quot;Resource-Priority&quot; HCOLON</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; (*COMMA
Resource-value)</font>
<br><font size=2 face="Courier New">&nbsp;</font>
<br><font size=2 face="Courier New">But I guess, no header can start with
a COMMA . It should actually be </font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp;Resource-Priority
&nbsp;= &nbsp;&quot;Resource-Priority&quot; HCOLON</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; </font><font size=2 color=blue face="Courier New">Resource-value</font><font size=2 face="Courier New">(*COMMA
Resource-value)</font>
<br>
<br><font size=2 face="Courier New">This is the way the grammar is given
for Accept-Resource-Priority. </font>
<br><font size=2 face="Courier New">Can anyone confirm on this pls? </font>
<br>
<br><font size=2 face="Courier New">thanks in advance... </font>
<br>
--=_alternative 004C9F4365256E3C_=--

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From: Paul Kyzivat <pkyzivat@cisco.com>
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Cullen,

This is an attractive proposal because it has potential to make sip 
implementations simpler. But before I can answer I need clarification on 
one thing:

Does this apply to content referenced and fetched via content-indirection?

This proposal only has value if my mime processor *never* needs to deal 
with any content-transfer-encoding other than binary. If content 
referenced by content indirection still might use other encodings, then 
that value is lost. Unfortunately, given the potential range of 
transports that might be used with content indirection, I don't know if 
requiring binary can be done in general.

	Paul

Cullen Jennings wrote:
>     
> I submitted a draft on binary encoding for SIP. This improves
> interoperability, improves speed, decreases sizes of messages, and generally
> makes life better for SIP which will no doubt eventually help with World
> Hunger. I need feedback, please let me know:
> 
> a) Yah, lets do it
> 
> b) No this is a stupid ideas and here are the reasons why
> 
> c) There has not been many list comments, this draft involves some very
> complex issues. I need a year or two to develop an opinion on the single
> sentence inside it.
> 
> d) I agree with Cowboy Willis
> 
> Until it arrives in the archives, you can find it at
> 
> http://www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.html
> 
> or, if you prefer ASCII encodings of everything,
> 
> www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.txt
> 
> Thanks, Cullen
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


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Subject: RE: [Sip] FW: I-D ACTION:draft-khartabil-sip-auth-analysis-00.txt
Date: Tue, 17 Feb 2004 16:57:17 +0200
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Scott,

Thanks for the comments. Responses inline...

> -----Original Message-----
> From: ext Scott Lawrence [mailto:slawrence@pingtel.com]
> Sent: 16.February.2004 22:10
> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> Cc: sip@ietf.org
> Subject: Re: [Sip] FW: I-D
> ACTION:draft-khartabil-sip-auth-analysis-00.txt
>=20
>=20
> On Fri, 2004-02-06 at 02:48, hisham.khartabil@nokia.com wrote:
> > Having tried to implement HTTP Authentication for SIP, we faced=20
> > a few problems. This draft tries to highlight those problems and=20
> > in some cases suggests a solution.
>=20
> Comments on draft-khartabil-sip-auth-analysis-00
>=20
> > 2. Use of 401 and 407 Responses, www-authenticate and
> >  proxy-authenticate headers
> >  [...]
> >  The use of 401 and 407 must be more normative. Text should read: A
> >  UAS that require authentication MUST use 401 in a response and MUST
> >  challenge with a www-authenticate header. Registrars and redirect
> >  servers MUST use 401 and www-authenticate header. Proxies MUST use
> >  407 and proxy-authenticate header.
>=20
> That would be a clear improvement, I think.
>=20
> > 3. Rendering to user
>=20
> It's not clear to me what it is you're objecting to in this section,
> except perhaps that the user interface guideline is poorly rendered.

Yes, this is the point I was trying to make. The fact that a username =
and password for a realm is preconfigured does not mean that the user =
does not get to re-enter those credentials if the server rejected the =
ones sent earlier.

>=20
> The real requirement, I think, is that if the user is being asked for
> credentials, they should be shown the realm value at that time; a fact
> that is made clear in RFC 2617 (sec 3.2.1):
>=20
>   realm
>     A string to be displayed to users so they know which username and
>     password to use.

I don't think this takes into consideration pre-configuration.

>=20
> > 4. Rejected or not
> >
> >  Section 22.1 of RFC 3261 says: "A UAC MUST NOT re-attempt requests
> >  with the credentials that have just been rejected (though=20
> the request
> >  may be retried if the nonce was stale)".
> >
> >  The problem with this is that how do you know if the=20
> request has been
> >  rejected or it is just the next hop proxy challenging with the same
> >  realm?
>=20
> This case couldn't come up in HTTP, since proxy authentication in HTTP
> is hop-by-hop, and 407 challenges are never passed back to the user
> agent by a proxy. =20

In SIP, unfortunately it is possible.

> However, the reset of the description of realm is
> good advice:
>=20
>                 ... This string should contain at least the name of
>    the host performing the authentication and might additionally
>    indicate the collection of users who might have access. An example
>    might be "registered_users@gotham.news.com".

Can we mandate that for SIP??

>=20
> > 5. Use of To-header vs. Remote URI (Request-URI)
> >
> >  Section 22.2 of RFC 3261 says: "The UAC may require input from the
> >  originating user before proceeding.  Once authentication=20
> credentials
> >  have been supplied (either directly by the user, or=20
> discovered in an
> >  internal key-ring), UAs SHOULD cache the credentials for a given
> >  value of the To header field and "realm" and attempt to=20
> re-use these
> >  values on the next request for that destination".
> >
> >  It is more appropriate to cache request-URI, and not rely=20
> on what is
> >  in the To-header.
>=20
> Agreed - the To header value is not protected by the hash, so
> associating it is not appropriate.  In addition, this paragraph
> encourages the existing confusion between these two values
> (admittedly, they are often the same, but it is the Request URI that
> is used in authentication).
>=20
> > 7. Caching outbound proxy credentials
> >
> >  Section 22.3 of RFC 3261 says: "if a UA is configured with=20
> the realm
> >  of its local outbound proxy, when one exists, then the UA MAY cache
> >  credentials for that realm across dialogs".
> >
> >  It cannot be assumed that the domain name of a configured outbound
> >  proxy is its realm. Therefore a terminal configured with only the
> >  outbound proxy URI cannot and MUST NOT cache credentials=20
> or any proxy
> >  challenge across dialogs.
> >
> >  This introduces the problem of multiple proxies in a chain
> >  challenging with the same realm. How does a UAC know that the
> >  challenge was from the outbound proxy and not from a=20
> downstream proxy
> >  with the same realm as the outbound proxy? Can we mandate that the
> >  outbound proxy must have a unique realm?
>=20
> As a practical matter, if you have the same username and password in
> both domains it doesn't matter - if you don't, then it's an impossibly
> clumsy situation (this problem is part of the reason that we defined
> proxy authentication to by hop-by-hop in HTTP in the first place).

Perhaps I was not clear in my text. Imagine this scenario: I have cached =
my credentials for an outbound proxy with realm proxy.nokia.com from an =
earlier transaction. Now I send a new INVITE routed through the outbound =
proxy that gets a 407 from a second proxy with the same realm. How do I =
know if the 407 came from the outbound proxy (the username password =
might have changed in the meantime) or from the next hop proxy?

>=20
> > 9. The realm Issue
> >
> >  Section 22.3 of RFC 3261 says: "When multiple proxies are used in a
> >  chain, a Proxy-Authorization header field value MUST NOT=20
> be consumed
> >  by any proxy whose realm does not match the "realm" parameter
> >  specified in that value".
> >
> >  Another question: Is it possible that multiple proxies=20
> with the same
> >  realm are placed in a chain? I.e. the first proxy challenges, user
> >  provide correct authentication. That proxy authenticates=20
> and forwards
> >  the request to a down stream proxy. That down stream proxy has the
> >  same realm as first proxy. There are a few issues here:
>=20
> >  examining the nonce is the only solution.
>=20
> Essentially, I think that putting two proxies in a row with the same
> realm is a difficult configuration to diagnose.  If administrators do
> this, they are asking for trouble.  But there is another approach I
> don't think you examined.
>=20
> >  Another issue is at the UAC side. If 2 proxies are in a chain and
> >  share a realm, how does the UAC know, when the second proxy
> >  challenges, that it is not the first proxy re-challenging=20
> because the
> >  credentials provided to it were wrong?
>=20
> If the proxy is using the 2617 version of Digest (with the qop
> attribute) as opposed to the 1867 version, then the UA can tell
> the difference.  The message flow looks like this:
>=20
>         UA                    P1                  P2
>     a    +------REQ 1--------->|                   |
>     b    |<----407(1)----------+                   |
>     c    +------REQ 2 -------->+----REQ 2--------->|
>     e    |<---407(2)-----------+<-----407(2)-------|
>          +------REQ 3--------->+----REQ 3--------->|
>=20
>  a) initial request
>  b) challenge from P1, with the qop parameter specified
>  c) retried request, with a qop parameter, and therefor also a cnonce
>     chosen by the UA; it is passed on (now without=20
> authentication) to P2
>  d) P2 challenges, also specifying a qop parameter
>     P2 challenge is forwarded by P1, with the WWW-Authenticate
>     header as provided by P2, but also with an Authentication-Info
>     provided by P1.
>=20
> The UA can recognize that authentication to P1 was successful, because
> P1 returned an Authentication-Info header that contained the cnonce
> and nc values from message c (REQ 2), and a correct response hash to
> show that P1 knows the authentication secret.

This is not mandated for SIP entities, maybe it aught to be.

>=20
> > 10. Authentication-Info
> >
> >  RFC 3261 allows the usage of Authentication-Info header. The BNF in
> >  RFC 3261 allows multiple authentication-info headers where RFC 2617
> >  allows only one. Is it only the terminating UAS that is allowed to
> >  insert this header. If so, why allow multiples to be=20
> present. If not,
> >  how does the UAC know which proxy added this header? It cannot know
> >  since there is no parameter indicating so, not even nonce.
>=20
> As I noted earlier, 2617 allows only one because proxy authentication
> is hop-by-hop, but the cnonce value allows the UA to disambiguate them
> (if they are carefully constructed, but then they should be anyway).

SIP BNF allows multiples of those headers. Is this a BNF bug? If not, =
then some Authentication-Info parameters need to be mandated.

/Hisham

>=20
> All this is even more reason to use qop (which was the intent of 2617
> in the first place).
>=20
> --=20
> Scott Lawrence       =20
>   Pingtel Corp.  =20
>=20
>=20
>=20

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



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Date: Tue, 17 Feb 2004 07:31:28 -0800
Subject: Re: [Sip] SIP instance identifiers
From: Cullen Jennings <fluffy@cisco.com>
To: Brian Stucker <bstucker@nortelnetworks.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>, <sip@ietf.org>
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I can imagine cases where we might want to be able to route by the instance
ID. I think the +sip.instance approach is a better than the general contact
header I proposed. It provides more options for routing and has no
disadvantages.

Cullen

On 2/16/04 7:57 AM, Jonathan Rosenberg wrote:

> Then, the question is whether to use a regular Contact header field
> parameter as Cullen has done, or a callee cap parameter as I have done?
> The callee cap approach allows for caller preferences to be applied to
> routing to a specific instance. Now, a good question is whether such a
> thing is needed or even a good idea, if we have GRUU as the ideal way to
> route to a UA instance. I can go either way, but welcome thoughts on the
> topic. 



--B_3159847888_1111971
Content-type: text/html; charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [Sip] SIP instance identifiers</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><BR>
I can imagine cases where we might want to be able to route by the instance=
 ID. I think the +sip.instance approach is a better than the general contact=
 header I proposed. It provides more options for routing and has no disadvan=
tages.<BR>
<BR>
Cullen<BR>
<BR>
On 2/16/04 7:57 AM, Jonathan Rosenberg wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0p=
x'>Then, the question is whether to use a regular Contact header field <BR>
parameter as Cullen has done, or a callee cap parameter as I have done? <BR=
>
The callee cap approach allows for caller preferences to be applied to <BR>
routing to a specific instance. Now, a good question is whether such a <BR>
thing is needed or even a good idea, if we have GRUU as the ideal way to <B=
R>
route to a UA instance. I can go either way, but welcome thoughts on the <B=
R>
topic. <BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0=
px'><BR>
</SPAN></FONT>
</BODY>
</HTML>


--B_3159847888_1111971--


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From exim@www1.ietf.org  Tue Feb 17 13:15:47 2004
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Subject: [Sip] Abstraction:  SIP instance identifiers requirements
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Based on the discussion of "routing" based on instance identifiers, I 
think we may be getting involved in triggerwork that may become more 
complex than is actually required, because we're all reasoning from 
secondary positions. So I'd like to take a step back and consider how we 
got here, and what we think we're trying to accomplish.

I see four high-level cases around request routing that illustrate 
different aspects of this discussion.

1) General case: We wish to address a request to an AOR, such that any 
contact at that AOR might respond. This is basic 2543-type routing behavior.

2) Capability case: We wish to address a request to an AOR, such that 
any contact capable of satisfying the request in a way that we like 
might respond. This behavior is brought out in callee-caps and 
caller-preferences and the accept/supported/required syntax of 3261.

3) Followups to a specific instance: We have previously done something 
(such as case 1 or 2) such that the operation of interest has resolved 
to a specific contact instance associated with an AoR, and we wish 
subsequent operations (for example, a transfer) to reach that same 
specific contact instance, even if that specific contact instance is 
hidden behind an indirection layer (firewall, anonymizer, etc.). As I 
understand it, this is the heart of the GRUU concept.

4) Selection of a specific instance in an initial request: We have a 
request that we wish to reach one specific instance of a contact 
associated with an AoR. It has been proposed that we accomplish this by 
binding some sort of instance tag to the AoR. It has been proposed that 
we use some sort of URI convention (like the msuri), +based 
subaddressing, or that we use a header convention for binding the 
instance tag to the AoR. To my knowledge, we haven't really described 
the requirement or use case for reaching a specific instance on an 
initial request, but I can think of several offhand (including diagnostics).

It seems evident that there is a huge overlap between the requirements 
of 3 and 4. Indeed, I think #3 (GRUU) came out of early ideas I had on 
using a REGISTER-response extension to assign an instance-AoR to a 
Contact. It would appear that IF each contact for an AoR were to have an 
associated GRUU, and that knowledge of those GRUU assigments were 
available to a requestor, that the requester could use the 
target-specific GRUU to achieve the use cases I currently imagine for #4.


What am I missing here?

--
Dean

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Subject: [Sip] Re: draft-ietf-sip-resource-priority-02 - syntax
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Henning,

(Splitting comments by subject.) There are still problems with the ABNF, 
around the usage of "*" to indicate repetition. As it stands, it accepts:

     Resource-Priority: f.1,,,-.-
     Resource-Priority: 9.a f.-

but it doesn't accept:

     Resource-Priority: foo.bar, foo-1.bar-1, foo-1.bar-2

Assuming the intent is to permit more than two values, separation of
values by exactly one comma, and namespaces and priorities that can
include multiple dashes and alphanums, then I think the following is 
what you want:

     Resource-Priority  = "Resource-Priority" HCOLON
                          Resource-value *(COMMA Resource-value)
...
     namespace          = *(alphanum / "-")
     r-priority         = *(alphanum / "-")

ABNF for Accept-Resource-Priority is similarly flawed. It should be:

     Accept-Resource-Priority = "Accept-Resource-Priority" HCOLON
                                [Resource-value *(COMMA Resource-value)]


	Paul


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Henning,

OPTIONS handling: there is good explanation of options processing in a
UAS, but it is very UAS specific. There isn't any discussion of options
processing in proxies that support this extension. It is possible to get
a proxy to respond to an options request (using Max-Forwards). This
might be used in testing, just a strict mode might.

I think what a proxy should return is pretty much the same as what a UAS 
returns.

	Paul


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I think that your list is missing the opportunity to define this instance 
ID in such a way that it can be used to identify a particular endpoint 
for other operations beyond simply routing. The reuse potential seems 
very good if we don't lock the instance identifier down to simply
routing operations, and instead open it up for other general-purpose
operations.

That is at the heart of my draft. Don't limit yourself to one application
when with a small amount of work we can use this to solve other problems
both current and future. Likewise, don't exclude routing operations from
making use of the instance identifier either.

Are you concerned that opening this up to having the instance identifier
transported (in addition to the registered contact) outside of the 
registered contact is really going to be that complex? 

Regards,

Brian



> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Tuesday, February 17, 2004 12:11 PM
> To: Cullen Jennings
> Cc: Stucker, Brian [NGC:B617:EXCH]; Jonathan Rosenberg; sip@ietf.org
> Subject: Abstraction: SIP instance identifiers requirements
> 
> 
> 
> Based on the discussion of "routing" based on instance identifiers, I 
> think we may be getting involved in triggerwork that may become more 
> complex than is actually required, because we're all reasoning from 
> secondary positions. So I'd like to take a step back and 
> consider how we 
> got here, and what we think we're trying to accomplish.
> 
> I see four high-level cases around request routing that illustrate 
> different aspects of this discussion.
> 
> 1) General case: We wish to address a request to an AOR, such 
> that any 
> contact at that AOR might respond. This is basic 2543-type 
> routing behavior.
> 
> 2) Capability case: We wish to address a request to an AOR, such that 
> any contact capable of satisfying the request in a way that we like 
> might respond. This behavior is brought out in callee-caps and 
> caller-preferences and the accept/supported/required syntax of 3261.
> 
> 3) Followups to a specific instance: We have previously done 
> something 
> (such as case 1 or 2) such that the operation of interest has 
> resolved 
> to a specific contact instance associated with an AoR, and we wish 
> subsequent operations (for example, a transfer) to reach that same 
> specific contact instance, even if that specific contact instance is 
> hidden behind an indirection layer (firewall, anonymizer, etc.). As I 
> understand it, this is the heart of the GRUU concept.
> 
> 4) Selection of a specific instance in an initial request: We have a 
> request that we wish to reach one specific instance of a contact 
> associated with an AoR. It has been proposed that we 
> accomplish this by 
> binding some sort of instance tag to the AoR. It has been 
> proposed that 
> we use some sort of URI convention (like the msuri), +based 
> subaddressing, or that we use a header convention for binding the 
> instance tag to the AoR. To my knowledge, we haven't really described 
> the requirement or use case for reaching a specific instance on an 
> initial request, but I can think of several offhand 
> (including diagnostics).
> 
> It seems evident that there is a huge overlap between the 
> requirements 
> of 3 and 4. Indeed, I think #3 (GRUU) came out of early ideas 
> I had on 
> using a REGISTER-response extension to assign an instance-AoR to a 
> Contact. It would appear that IF each contact for an AoR were 
> to have an 
> associated GRUU, and that knowledge of those GRUU assigments were 
> available to a requestor, that the requester could use the 
> target-specific GRUU to achieve the use cases I currently 
> imagine for #4.
> 
> 
> What am I missing here?
> 
> --
> Dean
> 

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<P><FONT SIZE=2>I think that your list is missing the opportunity to define this instance </FONT>
<BR><FONT SIZE=2>ID in such a way that it can be used to identify a particular endpoint </FONT>
<BR><FONT SIZE=2>for other operations beyond simply routing. The reuse potential seems </FONT>
<BR><FONT SIZE=2>very good if we don't lock the instance identifier down to simply</FONT>
<BR><FONT SIZE=2>routing operations, and instead open it up for other general-purpose</FONT>
<BR><FONT SIZE=2>operations.</FONT>
</P>

<P><FONT SIZE=2>That is at the heart of my draft. Don't limit yourself to one application</FONT>
<BR><FONT SIZE=2>when with a small amount of work we can use this to solve other problems</FONT>
<BR><FONT SIZE=2>both current and future. Likewise, don't exclude routing operations from</FONT>
<BR><FONT SIZE=2>making use of the instance identifier either.</FONT>
</P>

<P><FONT SIZE=2>Are you concerned that opening this up to having the instance identifier</FONT>
<BR><FONT SIZE=2>transported (in addition to the registered contact) outside of the </FONT>
<BR><FONT SIZE=2>registered contact is really going to be that complex? </FONT>
</P>

<P><FONT SIZE=2>Regards,</FONT>
</P>

<P><FONT SIZE=2>Brian</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: Dean Willis [<A HREF="mailto:dean.willis@softarmor.com">mailto:dean.willis@softarmor.com</A>]</FONT>
<BR><FONT SIZE=2>&gt; Sent: Tuesday, February 17, 2004 12:11 PM</FONT>
<BR><FONT SIZE=2>&gt; To: Cullen Jennings</FONT>
<BR><FONT SIZE=2>&gt; Cc: Stucker, Brian [NGC:B617:EXCH]; Jonathan Rosenberg; sip@ietf.org</FONT>
<BR><FONT SIZE=2>&gt; Subject: Abstraction: SIP instance identifiers requirements</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Based on the discussion of &quot;routing&quot; based on instance identifiers, I </FONT>
<BR><FONT SIZE=2>&gt; think we may be getting involved in triggerwork that may become more </FONT>
<BR><FONT SIZE=2>&gt; complex than is actually required, because we're all reasoning from </FONT>
<BR><FONT SIZE=2>&gt; secondary positions. So I'd like to take a step back and </FONT>
<BR><FONT SIZE=2>&gt; consider how we </FONT>
<BR><FONT SIZE=2>&gt; got here, and what we think we're trying to accomplish.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I see four high-level cases around request routing that illustrate </FONT>
<BR><FONT SIZE=2>&gt; different aspects of this discussion.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 1) General case: We wish to address a request to an AOR, such </FONT>
<BR><FONT SIZE=2>&gt; that any </FONT>
<BR><FONT SIZE=2>&gt; contact at that AOR might respond. This is basic 2543-type </FONT>
<BR><FONT SIZE=2>&gt; routing behavior.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 2) Capability case: We wish to address a request to an AOR, such that </FONT>
<BR><FONT SIZE=2>&gt; any contact capable of satisfying the request in a way that we like </FONT>
<BR><FONT SIZE=2>&gt; might respond. This behavior is brought out in callee-caps and </FONT>
<BR><FONT SIZE=2>&gt; caller-preferences and the accept/supported/required syntax of 3261.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 3) Followups to a specific instance: We have previously done </FONT>
<BR><FONT SIZE=2>&gt; something </FONT>
<BR><FONT SIZE=2>&gt; (such as case 1 or 2) such that the operation of interest has </FONT>
<BR><FONT SIZE=2>&gt; resolved </FONT>
<BR><FONT SIZE=2>&gt; to a specific contact instance associated with an AoR, and we wish </FONT>
<BR><FONT SIZE=2>&gt; subsequent operations (for example, a transfer) to reach that same </FONT>
<BR><FONT SIZE=2>&gt; specific contact instance, even if that specific contact instance is </FONT>
<BR><FONT SIZE=2>&gt; hidden behind an indirection layer (firewall, anonymizer, etc.). As I </FONT>
<BR><FONT SIZE=2>&gt; understand it, this is the heart of the GRUU concept.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 4) Selection of a specific instance in an initial request: We have a </FONT>
<BR><FONT SIZE=2>&gt; request that we wish to reach one specific instance of a contact </FONT>
<BR><FONT SIZE=2>&gt; associated with an AoR. It has been proposed that we </FONT>
<BR><FONT SIZE=2>&gt; accomplish this by </FONT>
<BR><FONT SIZE=2>&gt; binding some sort of instance tag to the AoR. It has been </FONT>
<BR><FONT SIZE=2>&gt; proposed that </FONT>
<BR><FONT SIZE=2>&gt; we use some sort of URI convention (like the msuri), +based </FONT>
<BR><FONT SIZE=2>&gt; subaddressing, or that we use a header convention for binding the </FONT>
<BR><FONT SIZE=2>&gt; instance tag to the AoR. To my knowledge, we haven't really described </FONT>
<BR><FONT SIZE=2>&gt; the requirement or use case for reaching a specific instance on an </FONT>
<BR><FONT SIZE=2>&gt; initial request, but I can think of several offhand </FONT>
<BR><FONT SIZE=2>&gt; (including diagnostics).</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; It seems evident that there is a huge overlap between the </FONT>
<BR><FONT SIZE=2>&gt; requirements </FONT>
<BR><FONT SIZE=2>&gt; of 3 and 4. Indeed, I think #3 (GRUU) came out of early ideas </FONT>
<BR><FONT SIZE=2>&gt; I had on </FONT>
<BR><FONT SIZE=2>&gt; using a REGISTER-response extension to assign an instance-AoR to a </FONT>
<BR><FONT SIZE=2>&gt; Contact. It would appear that IF each contact for an AoR were </FONT>
<BR><FONT SIZE=2>&gt; to have an </FONT>
<BR><FONT SIZE=2>&gt; associated GRUU, and that knowledge of those GRUU assigments were </FONT>
<BR><FONT SIZE=2>&gt; available to a requestor, that the requester could use the </FONT>
<BR><FONT SIZE=2>&gt; target-specific GRUU to achieve the use cases I currently </FONT>
<BR><FONT SIZE=2>&gt; imagine for #4.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; What am I missing here?</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; --</FONT>
<BR><FONT SIZE=2>&gt; Dean</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

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Date: Tue, 17 Feb 2004 13:16:36 -0600
From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sip] Re: Abstraction:  SIP instance identifiers requirements
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Brian Stucker wrote:
> I think that your list is missing the opportunity to define this instance
> ID in such a way that it can be used to identify a particular endpoint
> for other operations beyond simply routing. The reuse potential seems
> very good if we don't lock the instance identifier down to simply
> routing operations, and instead open it up for other general-purpose
> operations.

So you see a requirement to uniquely identify a specific SIP endpoint 
OUTSIDE of the domain of SIP routing, such that this identification can 
support further applications?

Use cases here might include:

1) billing -- "Alice called Bob from phone ID000001 through gateway 
ID3344555 on Tuesday, March 1, 2011"

2) Debugging -- "This request was responded to by node IDYIYIOHIHIUH, 
which is a Barfia model 9995 phone running Finux 21.05A and the Vocal 
user agent release 89.6"

This begs the question of identity namespace.

> Are you concerned that opening this up to having the instance identifier
> transported (in addition to the registered contact) outside of the
> registered contact is really going to be that complex?

No, I'm concerned that solving the general 
device-identity-representation problem, including such aspects as 
nonrepudiability, may be outside the scope of SIP/SIPPING. At what point 
does this become HIP (Host Identity Payload)?

One way to bound this to a manageable scope is to consider the 
"identity" being represented as simply informational and 
not-to-be-relied upon, except to the extent supported by basic SIP 
security constructs. Like "To:" and "From:" . . .

Another approach is to limit the expression of identity herein to that 
subset of identify which is expressly describable using SIP, which makes 
it analogous to the SIP routing mechanisms underlying the concepts of 
AoR, asserted identity, and verified/vouchsafed identity in the various 
documents that touch on those concepts. In other words, we restrict 
ourselves to the namspace of SIP AoRs.

--
Dean

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Subject: Re: [Sip] draft-ietf-sip-resource-priority-02 - OPTIONS
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At 01:19 PM 2/17/2004 -0500, Paul Kyzivat wrote:
>Henning,
>
>OPTIONS handling: there is good explanation of options processing in a
>UAS, but it is very UAS specific. There isn't any discussion of options
>processing in proxies that support this extension. It is possible to get
>a proxy to respond to an options request (using Max-Forwards). This
>might be used in testing, just a strict mode might.
>
>I think what a proxy should return is pretty much the same as what a UAS 
>returns.

I agree with this point


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


cheers,
James

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


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Jonathan,

I like using a callee-cap feature tag in a registration. OTOH, I can see 
validity in Brian's desire to have it in a header of its own, 
potentially in any message. I don't see those usages as conflicting.

The only potential conflict would be in REGISTER, but that isn't a real 
conflict - a Guid header would describe the entity sending the REGISTER 
message, while the +sip.instance tag on a contact would describe the 
endpoint being registered. Often they would be the same, but not 
necessarily.

Regarding the form of guids, I don't think any single scheme works in 
all cases. My preference is to:

- represent the instance id as a URI

- make it the responsibility of the endpoint to select a URI
   that is unique

- Recommend that a URN be used.

- Define a couple of new URN namespaces that are especially
   suitable for this purpose. (I have in mind urn:mac:xxx for
   mac addresses, and possibly urn:random:xxx for random numbers
   in some large range.)

A couple of other comments:

Section 7.1 says:

    The combination of a UA instance ID and an AOR is referred to as an
    instance ID/AOR pair. There is a one-to-one mapping between such a
    pair and a GRUU; ...

And figure 2 shows the corresponding 1:1 relationship. But it isn't 
necessarily 1:1. It should be 1:0..1 because it is possible to have an 
instance ID/AOR Pair that doesn't have a GRUU. (Instance IDs are 
valuable in their own right.)

This 1:0..1 relationship imposes another error condition on registrars. 
If an attempt is made to register a contact with an instance id in use 
by some other contact of the same AOR, the registrar will have to refuse 
it. This needs to be spelled out somewhere, and a suitable error 
identified. (I don't know if any of the existing responses works for 
this. At the least, a new Reason code is likely to be needed.)

	Paul

Jonathan Rosenberg wrote:
> Folks,
> 
> We have three drafts this round that all propose a way to define an 
> instance identifier for a UA:
> 
> 
> http://www.jdrosen.net/papers/draft-ietf-sip-gruu-01.txt
> http://www.ietf.org/internet-drafts/draft-jennings-sipping-instance-id-00.txt 
> 
> http://www.ietf.org/internet-drafts/draft-stucker-sip-guid-00.txt
> 
> all three are similar. Mine uses a contact header field parameter, and 
> uses the callee capabilities framework to enable caller preferences 
> routing based on instance iDs. Cullen also uses a contact parameter, but 
> not within the callee caps framework. Brian defines a header field.
> 
> I think we generally understand the requirement here, though it would 
> probably be good to obtain use cases to be sure. The design choice is 
> where to include this parameter, and this represents the difference 
> between the three approaches.
> 
> I tend to prefer a Contact URI parameter over a header. My main reason 
> for that is REGISTER requests. If a register request has multiple 
> contacts, which contact does the instance identifier apply to? As a 
> header field, it would be hard to know. If you make it a contact header 
> field parameter, its no problem. Those header field parameters could 
> also then be used in the Contact in INVITE/200 SUB/NOT, etc.
> 
> Then, the question is whether to use a regular Contact header field 
> parameter as Cullen has done, or a callee cap parameter as I have done? 
> The callee cap approach allows for caller preferences to be applied to 
> routing to a specific instance. Now, a good question is whether such a 
> thing is needed or even a good idea, if we have GRUU as the ideal way to 
> route to a UA instance. I can go either way, but welcome thoughts on the 
> topic.
> 
> Thanks,
> Jonathan R.


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I agree in general, but the details are a bit tricky. If I address an 
OPTIONS request to pkyzivat@cisco.com, should the cisco.com proxy 
respond with its Accept-R-P version or should it let it go through and 
have your end system speak for itself? The two lists may obviously not 
be the same.

Paul Kyzivat wrote:

> Henning,
> 
> OPTIONS handling: there is good explanation of options processing in a
> UAS, but it is very UAS specific. There isn't any discussion of options
> processing in proxies that support this extension. It is possible to get
> a proxy to respond to an options request (using Max-Forwards). This
> might be used in testing, just a strict mode might.
> 
> I think what a proxy should return is pretty much the same as what a UAS 
> returns.
> 
>     Paul
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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> - Now that Resource-Priority is optional in all requests, shouldn't
> Accept-Resource-Priority be permitted in 420 responses to all the same
> messages?

Agreed; except for ACK...

> 
> - Shouldn't the R-P be optional in responses to any request where it was
> permitted?

Same quick cut-and-paste error; yes.

> 
> - If A-R-P is mandatory in 417 responses to SUB, NOT, and MSG, shouldn't
> it also be mandatory (rather than optional) in 417 response to INVITE?
> And by logic just above, shouldn't it be mandatory in responses to any
> message where R-P is optional?
> 
> - So I think the table probably ought to look like:
> 
>     Header field             where proxy INV ACK CAN BYE REG OPT PRA
>     ----------------------------------------------------------------
>     Resource-Priority        R     amd    o   o   o   o   o   o   o
>     Resource-Priority        200   -      o   o   o   o   o   o   o
>     Accept-Resource-Priority 200   -      o   o   o   o   o   o   o
>     Accept-Resource-Priority 417   -      m   m   m   m   m   m   m
>     Accept-Resource-Priority 420   -      o   o   o   o   o   o   o
> 
>     Header field             where proxy SUB NOT UPD MSG REF INF PUB
>     ----------------------------------------------------------------
>     Resource-Priority        R     amd    o   o   o   o   o   o   o
>     Resource-Priority        200   -      o   o   o   o   o   o   o
>     Accept-Resource-Priority 200   -      o   o   o   o   o   o   o
>     Accept-Resource-Priority 417   -      m   m   m   m   m   m   m
>     Accept-Resource-Priority 420   -      o   o   o   o   o   o   o

Yup, minus the responses for ACK.

> 
> In section 4.1, why do you single out INVITE, MESSAGE, UPDATE, SUBSCRIBE
> and NOTIFY, but omit ACK and PRACK? If INVITE is important, then ACK and
> PRACK must be important too. This probably isn't an issue in a UAS,
> because it considers those to be part of the INVITE. But it could make a
> difference in a proxy.
> 
>     Paul

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CC: sip@ietf.org
Subject: Re: [Sip] draft-ietf-sip-resource-priority-02 - OPTIONS
References: <402FF85F.4020202@cs.columbia.edu> <40325B42.8000608@cisco.com> <40329FF1.4090006@cs.columbia.edu>
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I thought this was clear in the definition of OPTIONS. If Max-Forwards 
is zero when it gets to the proxy, then the proxy is supposed to 
generate its own response to the OPTIONS. Otherwise it is supposed to 
proxy the request on.

	Paul

Henning Schulzrinne wrote:
> I agree in general, but the details are a bit tricky. If I address an 
> OPTIONS request to pkyzivat@cisco.com, should the cisco.com proxy 
> respond with its Accept-R-P version or should it let it go through and 
> have your end system speak for itself? The two lists may obviously not 
> be the same.
> 
> Paul Kyzivat wrote:
> 
>> Henning,
>>
>> OPTIONS handling: there is good explanation of options processing in a
>> UAS, but it is very UAS specific. There isn't any discussion of options
>> processing in proxies that support this extension. It is possible to get
>> a proxy to respond to an options request (using Max-Forwards). This
>> might be used in testing, just a strict mode might.
>>
>> I think what a proxy should return is pretty much the same as what a 
>> UAS returns.
>>
>>     Paul
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 
> 


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Henning Schulzrinne wrote:
> 
>> - Now that Resource-Priority is optional in all requests, shouldn't
>> Accept-Resource-Priority be permitted in 420 responses to all the same
>> messages?
> 
> 
> Agreed; except for ACK...

Duh! Sorry about that one.

	Paul

	


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Paul Kyzivat wrote:

> Jonathan,
> 
> I like using a callee-cap feature tag in a registration. OTOH, I can see 
> validity in Brian's desire to have it in a header of its own, 
> potentially in any message. I don't see those usages as conflicting.

No, but it represents additional work.

> 
> The only potential conflict would be in REGISTER, but that isn't a real 
> conflict - a Guid header would describe the entity sending the REGISTER 
> message, while the +sip.instance tag on a contact would describe the 
> endpoint being registered. Often they would be the same, but not 
> necessarily.

This makes sense.

> 
> Regarding the form of guids, I don't think any single scheme works in 
> all cases. My preference is to:
> 
> - represent the instance id as a URI
> 
> - make it the responsibility of the endpoint to select a URI
>   that is unique
> 
> - Recommend that a URN be used.
> 
> - Define a couple of new URN namespaces that are especially
>   suitable for this purpose. (I have in mind urn:mac:xxx for
>   mac addresses, and possibly urn:random:xxx for random numbers
>   in some large range.)

This is a well trod area, and I think it would behoove us to look at 
work in other groups. Dean has mentioned HIP as one obvious candidate; 
generally speaking this exact issue - of endpoint identifiers that work 
better than IP address, is the subject of much other work at this 
moment, including MAST. I'm not familiar enough with it to know if its 
relevant... its on my reading list for this ietf.


> 
> A couple of other comments:
> 
> Section 7.1 says:
> 
>    The combination of a UA instance ID and an AOR is referred to as an
>    instance ID/AOR pair. There is a one-to-one mapping between such a
>    pair and a GRUU; ...
> 
> And figure 2 shows the corresponding 1:1 relationship. But it isn't 
> necessarily 1:1. It should be 1:0..1 because it is possible to have an 
> instance ID/AOR Pair that doesn't have a GRUU. (Instance IDs are 
> valuable in their own right.)

OK, I'll update the diagram.

> 
> This 1:0..1 relationship imposes another error condition on registrars. 
> If an attempt is made to register a contact with an instance id in use 
> by some other contact of the same AOR, the registrar will have to refuse 
> it. 

Well, thats one option.

This needs to be spelled out somewhere, and a suitable error
> identified. (I don't know if any of the existing responses works for 
> this. At the least, a new Reason code is likely to be needed.)

Here is the use case for this. The use case is when a UA crashes, 
recovers, obtains a new IP, and re-registers. In this case, several 
things could possibly happen:

1. The registrar accepts the second registration, as per normal 
registration procedures. An INVITE to the gruu bound to that id/aor pair 
gets translated into BOTH contacts, causing the request to fork. There 
is never any response from the old IP. Within one refresh interval, the 
old contact is removed and all is well.

2. The registrar accepts the second registration, and automatically 
deletes the old one. I believe Brian proposed this behavior.

3. The registrar returns an error. The client fetches the current list 
of contacts, finds the old one listed, and unregisters it.



Approach 1 is problematic, as it will introduce herfp; any non-2xx from 
the new contact will wait for the 2nd branch to timeout. Its not clear 
to me whether 2 or 3 is more right. It depends a lot on who owns the 
policy for how to resolve the conflict - is it the server or the client?

-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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inline.

Dean Willis wrote:

> 
> Based on the discussion of "routing" based on instance identifiers, I
>  think we may be getting involved in triggerwork that may become more
>  complex than is actually required, because we're all reasoning from 
> secondary positions. So I'd like to take a step back and consider how
> we got here, and what we think we're trying to accomplish.
> 
> I see four high-level cases around request routing that illustrate 
> different aspects of this discussion.
> 
> 1) General case: We wish to address a request to an AOR, such that 
> any contact at that AOR might respond. This is basic 2543-type 
> routing behavior.

Well, I prefer 3261-style, but yes.

> 
> 2) Capability case: We wish to address a request to an AOR, such that
>  any contact capable of satisfying the request in a way that we like 
> might respond. This behavior is brought out in callee-caps and 
> caller-preferences and the accept/supported/required syntax of 3261.

Right.

> 
> 3) Followups to a specific instance: We have previously done 
> something (such as case 1 or 2) such that the operation of interest 
> has resolved to a specific contact instance associated with an AoR, 
> and we wish subsequent operations (for example, a transfer) to reach 
> that same specific contact instance, even if that specific contact 
> instance is hidden behind an indirection layer (firewall, anonymizer,
>  etc.). As I understand it, this is the heart of the GRUU concept.

Yes.

> 
> 4) Selection of a specific instance in an initial request: We have a 
> request that we wish to reach one specific instance of a contact 
> associated with an AoR.

My proposal of using a callee capabilities parameter for instance ID
would accomplish that. However, I am not sure that its really useful, to
be honest. How would the caller know the instance ID ahead of time? Any
kind of mechanism that can transfer the instance ID could transfer the
GRUU instead, and a GRUU will be much, much, much better.


It has been proposed that we accomplish this by
> binding some sort of instance tag to the AoR. It has been proposed 
> that we use some sort of URI convention (like the msuri), +based 
> subaddressing,

I wasnt proposing that. The +sip.instance is the convention used by
callee caps for identifying Contact header parameters that are subject
to the specification.


> It seems evident that there is a huge overlap between the 
> requirements of 3 and 4.

Yes.

> Indeed, I think #3 (GRUU) came out of early ideas I had on using a 
> REGISTER-response extension to assign an instance-AoR to a Contact. 
> It would appear that IF each contact for an AoR were to have an 
> associated GRUU, and that knowledge of those GRUU assigments were 
> available to a requestor, that the requester could use the 
> target-specific GRUU to achieve the use cases I currently imagine for
>  #4.

Right, this is exactly what I am saying above as well.

Dean later writes, in response to Brian:
> So you see a requirement to uniquely identify a specific SIP endpoint
>  OUTSIDE of the domain of SIP routing, such that this identification 
> can support further applications?
> 
> Use cases here might include:
> 
> 1) billing -- "Alice called Bob from phone ID000001 through gateway 
> ID3344555 on Tuesday, March 1, 2011"

I do not understand the need for a globally unique instance ID to
support billing, and would be very scared of fraud issues.

> 
> 2) Debugging -- "This request was responded to by node IDYIYIOHIHIUH,
>  which is a Barfia model 9995 phone running Finux 21.05A and the
> Vocal user agent release 89.6"

I'm not sure I understand the specific value gleaned from the instance
ID in this case.

> Another approach is to limit the expression of identity herein to
> that subset of identify which is expressly describable using SIP,
> which makes it analogous to the SIP routing mechanisms underlying the
> concepts of AoR, asserted identity, and verified/vouchsafed identity
> in the various documents that touch on those concepts. In other
> words, we restrict ourselves to the namspace of SIP AoRs.


Well said.

-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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From: "Brian Stucker" <bstucker@nortelnetworks.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Paul Kyzivat
	 <pkyzivat@cisco.com>
Cc: sip@ietf.org
Subject: RE: [Sip] SIP instance identifiers
Date: Wed, 18 Feb 2004 01:26:52 -0600
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Inline...

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Tuesday, February 17, 2004 10:32 PM
> To: Paul Kyzivat
> Cc: sip@ietf.org
> Subject: Re: [Sip] SIP instance identifiers
> 
> 
> 
> 
> Paul Kyzivat wrote:
> 
> > Jonathan,
> > 
> > I like using a callee-cap feature tag in a registration. 
> OTOH, I can see 
> > validity in Brian's desire to have it in a header of its own, 
> > potentially in any message. I don't see those usages as conflicting.
> 
> No, but it represents additional work.

A very small amount of additional work, and I believe the majority of it
has already been handled in my draft (specifically defining the header). 
We've agreed that there's a need for a mechanism like this
for PUBLISH. There's an obvious hole in generating meaningful instance 
identifiers for PIDF documents and the such. Other state aggregation
mechanisms
in the future are easily envisionable that would need to know which unique 
client sent a request, where it would be useful to have that information
transported in a request in a handy mechanism.

Why not take care of both problems in one shot, define the (minimal)
interactions
and then follow things up in the various drafts like PUBLISH and any others
that
see a need for this where they define the appropriate policy for using the
instance identifier they've been given ala RFC-3265. Define the basic
mechanism
(in that case SUBSCRIBE/NOTIFY) and let others define the usages.

> 
> > 
> > The only potential conflict would be in REGISTER, but that 
> isn't a real 
> > conflict - a Guid header would describe the entity sending 
> the REGISTER 
> > message, while the +sip.instance tag on a contact would 
> describe the 
> > endpoint being registered. Often they would be the same, but not 
> > necessarily.
> 
> This makes sense.

Agreed. The two mechanisms can easily live together much like an 
expires header in the registration does along side one in the contact.

> 
> > 
> > Regarding the form of guids, I don't think any single 
> scheme works in 
> > all cases. My preference is to:
> > 
> > - represent the instance id as a URI
> > 
> > - make it the responsibility of the endpoint to select a URI
> >   that is unique
> > 
> > - Recommend that a URN be used.
> > 
> > - Define a couple of new URN namespaces that are especially
> >   suitable for this purpose. (I have in mind urn:mac:xxx for
> >   mac addresses, and possibly urn:random:xxx for random numbers
> >   in some large range.)
> 
> This is a well trod area, and I think it would behoove us to look at 
> work in other groups. Dean has mentioned HIP as one obvious 
> candidate; 
> generally speaking this exact issue - of endpoint identifiers 
> that work 
> better than IP address, is the subject of much other work at this 
> moment, including MAST. I'm not familiar enough with it to 
> know if its 
> relevant... its on my reading list for this ietf.
> 
> 

This represents additional work as well. =)

I think we're limiting ourselves to just MACs and random numbers. Having
looked around a bit, I've seen material (I want to say it's from WebDAV
but I could be remembering incorrectly) that the MAC address isn't always
a good thing to pass around because you can figure out that the UA is 
running on a particular piece of hardware without otherwise having access
to their MAC (because you're on the other side of the world many routers
away). That can be used to determine a method of attack because you can 
inferr what operating systems, etc might be running if you know what kind
of hardware that box is plugged into the network with.

That said, if you want to use a MAC, fine. Buyer beware. If you want to use
a random number, fine. If you want to use the GPS coordinates of the box to
some ridiculous level of precision (and that's good enough for your
application)
fine. 

I don't see why we necessarily need to put any restrictions on this
instance ID to make it a URN or anything else when it really need not be any
more complex than a VIA branch. I also don't see a reason why we should
expect
other nodes in the network to be able to use the instance identifier for
anything
other than a token (ie. if it's a mac address, and two UA instances do
something
because it's a MAC address, that's OK, but other's may not take that tact).
There's
no guarantee others are going to be able to take whatever action might be
denoted
by the contents of the instance identifier itself, so it becomes entirely 
proprietary at that point. 

> > 
> > A couple of other comments:
> > 
> > Section 7.1 says:
> > 
> >    The combination of a UA instance ID and an AOR is 
> referred to as an
> >    instance ID/AOR pair. There is a one-to-one mapping 
> between such a
> >    pair and a GRUU; ...
> > 
> > And figure 2 shows the corresponding 1:1 relationship. But it isn't 
> > necessarily 1:1. It should be 1:0..1 because it is possible 
> to have an 
> > instance ID/AOR Pair that doesn't have a GRUU. (Instance IDs are 
> > valuable in their own right.)
> 
> OK, I'll update the diagram.
> 
> > 
> > This 1:0..1 relationship imposes another error condition on 
> registrars. 
> > If an attempt is made to register a contact with an 
> instance id in use 
> > by some other contact of the same AOR, the registrar will 
> have to refuse 
> > it. 
> 
> Well, thats one option.

This is why I put some boundaries on the length of the GUID and some
basic guidelines, to make this very unlikely to occur for reasonably 
well behaved implementations.

> 
> This needs to be spelled out somewhere, and a suitable error
> > identified. (I don't know if any of the existing responses 
> works for 
> > this. At the least, a new Reason code is likely to be needed.)
> 
> Here is the use case for this. The use case is when a UA crashes, 
> recovers, obtains a new IP, and re-registers. In this case, several 
> things could possibly happen:
> 
> 1. The registrar accepts the second registration, as per normal 
> registration procedures. An INVITE to the gruu bound to that 
> id/aor pair 
> gets translated into BOTH contacts, causing the request to 
> fork. There 
> is never any response from the old IP. Within one refresh 
> interval, the 
> old contact is removed and all is well.
> 
> 2. The registrar accepts the second registration, and automatically 
> deletes the old one. I believe Brian proposed this behavior.
> 
> 3. The registrar returns an error. The client fetches the 
> current list 
> of contacts, finds the old one listed, and unregisters it.
> 
> 
> 
> Approach 1 is problematic, as it will introduce herfp; any 
> non-2xx from 
> the new contact will wait for the 2nd branch to timeout. Its 
> not clear 
> to me whether 2 or 3 is more right. It depends a lot on who owns the 
> policy for how to resolve the conflict - is it the server or 
> the client?

Agree on 1 being bad.

IMHO it's the server that owns the policy. If the client is too dumb to 
remember their previous contact address, then it's probably too stateless
to make policy decisions about which contact should be removed too
(otherwise
it would have simply deregistered that contact in the first place).

> 
> -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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<P><FONT SIZE=2>Inline...</FONT>
</P>

<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: Jonathan Rosenberg [<A HREF="mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</A>]</FONT>
<BR><FONT SIZE=2>&gt; Sent: Tuesday, February 17, 2004 10:32 PM</FONT>
<BR><FONT SIZE=2>&gt; To: Paul Kyzivat</FONT>
<BR><FONT SIZE=2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=2>&gt; Subject: Re: [Sip] SIP instance identifiers</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Paul Kyzivat wrote:</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; Jonathan,</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; I like using a callee-cap feature tag in a registration. </FONT>
<BR><FONT SIZE=2>&gt; OTOH, I can see </FONT>
<BR><FONT SIZE=2>&gt; &gt; validity in Brian's desire to have it in a header of its own, </FONT>
<BR><FONT SIZE=2>&gt; &gt; potentially in any message. I don't see those usages as conflicting.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; No, but it represents additional work.</FONT>
</P>

<P><FONT SIZE=2>A very small amount of additional work, and I believe the majority of it</FONT>
<BR><FONT SIZE=2>has already been handled in my draft (specifically defining the header). </FONT>
<BR><FONT SIZE=2>We've agreed that there's a need for a mechanism like this</FONT>
<BR><FONT SIZE=2>for PUBLISH. There's an obvious hole in generating meaningful instance </FONT>
<BR><FONT SIZE=2>identifiers for PIDF documents and the such. Other state aggregation mechanisms</FONT>
<BR><FONT SIZE=2>in the future are easily envisionable that would need to know which unique </FONT>
<BR><FONT SIZE=2>client sent a request, where it would be useful to have that information</FONT>
<BR><FONT SIZE=2>transported in a request in a handy mechanism.</FONT>
</P>

<P><FONT SIZE=2>Why not take care of both problems in one shot, define the (minimal) interactions</FONT>
<BR><FONT SIZE=2>and then follow things up in the various drafts like PUBLISH and any others that</FONT>
<BR><FONT SIZE=2>see a need for this where they define the appropriate policy for using the</FONT>
<BR><FONT SIZE=2>instance identifier they've been given ala RFC-3265. Define the basic mechanism</FONT>
<BR><FONT SIZE=2>(in that case SUBSCRIBE/NOTIFY) and let others define the usages.</FONT>
</P>

<P><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; The only potential conflict would be in REGISTER, but that </FONT>
<BR><FONT SIZE=2>&gt; isn't a real </FONT>
<BR><FONT SIZE=2>&gt; &gt; conflict - a Guid header would describe the entity sending </FONT>
<BR><FONT SIZE=2>&gt; the REGISTER </FONT>
<BR><FONT SIZE=2>&gt; &gt; message, while the +sip.instance tag on a contact would </FONT>
<BR><FONT SIZE=2>&gt; describe the </FONT>
<BR><FONT SIZE=2>&gt; &gt; endpoint being registered. Often they would be the same, but not </FONT>
<BR><FONT SIZE=2>&gt; &gt; necessarily.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; This makes sense.</FONT>
</P>

<P><FONT SIZE=2>Agreed. The two mechanisms can easily live together much like an </FONT>
<BR><FONT SIZE=2>expires header in the registration does along side one in the contact.</FONT>
</P>

<P><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; Regarding the form of guids, I don't think any single </FONT>
<BR><FONT SIZE=2>&gt; scheme works in </FONT>
<BR><FONT SIZE=2>&gt; &gt; all cases. My preference is to:</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; - represent the instance id as a URI</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; - make it the responsibility of the endpoint to select a URI</FONT>
<BR><FONT SIZE=2>&gt; &gt;&nbsp;&nbsp; that is unique</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; - Recommend that a URN be used.</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; - Define a couple of new URN namespaces that are especially</FONT>
<BR><FONT SIZE=2>&gt; &gt;&nbsp;&nbsp; suitable for this purpose. (I have in mind urn:mac:xxx for</FONT>
<BR><FONT SIZE=2>&gt; &gt;&nbsp;&nbsp; mac addresses, and possibly urn:random:xxx for random numbers</FONT>
<BR><FONT SIZE=2>&gt; &gt;&nbsp;&nbsp; in some large range.)</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; This is a well trod area, and I think it would behoove us to look at </FONT>
<BR><FONT SIZE=2>&gt; work in other groups. Dean has mentioned HIP as one obvious </FONT>
<BR><FONT SIZE=2>&gt; candidate; </FONT>
<BR><FONT SIZE=2>&gt; generally speaking this exact issue - of endpoint identifiers </FONT>
<BR><FONT SIZE=2>&gt; that work </FONT>
<BR><FONT SIZE=2>&gt; better than IP address, is the subject of much other work at this </FONT>
<BR><FONT SIZE=2>&gt; moment, including MAST. I'm not familiar enough with it to </FONT>
<BR><FONT SIZE=2>&gt; know if its </FONT>
<BR><FONT SIZE=2>&gt; relevant... its on my reading list for this ietf.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

<P><FONT SIZE=2>This represents additional work as well. =)</FONT>
</P>

<P><FONT SIZE=2>I think we're limiting ourselves to just MACs and random numbers. Having</FONT>
<BR><FONT SIZE=2>looked around a bit, I've seen material (I want to say it's from WebDAV</FONT>
<BR><FONT SIZE=2>but I could be remembering incorrectly) that the MAC address isn't always</FONT>
<BR><FONT SIZE=2>a good thing to pass around because you can figure out that the UA is </FONT>
<BR><FONT SIZE=2>running on a particular piece of hardware without otherwise having access</FONT>
<BR><FONT SIZE=2>to their MAC (because you're on the other side of the world many routers</FONT>
<BR><FONT SIZE=2>away). That can be used to determine a method of attack because you can </FONT>
<BR><FONT SIZE=2>inferr what operating systems, etc might be running if you know what kind</FONT>
<BR><FONT SIZE=2>of hardware that box is plugged into the network with.</FONT>
</P>

<P><FONT SIZE=2>That said, if you want to use a MAC, fine. Buyer beware. If you want to use</FONT>
<BR><FONT SIZE=2>a random number, fine. If you want to use the GPS coordinates of the box to</FONT>
<BR><FONT SIZE=2>some ridiculous level of precision (and that's good enough for your application)</FONT>
<BR><FONT SIZE=2>fine. </FONT>
</P>

<P><FONT SIZE=2>I don't see why we necessarily need to put any restrictions on this</FONT>
<BR><FONT SIZE=2>instance ID to make it a URN or anything else when it really need not be any</FONT>
<BR><FONT SIZE=2>more complex than a VIA branch. I also don't see a reason why we should expect</FONT>
<BR><FONT SIZE=2>other nodes in the network to be able to use the instance identifier for anything</FONT>
<BR><FONT SIZE=2>other than a token (ie. if it's a mac address, and two UA instances do something</FONT>
<BR><FONT SIZE=2>because it's a MAC address, that's OK, but other's may not take that tact). There's</FONT>
<BR><FONT SIZE=2>no guarantee others are going to be able to take whatever action might be denoted</FONT>
<BR><FONT SIZE=2>by the contents of the instance identifier itself, so it becomes entirely </FONT>
<BR><FONT SIZE=2>proprietary at that point. </FONT>
</P>

<P><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; A couple of other comments:</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; Section 7.1 says:</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt;&nbsp;&nbsp;&nbsp; The combination of a UA instance ID and an AOR is </FONT>
<BR><FONT SIZE=2>&gt; referred to as an</FONT>
<BR><FONT SIZE=2>&gt; &gt;&nbsp;&nbsp;&nbsp; instance ID/AOR pair. There is a one-to-one mapping </FONT>
<BR><FONT SIZE=2>&gt; between such a</FONT>
<BR><FONT SIZE=2>&gt; &gt;&nbsp;&nbsp;&nbsp; pair and a GRUU; ...</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; And figure 2 shows the corresponding 1:1 relationship. But it isn't </FONT>
<BR><FONT SIZE=2>&gt; &gt; necessarily 1:1. It should be 1:0..1 because it is possible </FONT>
<BR><FONT SIZE=2>&gt; to have an </FONT>
<BR><FONT SIZE=2>&gt; &gt; instance ID/AOR Pair that doesn't have a GRUU. (Instance IDs are </FONT>
<BR><FONT SIZE=2>&gt; &gt; valuable in their own right.)</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; OK, I'll update the diagram.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; This 1:0..1 relationship imposes another error condition on </FONT>
<BR><FONT SIZE=2>&gt; registrars. </FONT>
<BR><FONT SIZE=2>&gt; &gt; If an attempt is made to register a contact with an </FONT>
<BR><FONT SIZE=2>&gt; instance id in use </FONT>
<BR><FONT SIZE=2>&gt; &gt; by some other contact of the same AOR, the registrar will </FONT>
<BR><FONT SIZE=2>&gt; have to refuse </FONT>
<BR><FONT SIZE=2>&gt; &gt; it. </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Well, thats one option.</FONT>
</P>

<P><FONT SIZE=2>This is why I put some boundaries on the length of the GUID and some</FONT>
<BR><FONT SIZE=2>basic guidelines, to make this very unlikely to occur for reasonably </FONT>
<BR><FONT SIZE=2>well behaved implementations.</FONT>
</P>

<P><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; This needs to be spelled out somewhere, and a suitable error</FONT>
<BR><FONT SIZE=2>&gt; &gt; identified. (I don't know if any of the existing responses </FONT>
<BR><FONT SIZE=2>&gt; works for </FONT>
<BR><FONT SIZE=2>&gt; &gt; this. At the least, a new Reason code is likely to be needed.)</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Here is the use case for this. The use case is when a UA crashes, </FONT>
<BR><FONT SIZE=2>&gt; recovers, obtains a new IP, and re-registers. In this case, several </FONT>
<BR><FONT SIZE=2>&gt; things could possibly happen:</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 1. The registrar accepts the second registration, as per normal </FONT>
<BR><FONT SIZE=2>&gt; registration procedures. An INVITE to the gruu bound to that </FONT>
<BR><FONT SIZE=2>&gt; id/aor pair </FONT>
<BR><FONT SIZE=2>&gt; gets translated into BOTH contacts, causing the request to </FONT>
<BR><FONT SIZE=2>&gt; fork. There </FONT>
<BR><FONT SIZE=2>&gt; is never any response from the old IP. Within one refresh </FONT>
<BR><FONT SIZE=2>&gt; interval, the </FONT>
<BR><FONT SIZE=2>&gt; old contact is removed and all is well.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 2. The registrar accepts the second registration, and automatically </FONT>
<BR><FONT SIZE=2>&gt; deletes the old one. I believe Brian proposed this behavior.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 3. The registrar returns an error. The client fetches the </FONT>
<BR><FONT SIZE=2>&gt; current list </FONT>
<BR><FONT SIZE=2>&gt; of contacts, finds the old one listed, and unregisters it.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Approach 1 is problematic, as it will introduce herfp; any </FONT>
<BR><FONT SIZE=2>&gt; non-2xx from </FONT>
<BR><FONT SIZE=2>&gt; the new contact will wait for the 2nd branch to timeout. Its </FONT>
<BR><FONT SIZE=2>&gt; not clear </FONT>
<BR><FONT SIZE=2>&gt; to me whether 2 or 3 is more right. It depends a lot on who owns the </FONT>
<BR><FONT SIZE=2>&gt; policy for how to resolve the conflict - is it the server or </FONT>
<BR><FONT SIZE=2>&gt; the client?</FONT>
</P>

<P><FONT SIZE=2>Agree on 1 being bad.</FONT>
</P>

<P><FONT SIZE=2>IMHO it's the server that owns the policy. If the client is too dumb to </FONT>
<BR><FONT SIZE=2>remember their previous contact address, then it's probably too stateless</FONT>
<BR><FONT SIZE=2>to make policy decisions about which contact should be removed too (otherwise</FONT>
<BR><FONT SIZE=2>it would have simply deregistered that contact in the first place).</FONT>
</P>

<P><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; -Jonathan R.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; -- </FONT>
<BR><FONT SIZE=2>&gt; Jonathan D. Rosenberg, Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 600 Lanidex Plaza</FONT>
<BR><FONT SIZE=2>&gt; Chief Technology Officer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Parsippany, NJ 07054-2711</FONT>
<BR><FONT SIZE=2>&gt; dynamicsoft</FONT>
<BR><FONT SIZE=2>&gt; jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FAX:&nbsp;&nbsp; (973) 952-5050</FONT>
<BR><FONT SIZE=2>&gt; <A HREF="http://www.jdrosen.net" TARGET="_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) 952-5000</FONT>
<BR><FONT SIZE=2>&gt; <A HREF="http://www.dynamicsoft.com" TARGET="_blank">http://www.dynamicsoft.com</A></FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; _______________________________________________</FONT>
<BR><FONT SIZE=2>&gt; Sip mailing list&nbsp; <A HREF="https://www1.ietf.org/mailman/listinfo/sip" TARGET="_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=2>&gt; This list is for NEW development of the core SIP Protocol</FONT>
<BR><FONT SIZE=2>&gt; Use sip-implementors@cs.columbia.edu for questions on current sip</FONT>
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From: "Brian Stucker" <bstucker@nortelnetworks.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Dean Willis
	 <dean.willis@softarmor.com>
Cc: Cullen Jennings <fluffy@cisco.com>, sip@ietf.org
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Subject: [Sip] RE: Abstraction:  SIP instance identifiers requirements
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> -----Original Message-----

<< SNIP >>

> 
> 
> Dean later writes, in response to Brian:
> > So you see a requirement to uniquely identify a specific 
> SIP endpoint
> >  OUTSIDE of the domain of SIP routing, such that this 
> identification 
> > can support further applications?
> > 
> > Use cases here might include:
> > 
> > 1) billing -- "Alice called Bob from phone ID000001 through gateway 
> > ID3344555 on Tuesday, March 1, 2011"
> 
> I do not understand the need for a globally unique instance ID to
> support billing, and would be very scared of fraud issues.

If the particular network implementation only hands out the ID to trusted
nodes, this is no less safe than our current authentication mechanisms IMHO.
I don't forsee using the ID as a "bill to" number, but it could be used to
help correlate billing records as a call goes through the system, in which 
case it does not matter what that value is, just that it's consistient from
hop to hop.

If you're scared of fraud issues, think about what happens when you send an
INVITE into a system with a Referred-By of some unsuspecting third-party
user
of that system. =) Much worse problem that exists today. You probably won't 
get a free call, but you might make some interesting international calls
before anyone figures out what's going on.

> 
> > 
> > 2) Debugging -- "This request was responded to by node 
> IDYIYIOHIHIUH,
> >  which is a Barfia model 9995 phone running Finux 21.05A and the
> > Vocal user agent release 89.6"
> 
> I'm not sure I understand the specific value gleaned from the instance
> ID in this case.

It survives BBUAs for one. Probably not the best use case, but may be
interesting for future thought. System admins will probably love to have
a consistient identifier throughout the system identifying what a 
particular endpoint did throughout a request or requests.

The specific use cases I had in mind were to recover from client crashes
where it's undesirable, and network policy to limit resource usage. For 
instance, duplicate subscriptions to the same resource for the same
event package from the same client instance. Sure, sometimes it might 
make sense (can't think of one, but let's assume that there is a case). 
That still represents network resources (ie. resources associated with
storing that subscription and handling it on state changes for that
event) and the network might not want to allow duplicates by its own
policy (tough crapola client, you only get so much). An instance ID
makes this very simple to deal with. Otherwise you're at the mercy of
the client, or the server has to guess which subscriptions it has no more
resources to handle and should eliminate.

Same goes for registrations which was pointed out on another thread. If
a client is stateless enough that it can't remember an old contact after
crashing, it may still be able to come up with a GUID that the server can
then use to figure out to eliminate the now defunct elder contact in
favor of the newer one.

> 
> > Another approach is to limit the expression of identity herein to
> > that subset of identify which is expressly describable using SIP,
> > which makes it analogous to the SIP routing mechanisms 
> underlying the
> > concepts of AoR, asserted identity, and verified/vouchsafed identity
> > in the various documents that touch on those concepts. In other
> > words, we restrict ourselves to the namspace of SIP AoRs.
> 

If we take that approach then how do we solve the problems I've stated
above that exist today? How do we solve problems with generating meaningful
identifiers for clients in the reginfo+xml message body, and cpim-pidf 
message bodies? How do I determine all PUBLISHes are from a particular 
user agent when it's contact information and other aspects of the PUBLISH 
may vary over long periods of time? 

We're going to have to solve these other issues. We're going to need to 
know which client is generating these requests. It's a similar problem
space,
and with or without GRUUs, these problems are still going to exist and will
eventually need to be tackled.

> 
> Well said.
> 
> -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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<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
</P>

<P><FONT SIZE=2>&lt;&lt; SNIP &gt;&gt;</FONT>
</P>

<P><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Dean later writes, in response to Brian:</FONT>
<BR><FONT SIZE=2>&gt; &gt; So you see a requirement to uniquely identify a specific </FONT>
<BR><FONT SIZE=2>&gt; SIP endpoint</FONT>
<BR><FONT SIZE=2>&gt; &gt;&nbsp; OUTSIDE of the domain of SIP routing, such that this </FONT>
<BR><FONT SIZE=2>&gt; identification </FONT>
<BR><FONT SIZE=2>&gt; &gt; can support further applications?</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; Use cases here might include:</FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; 1) billing -- &quot;Alice called Bob from phone ID000001 through gateway </FONT>
<BR><FONT SIZE=2>&gt; &gt; ID3344555 on Tuesday, March 1, 2011&quot;</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I do not understand the need for a globally unique instance ID to</FONT>
<BR><FONT SIZE=2>&gt; support billing, and would be very scared of fraud issues.</FONT>
</P>

<P><FONT SIZE=2>If the particular network implementation only hands out the ID to trusted</FONT>
<BR><FONT SIZE=2>nodes, this is no less safe than our current authentication mechanisms IMHO.</FONT>
<BR><FONT SIZE=2>I don't forsee using the ID as a &quot;bill to&quot; number, but it could be used to</FONT>
<BR><FONT SIZE=2>help correlate billing records as a call goes through the system, in which </FONT>
<BR><FONT SIZE=2>case it does not matter what that value is, just that it's consistient from</FONT>
<BR><FONT SIZE=2>hop to hop.</FONT>
</P>

<P><FONT SIZE=2>If you're scared of fraud issues, think about what happens when you send an</FONT>
<BR><FONT SIZE=2>INVITE into a system with a Referred-By of some unsuspecting third-party user</FONT>
<BR><FONT SIZE=2>of that system. =) Much worse problem that exists today. You probably won't </FONT>
<BR><FONT SIZE=2>get a free call, but you might make some interesting international calls</FONT>
<BR><FONT SIZE=2>before anyone figures out what's going on.</FONT>
</P>

<P><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; 2) Debugging -- &quot;This request was responded to by node </FONT>
<BR><FONT SIZE=2>&gt; IDYIYIOHIHIUH,</FONT>
<BR><FONT SIZE=2>&gt; &gt;&nbsp; which is a Barfia model 9995 phone running Finux 21.05A and the</FONT>
<BR><FONT SIZE=2>&gt; &gt; Vocal user agent release 89.6&quot;</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I'm not sure I understand the specific value gleaned from the instance</FONT>
<BR><FONT SIZE=2>&gt; ID in this case.</FONT>
</P>

<P><FONT SIZE=2>It survives BBUAs for one. Probably not the best use case, but may be</FONT>
<BR><FONT SIZE=2>interesting for future thought. System admins will probably love to have</FONT>
<BR><FONT SIZE=2>a consistient identifier throughout the system identifying what a </FONT>
<BR><FONT SIZE=2>particular endpoint did throughout a request or requests.</FONT>
</P>

<P><FONT SIZE=2>The specific use cases I had in mind were to recover from client crashes</FONT>
<BR><FONT SIZE=2>where it's undesirable, and network policy to limit resource usage. For </FONT>
<BR><FONT SIZE=2>instance, duplicate subscriptions to the same resource for the same</FONT>
<BR><FONT SIZE=2>event package from the same client instance. Sure, sometimes it might </FONT>
<BR><FONT SIZE=2>make sense (can't think of one, but let's assume that there is a case). </FONT>
<BR><FONT SIZE=2>That still represents network resources (ie. resources associated with</FONT>
<BR><FONT SIZE=2>storing that subscription and handling it on state changes for that</FONT>
<BR><FONT SIZE=2>event) and the network might not want to allow duplicates by its own</FONT>
<BR><FONT SIZE=2>policy (tough crapola client, you only get so much). An instance ID</FONT>
<BR><FONT SIZE=2>makes this very simple to deal with. Otherwise you're at the mercy of</FONT>
<BR><FONT SIZE=2>the client, or the server has to guess which subscriptions it has no more</FONT>
<BR><FONT SIZE=2>resources to handle and should eliminate.</FONT>
</P>

<P><FONT SIZE=2>Same goes for registrations which was pointed out on another thread. If</FONT>
<BR><FONT SIZE=2>a client is stateless enough that it can't remember an old contact after</FONT>
<BR><FONT SIZE=2>crashing, it may still be able to come up with a GUID that the server can</FONT>
<BR><FONT SIZE=2>then use to figure out to eliminate the now defunct elder contact in</FONT>
<BR><FONT SIZE=2>favor of the newer one.</FONT>
</P>

<P><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; Another approach is to limit the expression of identity herein to</FONT>
<BR><FONT SIZE=2>&gt; &gt; that subset of identify which is expressly describable using SIP,</FONT>
<BR><FONT SIZE=2>&gt; &gt; which makes it analogous to the SIP routing mechanisms </FONT>
<BR><FONT SIZE=2>&gt; underlying the</FONT>
<BR><FONT SIZE=2>&gt; &gt; concepts of AoR, asserted identity, and verified/vouchsafed identity</FONT>
<BR><FONT SIZE=2>&gt; &gt; in the various documents that touch on those concepts. In other</FONT>
<BR><FONT SIZE=2>&gt; &gt; words, we restrict ourselves to the namspace of SIP AoRs.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

<P><FONT SIZE=2>If we take that approach then how do we solve the problems I've stated</FONT>
<BR><FONT SIZE=2>above that exist today? How do we solve problems with generating meaningful</FONT>
<BR><FONT SIZE=2>identifiers for clients in the reginfo+xml message body, and cpim-pidf </FONT>
<BR><FONT SIZE=2>message bodies? How do I determine all PUBLISHes are from a particular </FONT>
<BR><FONT SIZE=2>user agent when it's contact information and other aspects of the PUBLISH </FONT>
<BR><FONT SIZE=2>may vary over long periods of time? </FONT>
</P>

<P><FONT SIZE=2>We're going to have to solve these other issues. We're going to need to </FONT>
<BR><FONT SIZE=2>know which client is generating these requests. It's a similar problem space,</FONT>
<BR><FONT SIZE=2>and with or without GRUUs, these problems are still going to exist and will</FONT>
<BR><FONT SIZE=2>eventually need to be tackled.</FONT>
</P>

<P><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Well said.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; -Jonathan R.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; -- </FONT>
<BR><FONT SIZE=2>&gt; Jonathan D. Rosenberg, Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 600 Lanidex Plaza</FONT>
<BR><FONT SIZE=2>&gt; Chief Technology Officer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Parsippany, NJ 07054-2711</FONT>
<BR><FONT SIZE=2>&gt; dynamicsoft</FONT>
<BR><FONT SIZE=2>&gt; jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FAX:&nbsp;&nbsp; (973) 952-5050</FONT>
<BR><FONT SIZE=2>&gt; <A HREF="http://www.jdrosen.net" TARGET="_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) 952-5000</FONT>
<BR><FONT SIZE=2>&gt; <A HREF="http://www.dynamicsoft.com" TARGET="_blank">http://www.dynamicsoft.com</A></FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

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Subject: RE: [Sip] SIP instance identifiers
Date: Wed, 18 Feb 2004 09:15:24 -0000
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Thread-Topic: [Sip] SIP instance identifiers
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From: "Chris Boulton" <CBoulton@ubiquity.net>
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>
>This needs to be spelled out somewhere, and a suitable error
>> identified. (I don't know if any of the existing responses works for
>> this. At the least, a new Reason code is likely to be needed.)
>
>Here is the use case for this. The use case is when a UA crashes,
>recovers, obtains a new IP, and re-registers. In this case, several
>things could possibly happen:
>
>1. The registrar accepts the second registration, as per normal
>registration procedures. An INVITE to the gruu bound to that id/aor
pair
>gets translated into BOTH contacts, causing the request to fork. There
>is never any response from the old IP. Within one refresh interval, the
>old contact is removed and all is well.
>
>2. The registrar accepts the second registration, and automatically
>deletes the old one. I believe Brian proposed this behavior.
>
>3. The registrar returns an error. The client fetches the current list
>of contacts, finds the old one listed, and unregisters it.
>
>
>
>Approach 1 is problematic, as it will introduce herfp; any non-2xx from
>the new contact will wait for the 2nd branch to timeout. Its not clear
>to me whether 2 or 3 is more right. It depends a lot on who owns the
>policy for how to resolve the conflict - is it the server or the
client?

[Chris Boulton] My initial feeling here is that it should be the
client's responsibility on receiving a 200 class response with two
contacts + different GRUU's should de-register the one it doesn't
recognize (The one with the unknown instance ID).

Chris.


>
>-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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From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sip] about call waiting using SUBSCRIBE/NOTIFY!
Date: Wed, 18 Feb 2004 04:32:38 -0800
To: "jack zhao" <embedsong@hotmail.com>
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Hi,

I don't think most folks on the list would describe this feature as=20
"call waiting", but a similar feature is described in=20
draft-ietf-sipping-dialog-package

thanks,
-rohan


On Mar 24, 2003, at 6:12 PM, jack zhao wrote:

> Dear all,
> =A0
> What I expect is like this:
> Assume: There are 2 UAs, A and B.
> =A0
> 1. A is busy, (In IP2006, that means both=A02 lines has=A0a connection=20=

> with somebody)
> 2. B call A and got a=A0486 BUSY reply.
> 3. B decide to wait for A.=A0B=A0send a=A0SUBSCRIBE to A=A0and hang =
up.=A0
> 4. After certain period of time, A finally close one of it's=20
> connection and has a free
> =A0=A0=A0 line.
>  5. A send a NOTIFY to B indicate that he is free to accept =
connection.
> 6. B receive this NOTIFY and start to ring. B UNSUBSCRIBE the event=20
> also.
> 7. After user pickup B, B send a INVITE to A try to establish a=20
> connection.
> 8. A start ringing when it receive INVITE. =46rom now on, it is the =
same=20
> as the
> =A0=A0=A0 regular call out.
> =A0
> If you have got the IETF draft or RFC about call-waiting please tell=20=

> me or send it to me.
> =A0
> Best Regards,
> jack zhao

--Apple-Mail-19-16110967
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Hi,


I don't think most folks on the list would describe this feature as
"call waiting", but a similar feature is described in
draft-ietf-sipping-dialog-package


thanks,

-rohan



On Mar 24, 2003, at 6:12 PM, jack zhao wrote:


<excerpt><color><param>0000,0000,FFFF</param><smaller>Dear =
all,</smaller></color>

=A0

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>What
I expect is like this:</smaller></color></fontfamily>

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>Assume:
There are 2 UAs, A and B.</smaller></color></fontfamily>

=A0

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>1.
A is busy, (In IP2006, that means both=A02 lines has=A0a connection with
somebody)</smaller></color></fontfamily>

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>2.
B call A and got a=A0486 BUSY reply.</smaller></color></fontfamily>

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>3.
B decide to wait for A.=A0B=A0send a=A0SUBSCRIBE to A=A0and hang =
up.=A0</smaller></color></fontfamily>

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>4.
After certain period of time, A finally close one of it's connection
and has a free</smaller></color></fontfamily>

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>=A0=A0=A0
line.</smaller></color></fontfamily>

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>
5. A send a NOTIFY to B indicate that he is free to accept =
connection.</smaller></color></fontfamily>

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>6.
B receive this NOTIFY and start to ring. B UNSUBSCRIBE the event =
also.</smaller></color></fontfamily>

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>7.
After user pickup B, B send a INVITE to A try to establish a
connection.</smaller></color></fontfamily>

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>8.
A start ringing when it receive INVITE. =46rom now on, it is the same as
the</smaller></color></fontfamily>

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>=A0=A0=A0
regular call out.</smaller></color></fontfamily>

=A0

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>If
you have got the IETF draft or RFC about call-waiting please tell me
or send it to me.</smaller></color></fontfamily>

=A0

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>Best
Regards,</smaller></color></fontfamily>

=
<fontfamily><param>Helvetica</param><color><param>0000,0000,FFFF</param><s=
maller>jack
zhao</smaller></color></fontfamily>

</excerpt>=

--Apple-Mail-19-16110967--


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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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Subject: Re: [Sip] Changing Via In NAT
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On Feb 13, 2004, at 11:07 AM, Anil Bollineni wrote:

> Dear All,
>
> In draft-ietf-sip-nat-01.txt, there are SIP extensions for NAT =20
> traversal. The received and rport parameter are used to send back the =20=

> responses correctly.

This is an extremely old reference.  The name was changed to =20
draft-ietf-sip-symmetric-response, and is now published as RFC 3581.

> However if we don't use these extensions, and change the via ip =20
> address and port with that of firewall/port then the SIP responses =20
> will come directly to firewall, in that case the firewall permits the =20=

> responses. UA1 inside the network sends requests to firewall/NAT.
>
>
> UA1
>
>  INVITE sip:user@domain.com SIP/2.0 --------------=E0 Firewall/NAT =20
> =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 =
---------------------------------------------=E0UA2
>
> Via: SIP/2.0/UDP 10.1.1.1:4540=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 INVITE =20
> sip:user@domain.com SIP/2.0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0
>
>  Via: SIP/2.0/UDP firewall_ip:5060
>
> =
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=20
>  (Maintain mapping)
>
>
> the UA2 will send the responses to firewall/NAT and firewall/NAT will =20=

> write back the addresses in Via to original IP/port. I understand from =
=20
> the draft, if there are no SIP ALG functionality in firewall, then the =
=20
> extensions are used to make SIP traverse through NAT. However if we =20=

> implement SIP ALG in firewall then we can make SIP traverse through =20=

> NAT. Can anybody tell that this way is per SIP spec.

The experience of the working group has shown that implementing a SIP =20=

ALG in a firewall is "a bad thing".  I have personal experience with =20
the failings of SIP ALGs at my own company, not the least of which is =20=

that they don't work at all when SIP is integrity-protected using TLS.

Many UAs now support RFC 3581 and can use the STUN protocol to get =20
valid public addresses and ports to advertise in SDP for their RTP =20
traffic. This has been relatively painless compared to working with the =20=

shortcomings of ALGs.

thanks,
-rohan


--Apple-Mail-20-16935754
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On Feb 13, 2004, at 11:07 AM, Anil Bollineni wrote:


<excerpt><fontfamily><param>Arial</param><x-tad-bigger>Dear =
All,</x-tad-bigger></fontfamily>


<fontfamily><param>Arial</param><x-tad-bigger>In
draft-ietf-sip-nat-01.txt, there are SIP extensions for NAT traversal.
The received and rport parameter are used to send back the responses
correctly.=20

</x-tad-bigger></fontfamily></excerpt>

This is an extremely old reference.  The name was changed to
draft-ietf-sip-symmetric-response, and is now published as RFC 3581.


<excerpt><fontfamily><param>Arial</param><x-tad-bigger>However if we
don't use these extensions, and change the via ip address and port
with that of firewall/port then the SIP responses will come directly
to firewall, in that case the firewall permits the responses. UA1
inside the network sends requests to =
firewall/NAT.</x-tad-bigger></fontfamily>



=
<fontfamily><param>Arial</param><x-tad-bigger>UA1</x-tad-bigger></fontfami=
ly>


<fontfamily><param>Arial</param><x-tad-bigger> INVITE
sip:user@domain.com SIP/2.0
=
--------------</x-tad-bigger></fontfamily><x-tad-bigger>=E0</x-tad-bigger>=
<fontfamily><param>Arial</param><x-tad-bigger>
Firewall/NAT =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0
=
---------------------------------------------</x-tad-bigger></fontfamily><=
x-tad-bigger>=E0</x-tad-bigger><fontfamily><param>Arial</param><x-tad-bigg=
er>UA2</x-tad-bigger></fontfamily>


<fontfamily><param>Arial</param><x-tad-bigger>Via: SIP/2.0/UDP
10.1.1.1:4540=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0 INVITE sip:user@domain.com
SIP/2.0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0</x-tad-bigger></fontfamily>


<fontfamily><param>Arial</param><x-tad-bigger> Via: SIP/2.0/UDP
firewall_ip:5060</x-tad-bigger></fontfamily>


=
<fontfamily><param>Arial</param><x-tad-bigger>=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0
(Maintain mapping)</x-tad-bigger></fontfamily>



<fontfamily><param>Arial</param><x-tad-bigger>the UA2 will send the
responses to firewall/NAT and firewall/NAT will write back the
addresses in Via to original IP/port. I understand from the draft, if
there are no SIP ALG functionality in firewall, then the extensions
are used to make SIP traverse through NAT. However if we implement SIP
ALG in firewall then we can make SIP traverse through NAT. Can anybody
tell that this way is per SIP spec.</x-tad-bigger></fontfamily>

</excerpt>

The experience of the working group has shown that implementing a SIP
ALG in a firewall is "a bad thing".  I have personal experience with
the failings of SIP ALGs at my own company, not the least of which is
that they don't work at all when SIP is integrity-protected using TLS.


Many UAs now support RFC 3581 and can use the STUN protocol to get
valid public addresses and ports to advertise in SDP for their RTP
traffic. This has been relatively painless compared to working with
the shortcomings of ALGs.


thanks,

-rohan



--Apple-Mail-20-16935754--


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



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Cc: "'Cullen Jennings'" <fluffy@cisco.com>, sip@ietf.org
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sip] Binary Encoding for SIP
Date: Wed, 18 Feb 2004 04:53:14 -0800
To: Raphael Tryster <raphael@tdsoft.com>
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Hi Raphael,

Just as an FYI:  The "unpopular" encoding for H.248 is ASN.1 encoding=20
of the entire protocol.  What Cullen is proposing is sending all *MIME=20=

bodies* in SIP as a stream of raw binary data (like HTTP). This is=20
instead of sending SIP MIME bodes as a stream of base64-encoded data=20
for example, which is unnecessary in modern protocols, but was required=20=

in email for backwards compatibility reasons.

(btw: I vote for a)

thanks,
-rohan


On Feb 15, 2004, at 12:13 AM, Raphael Tryster wrote:

> Hello Cullen,
>
> Given that this is my first venture into the SIP email forum, and the=20=

> first response to your posting, it is only fitting that I should vote=20=

> for option c:).
>
> During the coming year or two, I hope to develop an opinion on whether=20=

> the issues of backward compatibility and ease of debugging, which seem=20=

> to make the binary option rather unpopular in Megaco, apply to SIP as=20=

> well.=A0 If they do, it could have an adverse affect on World Hunger, =
as=20
> those with the right tools to debug binary messages will eat more, at=20=

> the expense of the rest.
>
> Raphael Tryster
>
> -----Original Message-----
> From:=A0=A0 Cullen Jennings [SMTP:fluffy@cisco.com]
> Sent:=A0=A0 Sunday, 15 February 2004 9:03
> To:=A0=A0=A0=A0 sip@ietf.org
> Subject:=A0=A0=A0=A0=A0=A0=A0 [Sip] Binary Encoding for SIP
>
> =A0=A0=A0
> I submitted a draft on binary encoding for SIP. This improves
> interoperability, improves speed, decreases sizes of messages, and=20
> generally
> makes life better for SIP which will no doubt eventually help with=20
> World
> Hunger. I need feedback, please let me know:
>
> a) Yah, lets do it
>
> b) No this is a stupid ideas and here are the reasons why
>
> c) There has not been many list comments, this draft involves some =
very
> complex issues. I need a year or two to develop an opinion on the=20
> single
> sentence inside it.
>
> d) I agree with Cowboy Willis
>
> Until it arrives in the archives, you can find it at
>
> http://www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.html
>
> or, if you prefer ASCII encodings of everything,
>
> www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.txt
>
> Thanks, Cullen
>
>
>
>
>
>
>
>
> _______________________________________________
> Sip mailing list=A0 https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

--Apple-Mail-21-17346777
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Hi Raphael,


Just as an FYI:  The "unpopular" encoding for H.248 is ASN.1 encoding
of the entire protocol.  What Cullen is proposing is sending all *MIME
bodies* in SIP as a stream of raw binary data (like HTTP). This is
instead of sending SIP MIME bodes as a stream of base64-encoded data
for example, which is unnecessary in modern protocols, but was
required in email for backwards compatibility reasons.


(btw: I vote for a)


thanks,

-rohan



On Feb 15, 2004, at 12:13 AM, Raphael Tryster wrote:


<excerpt><fontfamily><param>Arial</param><smaller>Hello
Cullen,</smaller></fontfamily>=20


<fontfamily><param>Arial</param><smaller>Given that this is my first
venture into the SIP email forum, and the first response to your
posting, it is only fitting that I should vote for option =
c:).</smaller></fontfamily>


<fontfamily><param>Arial</param><smaller>During the coming year or
two, I hope to develop an opinion on whether the issues of backward
compatibility and ease of debugging, which seem to make the binary
option rather unpopular in Megaco, apply to SIP as well.=A0 If they do,
it could have an adverse affect on World Hunger, as those with the
right tools to debug binary messages will eat more, at the expense of
the rest.</smaller></fontfamily>


<fontfamily><param>Tahoma</param><smaller>Raphael
Tryster</smaller></fontfamily>=20


<fontfamily><param>Arial</param><smaller>-----Original
Message-----</smaller></fontfamily>=20

<bold><fontfamily><param>Arial</param><smaller>From:=A0=A0 Cullen =
Jennings
[SMTP:fluffy@cisco.com]</smaller></fontfamily></bold>=20

=
<bold><fontfamily><param>Arial</param><smaller>Sent:=A0=A0</smaller></font=
family></bold>
<fontfamily><param>Arial</param><smaller>Sunday, 15 February 2004
9:03</smaller></fontfamily>=20

=
<bold><fontfamily><param>Arial</param><smaller>To:=A0=A0=A0=A0</smaller></=
fontfamily></bold>
=
<fontfamily><param>Arial</param><smaller>sip@ietf.org</smaller></fontfamil=
y>=20

=
<bold><fontfamily><param>Arial</param><smaller>Subject:=A0=A0=A0=A0=A0=A0=A0=
</smaller></fontfamily></bold>
<fontfamily><param>Arial</param><smaller>[Sip] Binary Encoding for
SIP</smaller></fontfamily>=20


<fixed><fontfamily><param>Courier New</param><smaller>=A0=A0=A0 =
</smaller></fontfamily></fixed>

<fixed><fontfamily><param>Courier New</param><smaller>I submitted a
draft on binary encoding for SIP. This
improves</smaller></fontfamily></fixed>=20

<fixed><fontfamily><param>Courier New</param><smaller>interoperability,
improves speed, decreases sizes of messages, and
generally</smaller></fontfamily></fixed>=20

<fixed><fontfamily><param>Courier New</param><smaller>makes life
better for SIP which will no doubt eventually help with
World</smaller></fontfamily></fixed>=20

<fixed><fontfamily><param>Courier New</param><smaller>Hunger. I need
feedback, please let me know:</smaller></fontfamily></fixed>=20


<fixed><fontfamily><param>Courier New</param><smaller>a) Yah, lets do
it</smaller></fontfamily></fixed>=20


<fixed><fontfamily><param>Courier New</param><smaller>b) No this is a
stupid ideas and here are the reasons
why</smaller></fontfamily></fixed>=20


<fixed><fontfamily><param>Courier New</param><smaller>c) There has not
been many list comments, this draft involves some
very</smaller></fontfamily></fixed>=20

<fixed><fontfamily><param>Courier New</param><smaller>complex issues.
I need a year or two to develop an opinion on the
single</smaller></fontfamily></fixed>=20

<fixed><fontfamily><param>Courier New</param><smaller>sentence inside
it.</smaller></fontfamily></fixed>=20


<fixed><fontfamily><param>Courier New</param><smaller>d) I agree with
Cowboy Willis</smaller></fontfamily></fixed>=20


<fixed><fontfamily><param>Courier New</param><smaller>Until it arrives
in the archives, you can find it at</smaller></fontfamily></fixed>=20


<fixed><fontfamily><param>Courier =
New</param><color><param>0000,0000,EEEE</param><smaller>http://www.employe=
es.org/~fluffy/ietf/draft-jennings-sip-mime-01.html</smaller></color></fon=
tfamily></fixed>=20


<fixed><fontfamily><param>Courier New</param><smaller>or, if you
prefer ASCII encodings of everything,</smaller></fontfamily></fixed>=20


<fixed><fontfamily><param>Courier =
New</param><smaller>www.employees.org/~fluffy/ietf/draft-jennings-sip-mime=
-01.txt</smaller></fontfamily></fixed>=20


<fixed><fontfamily><param>Courier New</param><smaller>Thanks,
Cullen</smaller></fontfamily></fixed>=20









<fixed><fontfamily><param>Courier =
New</param><smaller>_______________________________________________</small=
er></fontfamily></fixed>=20

<fixed><fontfamily><param>Courier New</param><smaller>Sip mailing
list=A0 =
<color><param>0000,0000,EEEE</param>https://www1.ietf.org/mailman/listinfo=
/sip</color></smaller></fontfamily></fixed>=20

<fixed><fontfamily><param>Courier New</param><smaller>This list is for
NEW development of the core SIP
Protocol</smaller></fontfamily></fixed>=20

<fixed><fontfamily><param>Courier New</param><smaller>Use
sip-implementors@cs.columbia.edu for questions on current
sip</smaller></fontfamily></fixed>=20

<fixed><fontfamily><param>Courier New</param><smaller>Use
sipping@ietf.org for new developments on the application of
sip</smaller></fontfamily></fixed>=20

</excerpt>=

--Apple-Mail-21-17346777--


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-publish-03.txt
Date: Wed, 18 Feb 2004 05:20:33 -0800
To: "'sip@ietf.org'" <sip@ietf.org>
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Hi Folks,

I just noticed a minor problem with publish ; it defines a new response 
code:

	412 Precondition Failed

which is confusingly similar to the existing response code:

	580 Precondition Failure

I propose changing The reason phrase for 412 to "Publication Condition 
Failed"

thanks,
-rohan


On Feb 16, 2004, at 12:41 PM, Internet-Drafts@ietf.org wrote:

> A New Internet-Draft is available from the on-line Internet-Drafts 
> directories.
> This draft is a work item of the Session Initiation Protocol Working 
> Group of the IETF.
>
> 	Title		: Session Initiation Protocol (SIP) Extension for Event State
>                               Publication
> 	Author(s)	: A. Niemi
> 	Filename	: draft-ietf-sip-publish-03.txt
> 	Pages		: 40
> 	Date		: 2004-2-16
> 	
> This document describes an extension to the Session Initiation
> Protocol (SIP) for publishing event state used within the framework
> for SIP Event Notification. The first application of this extension
> is targeted at the publication of presence information.
> The mechanism described in this document can be extended to support
> publication of any event state, for which there exists an appropriate
> event package. It is not intended to be a general-purpose mechanism
> for transport of arbitrary data, as there are better-suited
> mechanisms for this purpose (FTP, HTTP, etc.)
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-publish-03.txt
>
> To remove yourself from the IETF Announcement list, send a message to
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> Internet-Drafts are also available by anonymous FTP. Login with the 
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>
> A list of Internet-Drafts directories can be found in
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>
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From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sip] RE: Abstraction:  SIP instance identifiers requirements
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On Feb 17, 2004, at 10:37 AM, Brian Stucker wrote:

> I think that your list is missing the opportunity to define this 
> instance
> ID in such a way that it can be used to identify a particular endpoint
> for other operations beyond simply routing.

For example please?  Your draft jumps straight into mechanism.

thanks,
-rohan


> The reuse potential seems
> very good if we don't lock the instance identifier down to simply
> routing operations, and instead open it up for other general-purpose
> operations.
>
> That is at the heart of my draft. Don't limit yourself to one 
> application
> when with a small amount of work we can use this to solve other 
> problems
> both current and future. Likewise, don't exclude routing operations 
> from
> making use of the instance identifier either.
>
> Are you concerned that opening this up to having the instance 
> identifier
> transported (in addition to the registered contact) outside of the
>  registered contact is really going to be that complex?
>
>  Regards,
>
> Brian
>
>
>
>
> > -----Original Message-----
> > From: Dean Willis [mailto:dean.willis@softarmor.com]
> > Sent: Tuesday, February 17, 2004 12:11 PM
> > To: Cullen Jennings
> > Cc: Stucker, Brian [NGC:B617:EXCH]; Jonathan Rosenberg; sip@ietf.org
> > Subject: Abstraction: SIP instance identifiers requirements
> >
> >
> >
> > Based on the discussion of "routing" based on instance identifiers, I
> > think we may be getting involved in triggerwork that may become more
>  > complex than is actually required, because we're all reasoning from
> > secondary positions. So I'd like to take a step back and
> > consider how we
> > got here, and what we think we're trying to accomplish.
> >
> > I see four high-level cases around request routing that illustrate
> > different aspects of this discussion.
> >
> > 1) General case: We wish to address a request to an AOR, such
> > that any
> > contact at that AOR might respond. This is basic 2543-type
> > routing behavior.
> >
> > 2) Capability case: We wish to address a request to an AOR, such that
> > any contact capable of satisfying the request in a way that we like
> > might respond. This behavior is brought out in callee-caps and
> > caller-preferences and the accept/supported/required syntax of 3261.
> >
> > 3) Followups to a specific instance: We have previously done
> > something
> > (such as case 1 or 2) such that the operation of interest has
> > resolved
> > to a specific contact instance associated with an AoR, and we wish
> > subsequent operations (for example, a transfer) to reach that same
> > specific contact instance, even if that specific contact instance is
> > hidden behind an indirection layer (firewall, anonymizer, etc.). As I
> > understand it, this is the heart of the GRUU concept.
> >
> > 4) Selection of a specific instance in an initial request: We have a
> > request that we wish to reach one specific instance of a contact
> > associated with an AoR. It has been proposed that we
> > accomplish this by
> > binding some sort of instance tag to the AoR. It has been
> > proposed that
> > we use some sort of URI convention (like the msuri), +based
> > subaddressing, or that we use a header convention for binding the
> > instance tag to the AoR. To my knowledge, we haven't really described
> > the requirement or use case for reaching a specific instance on an
> > initial request, but I can think of several offhand
> > (including diagnostics).
> >
> > It seems evident that there is a huge overlap between the
> > requirements
> > of 3 and 4. Indeed, I think #3 (GRUU) came out of early ideas
> > I had on
> > using a REGISTER-response extension to assign an instance-AoR to a
> > Contact. It would appear that IF each contact for an AoR were
> > to have an
> > associated GRUU, and that knowledge of those GRUU assigments were
> > available to a requestor, that the requester could use the
> > target-specific GRUU to achieve the use cases I currently
> > imagine for #4.
> >
> >
> > What am I missing here?
> >
> > --
> > Dean
> >

--Apple-Mail-22-19573831
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	charset=US-ASCII
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On Feb 17, 2004, at 10:37 AM, Brian Stucker wrote:


<excerpt><smaller>I think that your list is missing the opportunity to
define this instance </smaller>

<smaller>ID in such a way that it can be used to identify a particular
endpoint </smaller>

<smaller>for other operations beyond simply routing. 

</smaller></excerpt>

For example please?  Your draft jumps straight into mechanism.


thanks,

-rohan



<excerpt><smaller>The reuse potential seems </smaller>

<smaller>very good if we don't lock the instance identifier down to
simply</smaller> 

<smaller>routing operations, and instead open it up for other
general-purpose</smaller> 

<smaller>operations.</smaller> 


<smaller>That is at the heart of my draft. Don't limit yourself to one
application</smaller> 

<smaller>when with a small amount of work we can use this to solve
other problems</smaller> 

<smaller>both current and future. Likewise, don't exclude routing
operations from</smaller> 

<smaller>making use of the instance identifier either.</smaller> 


<smaller>Are you concerned that opening this up to having the instance
identifier</smaller> 

<smaller>transported (in addition to the registered contact) outside
of the</smaller> 

 <smaller>registered contact is really going to be that
complex?</smaller> 


 <smaller>Regards,</smaller> 


<smaller>Brian</smaller> 





<smaller>> -----Original Message-----</smaller> 

<smaller>> From: Dean Willis
[<color><param>0000,0000,EEEE</param>mailto:dean.willis@softarmor.com</color>]</smaller> 

<smaller>> Sent: Tuesday, February 17, 2004 12:11 PM</smaller> 

<smaller>> To: Cullen Jennings</smaller> 

<smaller>> Cc: Stucker, Brian [NGC:B617:EXCH]; Jonathan Rosenberg;
sip@ietf.org</smaller> 

<smaller>> Subject: Abstraction: SIP instance identifiers
requirements</smaller> 

<smaller>> </smaller>

<smaller>> </smaller>

<smaller>> </smaller>

<smaller>> Based on the discussion of "routing" based on instance
identifiers, I </smaller>

<smaller>> think we may be getting involved in triggerwork that may
become more</smaller> 

 <smaller>> complex than is actually required, because we're all
reasoning from </smaller>

<smaller>> secondary positions. So I'd like to take a step back and </smaller>

<smaller>> consider how we </smaller>

<smaller>> got here, and what we think we're trying to
accomplish.</smaller> 

<smaller>> </smaller>

<smaller>> I see four high-level cases around request routing that
illustrate </smaller>

<smaller>> different aspects of this discussion.</smaller> 

<smaller>> </smaller>

<smaller>> 1) General case: We wish to address a request to an AOR,
such </smaller>

<smaller>> that any </smaller>

<smaller>> contact at that AOR might respond. This is basic 2543-type </smaller>

<smaller>> routing behavior.</smaller> 

<smaller>> </smaller>

<smaller>> 2) Capability case: We wish to address a request to an AOR,
such that </smaller>

<smaller>> any contact capable of satisfying the request in a way that
we like </smaller>

<smaller>> might respond. This behavior is brought out in callee-caps
and </smaller>

<smaller>> caller-preferences and the accept/supported/required syntax
of 3261.</smaller> 

<smaller>> </smaller>

<smaller>> 3) Followups to a specific instance: We have previously
done </smaller>

<smaller>> something </smaller>

<smaller>> (such as case 1 or 2) such that the operation of interest
has </smaller>

<smaller>> resolved </smaller>

<smaller>> to a specific contact instance associated with an AoR, and
we wish </smaller>

<smaller>> subsequent operations (for example, a transfer) to reach
that same </smaller>

<smaller>> specific contact instance, even if that specific contact
instance is </smaller>

<smaller>> hidden behind an indirection layer (firewall, anonymizer,
etc.). As I </smaller>

<smaller>> understand it, this is the heart of the GRUU
concept.</smaller> 

<smaller>> </smaller>

<smaller>> 4) Selection of a specific instance in an initial request:
We have a </smaller>

<smaller>> request that we wish to reach one specific instance of a
contact </smaller>

<smaller>> associated with an AoR. It has been proposed that we </smaller>

<smaller>> accomplish this by </smaller>

<smaller>> binding some sort of instance tag to the AoR. It has been </smaller>

<smaller>> proposed that </smaller>

<smaller>> we use some sort of URI convention (like the msuri), +based </smaller>

<smaller>> subaddressing, or that we use a header convention for
binding the </smaller>

<smaller>> instance tag to the AoR. To my knowledge, we haven't really
described </smaller>

<smaller>> the requirement or use case for reaching a specific
instance on an </smaller>

<smaller>> initial request, but I can think of several offhand </smaller>

<smaller>> (including diagnostics).</smaller> 

<smaller>> </smaller>

<smaller>> It seems evident that there is a huge overlap between the </smaller>

<smaller>> requirements </smaller>

<smaller>> of 3 and 4. Indeed, I think #3 (GRUU) came out of early
ideas </smaller>

<smaller>> I had on </smaller>

<smaller>> using a REGISTER-response extension to assign an
instance-AoR to a </smaller>

<smaller>> Contact. It would appear that IF each contact for an AoR
were </smaller>

<smaller>> to have an </smaller>

<smaller>> associated GRUU, and that knowledge of those GRUU
assigments were </smaller>

<smaller>> available to a requestor, that the requester could use the </smaller>

<smaller>> target-specific GRUU to achieve the use cases I currently </smaller>

<smaller>> imagine for #4.</smaller> 

<smaller>> </smaller>

<smaller>> </smaller>

<smaller>> What am I missing here?</smaller> 

<smaller>> </smaller>

<smaller>> --</smaller> 

<smaller>> Dean</smaller> 

<smaller>></smaller> 

</excerpt>
--Apple-Mail-22-19573831--


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sip] Proxy behavior
Date: Wed, 18 Feb 2004 05:44:35 -0800
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On Feb 4, 2004, at 2:00 PM, Rajesh Khandewale wrote:

> I have a question about the proxy behavior under the following 
> scenario,
> could someone please provide a clarification.
>
> 1. UAC sends an INVITE via proxy.
> 2. Proxy responds back with 100 trying message to UAC and forwards 
> INVITE to
> UAS.
> 3. UAS sends 200 OK which is relayed by the proxy to UAC.
> 4. UAC sends an ACK which is being processed by the proxy to be 
> forwarded to
> the UAS.
> ** Here ACK has sequence number 1
> 5. UAC sends another INVITE and incidentally this gets forwarded to 
> the UAS
> before the ACK from previous INVITE by the proxy.
> ** Here INVITE has sequence number 2
> 6. Since UAS did not receive the ACK for previous INVITE (nor did it
> received CANCEL/BYE), but received another INVITE with same Call-ID 
> etc. it
> throws a "500 Internal Server Error" error.
>
> According to RFC 3261, the UAS behavior to throw a "500 Internal Server
> Error" is correct. But I am not sure about the proxy behavior here, 
> could
> not be clearly figured out of 3261.
>
> Is proxy supposed to correlate all the messages based on the sequence 
> number
> and call-id and essentially send an ACK before sending out second 
> INVITE. Or
> they are completely different transactions and this is a normal proxy
> behavior.

They are completely different transactions.  The proxy just forwards 
the 500 response normally.

In future, please direct these kinds of requests to 
sip-implementors@cs.columbia.edu

thanks,
-rohan


> Thanks,
>
> Rajesh.
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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] Transport parameter in request URI
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Hi,

In future, please ask these types of "how to" questions on 
sip-implementors@cs.columbia.edu

UA2 should send its BYE request to the address in the Contact from UA1. 
  The transport parameter is optional, so if there is no transport 
parameter in its request URI, UA2 will use the default transport (UDP 
for sip:  and TCP for sips: ).  The response to the BYE will use the 
same transport as the request.

If you have a followup question, please post it to sip-implementors.

thanks,
-rohan


On Feb 5, 2004, at 6:52 AM, Vijay Gaur wrote:

> Hi,
>    My setup is like this:
>
>       UA1----->(TLS)----->Proxy------>(UDP)--->UA2
>
> Here UA1 makes call and dialog gets established between UA1 and UA2. 
> Contact of UA2 contains transport parameter tls (Contact: 
> sip@ua1.com;transport=tls). Now when UA2 generates BYE it doesn't put 
> any transport parameter in request URI (BYE sip:ua1.com SIP/2.0). 
> Because of no transport parameter BYE doesn't get 200OK.
> Now my question is which transport (TLS or UDP) UA2 should put in 
> request URI?
> RFC 3261 says that in request message agent should put transport 
> parameter in request URI with transport which it is using.
> If I put transport=udp in request URI UA2 BYE doesn't get 200OK, but 
> in case of transport=tls it works fine.
>
> I appreciate your help.
> Thanks,
> Vijay
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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To: Brian Stucker <bstucker@nortelnetworks.com>
CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: Re: [Sip] SIP instance identifiers
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Responding to both Jonathan and Brian inline...

	Paul

Brian Stucker wrote:
> Inline...
> 
>  > -----Original Message-----
>  > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
...
>  > > Regarding the form of guids, I don't think any single
>  > scheme works in
>  > > all cases. My preference is to:
>  > >
>  > > - represent the instance id as a URI
>  > >
>  > > - make it the responsibility of the endpoint to select a URI
>  > >   that is unique
>  > >
>  > > - Recommend that a URN be used.
>  > >
>  > > - Define a couple of new URN namespaces that are especially
>  > >   suitable for this purpose. (I have in mind urn:mac:xxx for
>  > >   mac addresses, and possibly urn:random:xxx for random numbers
>  > >   in some large range.)
>  >
>  > This is a well trod area, and I think it would behoove us to look at
>  > work in other groups. Dean has mentioned HIP as one obvious
>  > candidate;
>  > generally speaking this exact issue - of endpoint identifiers
>  > that work
>  > better than IP address, is the subject of much other work at this
>  > moment, including MAST. I'm not familiar enough with it to
>  > know if its
>  > relevant... its on my reading list for this ietf.

I'm fine with reusing existing work. What is HIP?

My primary point is to give the endpoint flexibility in the form of 
identifier it uses by representing the identifier as a URI.

Beyond that, I wanted to ensure that we had a way to use a mac address 
as one of the options. I didn't find a way to do that in URI/URN form, 
which is why I suggest defining a new URN namespace for this. If there 
is some other standard that already achieves this, fine.

I'm not at all attached to any particular way of using random numbers. 
I'm happy with pretty much anything that permits an instance of a 
software endpoint on a multi-user/multi-application box to come up with 
a guid.

> This represents additional work as well. =)
> 
> I think we're limiting ourselves to just MACs and random numbers.

I'm not suggesting *limiting* to macs and random numbers. Just that tese 
be available options.

 > Having
> looked around a bit, I've seen material (I want to say it's from WebDAV
> but I could be remembering incorrectly) that the MAC address isn't always
> a good thing to pass around because you can figure out that the UA is
> running on a particular piece of hardware without otherwise having access
> to their MAC (because you're on the other side of the world many routers
> away). That can be used to determine a method of attack because you can
> inferr what operating systems, etc might be running if you know what kind
> of hardware that box is plugged into the network with.
> 
> That said, if you want to use a MAC, fine. Buyer beware. If you want to use
> a random number, fine. If you want to use the GPS coordinates of the box to
> some ridiculous level of precision (and that's good enough for your 
> application)
> fine.
> 
> I don't see why we necessarily need to put any restrictions on this
> instance ID to make it a URN or anything else when it really need not be 
> any
> more complex than a VIA branch. I also don't see a reason why we should 
> expect
> other nodes in the network to be able to use the instance identifier for 
> anything
> other than a token (ie. if it's a mac address, and two UA instances do 
> something
> because it's a MAC address, that's OK, but other's may not take that 
> tact). There's
> no guarantee others are going to be able to take whatever action might 
> be denoted
> by the contents of the instance identifier itself, so it becomes entirely
> proprietary at that point.

The reason to use URIs/URNs is to manage the namespaces. Your MAC-based 
id may collide with my timestamp-based id if there isn't something to 
distinguish that they come from different namespaces. For that matter, 
your mac-based id may conflict with my mac-based id if we choose to 
format them differently.

By using URIs, there is always some specification that governs the 
format and defines uniqueness within that format, while the scheme name 
itself ensures uniqueness of ids generated based on different 
specifications.

>  > > This 1:0..1 relationship imposes another error condition on
>  > registrars.
>  > > If an attempt is made to register a contact with an
>  > instance id in use
>  > > by some other contact of the same AOR, the registrar will
>  > have to refuse
>  > > it.
>  >
>  > Well, thats one option.

You are right - there are other options.

> This is why I put some boundaries on the length of the GUID and some
> basic guidelines, to make this very unlikely to occur for reasonably
> well behaved implementations.

That assumes it isn't done on purpose.

In any case, the whole concept of GRUUs is based on the uniqueness, so 
the registrar is bound to enforce this uniqueness, one way or another.

>  > This needs to be spelled out somewhere, and a suitable error
>  > > identified. (I don't know if any of the existing responses
>  > works for
>  > > this. At the least, a new Reason code is likely to be needed.)
>  >
>  > Here is the use case for this. The use case is when a UA crashes,
>  > recovers, obtains a new IP, and re-registers. In this case, several
>  > things could possibly happen:
>  >
>  > 1. The registrar accepts the second registration, as per normal
>  > registration procedures. An INVITE to the gruu bound to that
>  > id/aor pair
>  > gets translated into BOTH contacts, causing the request to
>  > fork. There
>  > is never any response from the old IP. Within one refresh
>  > interval, the
>  > old contact is removed and all is well.
>  >
>  > 2. The registrar accepts the second registration, and automatically
>  > deletes the old one. I believe Brian proposed this behavior.
>  >
>  > 3. The registrar returns an error. The client fetches the
>  > current list
>  > of contacts, finds the old one listed, and unregisters it.
>  >
>  > Approach 1 is problematic, as it will introduce herfp; any
>  > non-2xx from
>  > the new contact will wait for the 2nd branch to timeout.

This is also problematic because it violates the invariant on which GURR 
is based - that it always represents a single endpoint.

>  >  Its not clear
>  > to me whether 2 or 3 is more right. It depends a lot on who owns the
>  > policy for how to resolve the conflict - is it the server or
>  > the client?

I agree that the right choice here isn't immediately obvious. But having 
the registrar generate an error is a way to push the decision back on 
the endpoint. If it wants to prevail then it can send a new REGISTER 
that removes the old conflicting contact as well as adding its own new 
one. Or it can just give up.

	Paul


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To: Chris Boulton <CBoulton@ubiquity.net>
CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: Re: [Sip] SIP instance identifiers
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Chris,

See my prior response to Jonathan and Brian. I agree it is the 
endpoint's responsibility to fix this. But the registrar still needs to 
preserve uniqueness of the identifier. Your proposal has the potential 
of having a non-unique identifier.

	Paul

Chris Boulton wrote:
> 
>>This needs to be spelled out somewhere, and a suitable error
>>
>>>identified. (I don't know if any of the existing responses works for
>>>this. At the least, a new Reason code is likely to be needed.)
>>
>>Here is the use case for this. The use case is when a UA crashes,
>>recovers, obtains a new IP, and re-registers. In this case, several
>>things could possibly happen:
>>
>>1. The registrar accepts the second registration, as per normal
>>registration procedures. An INVITE to the gruu bound to that id/aor
> 
> pair
> 
>>gets translated into BOTH contacts, causing the request to fork. There
>>is never any response from the old IP. Within one refresh interval, the
>>old contact is removed and all is well.
>>
>>2. The registrar accepts the second registration, and automatically
>>deletes the old one. I believe Brian proposed this behavior.
>>
>>3. The registrar returns an error. The client fetches the current list
>>of contacts, finds the old one listed, and unregisters it.
>>
>>
>>
>>Approach 1 is problematic, as it will introduce herfp; any non-2xx from
>>the new contact will wait for the 2nd branch to timeout. Its not clear
>>to me whether 2 or 3 is more right. It depends a lot on who owns the
>>policy for how to resolve the conflict - is it the server or the
> 
> client?
> 
> [Chris Boulton] My initial feeling here is that it should be the
> client's responsibility on receiving a 200 class response with two
> contacts + different GRUU's should de-register the one it doesn't
> recognize (The one with the unknown instance ID).
> 
> Chris.


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Subject: Re: [Sip] draft-khartabil-sip-auth-analysis-00.txt
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> >   realm
> >     A string to be displayed to users so they know which username and
> >     password to use.
> 
> I don't think this takes into consideration pre-configuration.

That's not really the point; the realm needs to provide enough of a hint
for the UA to determine which credentials it should use.  Setting up a
configuration that requires different credentials for the same realm at
different servers is dumb, and not something I think we need to spend
much worry on.

> >                 ... This string should contain at least the name of
> >    the host performing the authentication and might additionally
> >    indicate the collection of users who might have access. An example
> >    might be "registered_users@gotham.news.com".
> 
> Can we mandate that for SIP??

I don't think that we can mandate anything that is by nature an
administrative decision - we already provide good advice.  However, 3261
does say (22.1):

      o  Realm strings MUST be globally unique.  It is RECOMMENDED that
         a realm string contain a hostname or domain name, following the
         recommendation in Section 3.2.1 of RFC 2617 [17].

> > As a practical matter, if you have the same username and password in
> > both domains it doesn't matter - if you don't, then it's an impossibly
> > clumsy situation (this problem is part of the reason that we defined
> > proxy authentication to by hop-by-hop in HTTP in the first place).
> 
> Perhaps I was not clear in my text. Imagine this scenario: I have cached my credentials for an outbound proxy with realm proxy.nokia.com from an earlier transaction. Now I send a new INVITE routed through the outbound proxy that gets a 407 from a second proxy with the same realm. How do I know if the 407 came from the outbound proxy (the username password might have changed in the meantime) or from the next hop proxy?

What difference does it make?  You just use the credentials you have in
either case, or you fail...  As I pointed out, you can tell the
difference if the server is using qop.


> > If the proxy is using the 2617 version of Digest (with the qop
> > attribute) as opposed to the 1867 version, then the UA can tell
> > the difference.  The message flow looks like this:
> > 
> >         UA                    P1                  P2
> >     a    +------REQ 1--------->|                   |
> >     b    |<----407(1)----------+                   |
> >     c    +------REQ 2 -------->+----REQ 2--------->|
> >     e    |<---407(2)-----------+<-----407(2)-------|
> >          +------REQ 3--------->+----REQ 3--------->|
> > 
> >  a) initial request
> >  b) challenge from P1, with the qop parameter specified
> >  c) retried request, with a qop parameter, and therefor also a cnonce
> >     chosen by the UA; it is passed on (now without 
> > authentication) to P2
> >  d) P2 challenges, also specifying a qop parameter
> >     P2 challenge is forwarded by P1, with the WWW-Authenticate
> >     header as provided by P2, but also with an Authentication-Info
> >     provided by P1.
> > 
> > The UA can recognize that authentication to P1 was successful, because
> > P1 returned an Authentication-Info header that contained the cnonce
> > and nc values from message c (REQ 2), and a correct response hash to
> > show that P1 knows the authentication secret.
> 
> This is not mandated for SIP entities, maybe it aught to be.

Actually, it is, but it's the usual unfortunate incorrect intrepretation
of SHOULD that is biting us again.  RFC 2617 says:

   qop-options
     This directive is optional, but is made so only for backward
     compatibility with RFC 2069 [6]; it SHOULD be used by all
     implementations compliant with this version of the Digest scheme.

Since SIP references only 2617, and backward compatibility with 1867 is
not important, all SIP servers should be using qop-options.

> > > 10. Authentication-Info
> > >
> > >  RFC 3261 allows the usage of Authentication-Info header. The BNF in
> > >  RFC 3261 allows multiple authentication-info headers where RFC 2617
> > >  allows only one. Is it only the terminating UAS that is allowed to
> > >  insert this header. If so, why allow multiples to be 
> > present. If not,
> > >  how does the UAC know which proxy added this header? It cannot know
> > >  since there is no parameter indicating so, not even nonce.
> > 
> > As I noted earlier, 2617 allows only one because proxy authentication
> > is hop-by-hop, but the cnonce value allows the UA to disambiguate them
> > (if they are carefully constructed, but then they should be anyway).
> 
> SIP BNF allows multiples of those headers. Is this a BNF bug? If not, then some Authentication-Info parameters need to be mandated.

The BNF is never the last word - the text MUST be read too, and the
requirement there is, I think, already clear.

-- 
Scott Lawrence        
  Pingtel Corp.   



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Subject: RE: [Sip] Changing Via In NAT
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An ALG is an ALG: a B2BUA.
=20
Do whatever you want.  It is not subject to standardization.  It is not =
a good idea.  Do so at your own peril.

-----Original Message-----
From: Anil Bollineni [mailto:ABollineni@netscreen.com]
Sent: Friday, February 13, 2004 2:08 PM
To: 'sip@ietf.org'
Subject: [Sip] Changing Via In NAT



Dear All,

=20

In draft-ietf-sip-nat-01.txt, there are SIP extensions for NAT =
traversal. The received and rport parameter are used to send back the =
responses correctly. However if we don't use these extensions, and =
change the via ip address and port with that of firewall/port then the =
SIP responses will come directly to firewall, in that case the firewall =
permits the responses. UA1 inside the network sends requests to =
firewall/NAT.

=20

UA1=20

INVITE sip:user@domain.com SIP/2.0 ----------------> Firewall/NAT        =
       ----------------------------------------------->UA2

Via: SIP/2.0/UDP 10.1.1.1:4540                          INVITE =
sip:user@domain.com SIP/2.0                            =20

Via: SIP/2.0/UDP firewall_ip:5060

                                                                        =
(Maintain mapping)=20

=20

the UA2 will send the responses to firewall/NAT and firewall/NAT will =
write back the addresses in Via to original IP/port. I understand from =
the draft, if there are no SIP ALG functionality in firewall, then the =
extensions are used to make SIP traverse through NAT. However if we =
implement SIP ALG in firewall then we can make SIP traverse through NAT. =
Can anybody tell that this way is per SIP spec.

=20

Thank you,

Anil

=20

This email contains material that is confidential. The content of this =
email is for the sole use of the intended recipient(s). Any review or =
distribution by persons other than the intended recipient(s) without the =
express permission of NetScreen Technologies, Inc. is strictly =
prohibited. If you are not the intended recipient, please contact the =
sender and delete/destroy all copies of this email and any related =
attachments. NetScreen does not guarantee the accuracy or completeness =
of third party materials or information.

=20


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</HEAD>
<BODY lang=3DEN-US vLink=3Dpurple link=3Dblue>
<DIV><SPAN class=3D586075814-18022004><FONT face=3DArial color=3D#0000ff =
size=3D2>An ALG=20
is an ALG: a B2BUA.</FONT></SPAN></DIV>
<DIV><SPAN class=3D586075814-18022004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D586075814-18022004><FONT face=3DArial color=3D#0000ff =
size=3D2>Do=20
whatever you want.&nbsp; It is not subject to standardization.&nbsp; It =
is not a=20
good idea.&nbsp; Do so at your own peril.</FONT></SPAN></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> Anil Bollineni=20
  [mailto:ABollineni@netscreen.com]<BR><B>Sent:</B> Friday, February 13, =
2004=20
  2:08 PM<BR><B>To:</B> 'sip@ietf.org'<BR><B>Subject:</B> [Sip] Changing =
Via In=20
  NAT<BR><BR></FONT></DIV>
  <DIV class=3DSection1>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Dear =
All,</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">In =
draft-ietf-sip-nat-01.txt,=20
  there are SIP extensions for NAT traversal. The received and rport =
parameter=20
  are used to send back the responses correctly. However if we don't use =
these=20
  extensions, and change the via ip address and port with that of =
firewall/port=20
  then the SIP responses will come directly to firewall, in that case =
the=20
  firewall permits the responses. UA1 inside the network sends requests =
to=20
  firewall/NAT.</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">UA1 </SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">INVITE =
sip:user@domain.com SIP/2.0=20
  --------------</SPAN></FONT><FONT face=3DWingdings size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Wingdings">=E0</SPAN></FONT><FONT=20
  face=3DArial size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">=20
  Firewall/NAT=20
  =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;=20
  ---------------------------------------------</SPAN></FONT><FONT=20
  face=3DWingdings size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Wingdings">=E0</SPAN></FONT><FONT=20
  face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">UA2</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Via: SIP/2.0/UDP=20
  =
10.1.1.1:4540&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
  INVITE sip:user@domain.com SIP/2.0=20
  =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;=20
  </SPAN></FONT></P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 2.5in; TEXT-INDENT: =
0.5in"><FONT=20
  face=3DArial size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Via:=20
  SIP/2.0/UDP firewall_ip:5060</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  (Maintain mapping) </SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">the UA2 will send the =
responses to=20
  firewall/NAT and firewall/NAT will write back the addresses in Via to =
original=20
  IP/port. I understand from the draft, if there are no SIP ALG =
functionality in=20
  firewall, then the extensions are used to make SIP traverse through =
NAT.=20
  However if we implement SIP ALG in firewall then we can make SIP =
traverse=20
  through NAT. Can anybody tell that this way is per SIP =
spec.</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Thank =
you,</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Anil</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
  <P><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">This email contains =
material that=20
  is confidential. The content of this email is for the sole use of the =
intended=20
  recipient(s). Any review or distribution by persons other than the =
intended=20
  recipient(s) without the express permission of NetScreen Technologies, =
Inc. is=20
  strictly prohibited. If you are not the intended recipient, please =
contact the=20
  sender and delete/destroy all copies of this email and any related=20
  attachments. NetScreen does not guarantee the accuracy or completeness =
of=20
  third party materials or information.</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV></BLOCKQUOTE></BODY></HTML>

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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  Wed Feb 18 10:47:48 2004
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Subject: RE: [Sip] CallId question
Date: Wed, 18 Feb 2004 10:44:09 -0500
Message-ID: <4A3384433CE2AB46A63468CB207E209DB1A19D@zoe.office.snowshore.com>
Thread-Topic: [Sip] CallId question
Thread-Index: AcPweESs+VPRa1k1S1mF69deO8TgPgFtkUHQ
From: "Eric Burger" <eburger@snowshore.com>
To: "Anil Bollineni" <ABollineni@netscreen.com>, <sip@ietf.org>
Cc: "Cullen Jennings" <fluffy@cisco.com>
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Upon reflection, this is an interesting theoretical problem that =
statistically is not a problem.

What is a Call-Id?  It is an opaque, globally unique string generated by =
the UAC.  To promote uniqueness, 3261 requires a string identifying the =
UAC, either IP address or hostname.  Note that NOTHING in SIP places =
semantics on the host part of the Call-Id.

So, in reality, don't bother to mung the Call-Id.  What is the =
likelihood of two UAC's having the following properties AT THE SAME =
TIME:
	same randomly generated LHS (unlikely, but possible)
	same IP address (possible)
	same destination (very unlikely)

IIF all of these conditions are SIMULTANEOUSLY true, then there is a =
problem, in that there will be two dialogs with the same dialog =
identifier.  You've got bigger problems then what to change an IP =
address to :-)


That said, if the reason you are munging the Call-Id host part is for =
network hiding, then you really, really, really should be hacking the =
user part, as well.  I'll leave the security considerations section as =
an exercise for the reader.

P.S., this whole thing breaks lots of stuff.  I'm sure Henry will revoke =
my License to SIP for even explaining some of it :-)


-----Original Message-----
From: Cullen Jennings [mailto:fluffy@cisco.com]
Sent: Tuesday, February 10, 2004 6:34 PM
To: Anil Bollineni; sip@ietf.org
Subject: Re: [Sip] CallId question



The NAT/FW must not go changing this stuff. If it does, a S/MIME signed =
sipfrag will fail. There is no need for the NAT to go change call IDs



On 2/10/04 12:27 PM, "Anil Bollineni" <ABollineni@netscreen.com> wrote:


Hello,
=20
            I have a question about CallId. Suppose two UA's generate =
callid's putting theire hostnames and send to firewall/NAT.
            UA1:INVITE
                 CallID: call1@192.10.10.10
            UA2: INVITE
                    CallId: call1@192.10.10.12
           =20
            Because those callid information contain IP addresses, the =
firewall/NAT should translate these IP addresses by putting it's own IP =
in callId. The callId's are changed to call1@firewall_ip, =
call1@firewall_ip. When responses come back from their respective UAS's =
the firewall/NAT translates the it's IP address in callId's to their =
original IP's. My question is should I change the string "call1" to =
something generated by firewall or Is it OK to leave like that. Even =
though they have same CallId, the proxy should discriminate the calls by =
from-tag, callid, and to-tag.
According to Spec, any UA should have a different means of generating =
callid's with those of other UA's.=20
=20
Anything to do correctly to the above procedure?
=20
Thanks in advance,
Anil
This email contains material that is confidential. The content of this =
email is for the sole use of the intended recipient(s). Any review or =
distribution by persons other than the intended recipient(s) without the =
express permission of NetScreen Technologies, Inc. is strictly =
prohibited. If you are not the intended recipient, please contact the =
sender and delete/destroy all copies of this email and any related =
attachments. NetScreen does not guarantee the accuracy or completeness =
of third party materials or information.


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



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CC: Anil Bollineni <ABollineni@netscreen.com>, sip@ietf.org,
        Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] CallId question
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I agree that you shouldn't muck with the call-id, but for different 
reasons. From 3261:

    In a new request created by a UAC outside of any dialog, the Call-ID
    header field MUST be selected by the UAC as a globally unique
    identifier over space and time unless overridden by method-specific
    behavior.  All SIP UAs must have a means to guarantee that the Call-
    ID header fields they produce will not be inadvertently generated by
    any other UA.

A UA behind a NAT can't use its IP address in its call-id because that 
won't be guaranteed to be unique. It might be able to use its DNS name. 
If it doesn't have one, then it is still obligated to come up with 
something unique. Otherwise it is just broken.

	Paul

Eric Burger wrote:
> Upon reflection, this is an interesting theoretical problem that statistically is not a problem.
> 
> What is a Call-Id?  It is an opaque, globally unique string generated by the UAC.  To promote uniqueness, 3261 requires a string identifying the UAC, either IP address or hostname.  Note that NOTHING in SIP places semantics on the host part of the Call-Id.
> 
> So, in reality, don't bother to mung the Call-Id.  What is the likelihood of two UAC's having the following properties AT THE SAME TIME:
> 	same randomly generated LHS (unlikely, but possible)
> 	same IP address (possible)
> 	same destination (very unlikely)
> 
> IIF all of these conditions are SIMULTANEOUSLY true, then there is a problem, in that there will be two dialogs with the same dialog identifier.  You've got bigger problems then what to change an IP address to :-)
> 
> 
> That said, if the reason you are munging the Call-Id host part is for network hiding, then you really, really, really should be hacking the user part, as well.  I'll leave the security considerations section as an exercise for the reader.
> 
> P.S., this whole thing breaks lots of stuff.  I'm sure Henry will revoke my License to SIP for even explaining some of it :-)
> 
> 
> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Tuesday, February 10, 2004 6:34 PM
> To: Anil Bollineni; sip@ietf.org
> Subject: Re: [Sip] CallId question
> 
> 
> 
> The NAT/FW must not go changing this stuff. If it does, a S/MIME signed sipfrag will fail. There is no need for the NAT to go change call IDs
> 
> 
> 
> On 2/10/04 12:27 PM, "Anil Bollineni" <ABollineni@netscreen.com> wrote:
> 
> 
> Hello,
>  
>             I have a question about CallId. Suppose two UA's generate callid's putting theire hostnames and send to firewall/NAT.
>             UA1:INVITE
>                  CallID: call1@192.10.10.10
>             UA2: INVITE
>                     CallId: call1@192.10.10.12
>             
>             Because those callid information contain IP addresses, the firewall/NAT should translate these IP addresses by putting it's own IP in callId. The callId's are changed to call1@firewall_ip, call1@firewall_ip. When responses come back from their respective UAS's the firewall/NAT translates the it's IP address in callId's to their original IP's. My question is should I change the string "call1" to something generated by firewall or Is it OK to leave like that. Even though they have same CallId, the proxy should discriminate the calls by from-tag, callid, and to-tag.
> According to Spec, any UA should have a different means of generating callid's with those of other UA's. 
>  
> Anything to do correctly to the above procedure?
>  
> Thanks in advance,
> Anil
> This email contains material that is confidential. The content of this email is for the sole use of the intended recipient(s). Any review or distribution by persons other than the intended recipient(s) without the express permission of NetScreen Technologies, Inc. is strictly prohibited. If you are not the intended recipient, please contact the sender and delete/destroy all copies of this email and any related attachments. NetScreen does not guarantee the accuracy or completeness of third party materials or information.
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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  Wed Feb 18 11:13:36 2004
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Subject: RE: [Sip] CallId question
Date: Wed, 18 Feb 2004 11:09:59 -0500
Message-ID: <4A3384433CE2AB46A63468CB207E209DB1A1A0@zoe.office.snowshore.com>
Thread-Topic: [Sip] CallId question
Thread-Index: AcP2OBzmKMw1rZvCTGeSmHUpdzj38AAAWxKQ
From: "Eric Burger" <eburger@snowshore.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: "Anil Bollineni" <ABollineni@netscreen.com>, <sip@ietf.org>
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Good point -- if you are going to be a B2BUA, you have to follow the =
*UA* rules, e.g., generate new, unique Call-Id's.

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Wednesday, February 18, 2004 10:59 AM
> To: Eric Burger
> Cc: Anil Bollineni; sip@ietf.org; Cullen Jennings
> Subject: Re: [Sip] CallId question
>=20
>=20
> I agree that you shouldn't muck with the call-id, but for different=20
> reasons. From 3261:
>=20
>     In a new request created by a UAC outside of any dialog,=20
> the Call-ID
>     header field MUST be selected by the UAC as a globally unique
>     identifier over space and time unless overridden by=20
> method-specific
>     behavior.  All SIP UAs must have a means to guarantee=20
> that the Call-
>     ID header fields they produce will not be inadvertently=20
> generated by
>     any other UA.
>=20
> A UA behind a NAT can't use its IP address in its call-id=20
> because that=20
> won't be guaranteed to be unique. It might be able to use its=20
> DNS name.=20
> If it doesn't have one, then it is still obligated to come up with=20
> something unique. Otherwise it is just broken.
>=20
> 	Paul
>=20
> Eric Burger wrote:
> > Upon reflection, this is an interesting theoretical problem=20
> that statistically is not a problem.
> >=20
> > What is a Call-Id?  It is an opaque, globally unique string=20
> generated by the UAC.  To promote uniqueness, 3261 requires a=20
> string identifying the UAC, either IP address or hostname. =20
> Note that NOTHING in SIP places semantics on the host part of=20
> the Call-Id.
> >=20
> > So, in reality, don't bother to mung the Call-Id.  What is=20
> the likelihood of two UAC's having the following properties=20
> AT THE SAME TIME:
> > 	same randomly generated LHS (unlikely, but possible)
> > 	same IP address (possible)
> > 	same destination (very unlikely)
> >=20
> > IIF all of these conditions are SIMULTANEOUSLY true, then=20
> there is a problem, in that there will be two dialogs with=20
> the same dialog identifier.  You've got bigger problems then=20
> what to change an IP address to :-)
> >=20
> >=20
> > That said, if the reason you are munging the Call-Id host=20
> part is for network hiding, then you really, really, really=20
> should be hacking the user part, as well.  I'll leave the=20
> security considerations section as an exercise for the reader.
> >=20
> > P.S., this whole thing breaks lots of stuff.  I'm sure=20
> Henry will revoke my License to SIP for even explaining some of it :-)
> >=20
> >=20
> > -----Original Message-----
> > From: Cullen Jennings [mailto:fluffy@cisco.com]
> > Sent: Tuesday, February 10, 2004 6:34 PM
> > To: Anil Bollineni; sip@ietf.org
> > Subject: Re: [Sip] CallId question
> >=20
> >=20
> >=20
> > The NAT/FW must not go changing this stuff. If it does, a=20
> S/MIME signed sipfrag will fail. There is no need for the NAT=20
> to go change call IDs
> >=20
> >=20
> >=20
> > On 2/10/04 12:27 PM, "Anil Bollineni"=20
> <ABollineni@netscreen.com> wrote:
> >=20
> >=20
> > Hello,
> > =20
> >             I have a question about CallId. Suppose two=20
> UA's generate callid's putting theire hostnames and send to=20
> firewall/NAT.
> >             UA1:INVITE
> >                  CallID: call1@192.10.10.10
> >             UA2: INVITE
> >                     CallId: call1@192.10.10.12
> >            =20
> >             Because those callid information contain IP=20
> addresses, the firewall/NAT should translate these IP=20
> addresses by putting it's own IP in callId. The callId's are=20
> changed to call1@firewall_ip, call1@firewall_ip. When=20
> responses come back from their respective UAS's the=20
> firewall/NAT translates the it's IP address in callId's to=20
> their original IP's. My question is should I change the=20
> string "call1" to something generated by firewall or Is it OK=20
> to leave like that. Even though they have same CallId, the=20
> proxy should discriminate the calls by from-tag, callid, and to-tag.
> > According to Spec, any UA should have a different means of=20
> generating callid's with those of other UA's.=20
> > =20
> > Anything to do correctly to the above procedure?
> > =20
> > Thanks in advance,
> > Anil
> > This email contains material that is confidential. The=20
> content of this email is for the sole use of the intended=20
> recipient(s). Any review or distribution by persons other=20
> than the intended recipient(s) without the express permission=20
> of NetScreen Technologies, Inc. is strictly prohibited. If=20
> you are not the intended recipient, please contact the sender=20
> and delete/destroy all copies of this email and any related=20
> attachments. NetScreen does not guarantee the accuracy or=20
> completeness of third party materials or information.
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
>=20


_______________________________________________
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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  Wed Feb 18 15:19:46 2004
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From: Robert Sparks <rsparks@dynamicsoft.com>
To: sip@ietf.org
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Subject: [Sip] Moving forward on non-invite issues
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All -

At the request of the chairs, I split the non-invite
draft into three parts :

- the description of the problems
- the things we seem to have consensus to fix now
- the things we might do in the future but need to
  argue over more.


The resulting drafts are available in the repository:
http://www.ietf.org/internet-drafts/draft-sparks-sip-nit-problems-00.txt
http://www.ietf.org/internet-drafts/draft-sparks-sip-nit-actions-00.txt
http://www.ietf.org/internet-drafts/draft-sparks-sip-nit-future-00.txt


My recommendation is to LC the problems and actions drafts
as soon as it is feasible to do so and send them to the IESG.

RjS

ps - if you prefer reading xml2rfc's html output, you
can get that at:
http://www.nostrum.com/~rjsparks/draft-sparks-sip-nit-problems-00.html
http://www.nostrum.com/~rjsparks/draft-sparks-sip-nit-actions-00.html
http://www.nostrum.com/~rjsparks/draft-sparks-sip-nit-future-00.html


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	Wed, 18 Feb 2004 14:27:51 -0600
Subject: RE: [Sip] CallId question
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Eric Burger <eburger@snowshore.com>
Cc: Paul Kyzivat <pkyzivat@cisco.com>,
        Anil Bollineni <ABollineni@netscreen.com>, sip@ietf.org
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	 <4A3384433CE2AB46A63468CB207E209DB1A1A0@zoe.office.snowshore.com>
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I would like to see us revisit the BNF for Call-ID at
some point and explicitly make its value a single bundle
of opaque bits.

It almost is already - it is defined as word @ word
(NOT word @ host). 

There are implementations that have been mislead by
the recommendation to use a host for the second
word (as a source of probably-unique bits) into
_parsing_ that field from received messages as 
a host (as defined in the BNF in 3261), and 
rejecting messages (or breaking) when the bits
they found didn't line up.

I'm willing to bet that a very large subset of
the existing implementations in the world would
not deal with two "@" signs show up gracefully.
(Something to add to the torture tests maybe).

We can make this less likely to occur in the future.

RjS

On Wed, 2004-02-18 at 10:09, Eric Burger wrote:
> Good point -- if you are going to be a B2BUA, you have to follow the *UA* rules, e.g., generate new, unique Call-Id's.
> 
> > -----Original Message-----
> > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > Sent: Wednesday, February 18, 2004 10:59 AM
> > To: Eric Burger
> > Cc: Anil Bollineni; sip@ietf.org; Cullen Jennings
> > Subject: Re: [Sip] CallId question
> > 
> > 
> > I agree that you shouldn't muck with the call-id, but for different 
> > reasons. From 3261:
> > 
> >     In a new request created by a UAC outside of any dialog, 
> > the Call-ID
> >     header field MUST be selected by the UAC as a globally unique
> >     identifier over space and time unless overridden by 
> > method-specific
> >     behavior.  All SIP UAs must have a means to guarantee 
> > that the Call-
> >     ID header fields they produce will not be inadvertently 
> > generated by
> >     any other UA.
> > 
> > A UA behind a NAT can't use its IP address in its call-id 
> > because that 
> > won't be guaranteed to be unique. It might be able to use its 
> > DNS name. 
> > If it doesn't have one, then it is still obligated to come up with 
> > something unique. Otherwise it is just broken.
> > 
> > 	Paul
> > 
> > Eric Burger wrote:
> > > Upon reflection, this is an interesting theoretical problem 
> > that statistically is not a problem.
> > > 
> > > What is a Call-Id?  It is an opaque, globally unique string 
> > generated by the UAC.  To promote uniqueness, 3261 requires a 
> > string identifying the UAC, either IP address or hostname.  
> > Note that NOTHING in SIP places semantics on the host part of 
> > the Call-Id.
> > > 
> > > So, in reality, don't bother to mung the Call-Id.  What is 
> > the likelihood of two UAC's having the following properties 
> > AT THE SAME TIME:
> > > 	same randomly generated LHS (unlikely, but possible)
> > > 	same IP address (possible)
> > > 	same destination (very unlikely)
> > > 
> > > IIF all of these conditions are SIMULTANEOUSLY true, then 
> > there is a problem, in that there will be two dialogs with 
> > the same dialog identifier.  You've got bigger problems then 
> > what to change an IP address to :-)
> > > 
> > > 
> > > That said, if the reason you are munging the Call-Id host 
> > part is for network hiding, then you really, really, really 
> > should be hacking the user part, as well.  I'll leave the 
> > security considerations section as an exercise for the reader.
> > > 
> > > P.S., this whole thing breaks lots of stuff.  I'm sure 
> > Henry will revoke my License to SIP for even explaining some of it :-)
> > > 
> > > 
> > > -----Original Message-----
> > > From: Cullen Jennings [mailto:fluffy@cisco.com]
> > > Sent: Tuesday, February 10, 2004 6:34 PM
> > > To: Anil Bollineni; sip@ietf.org
> > > Subject: Re: [Sip] CallId question
> > > 
> > > 
> > > 
> > > The NAT/FW must not go changing this stuff. If it does, a 
> > S/MIME signed sipfrag will fail. There is no need for the NAT 
> > to go change call IDs
> > > 
> > > 
> > > 
> > > On 2/10/04 12:27 PM, "Anil Bollineni" 
> > <ABollineni@netscreen.com> wrote:
> > > 
> > > 
> > > Hello,
> > >  
> > >             I have a question about CallId. Suppose two 
> > UA's generate callid's putting theire hostnames and send to 
> > firewall/NAT.
> > >             UA1:INVITE
> > >                  CallID: call1@192.10.10.10
> > >             UA2: INVITE
> > >                     CallId: call1@192.10.10.12
> > >             
> > >             Because those callid information contain IP 
> > addresses, the firewall/NAT should translate these IP 
> > addresses by putting it's own IP in callId. The callId's are 
> > changed to call1@firewall_ip, call1@firewall_ip. When 
> > responses come back from their respective UAS's the 
> > firewall/NAT translates the it's IP address in callId's to 
> > their original IP's. My question is should I change the 
> > string "call1" to something generated by firewall or Is it OK 
> > to leave like that. Even though they have same CallId, the 
> > proxy should discriminate the calls by from-tag, callid, and to-tag.
> > > According to Spec, any UA should have a different means of 
> > generating callid's with those of other UA's. 
> > >  
> > > Anything to do correctly to the above procedure?
> > >  
> > > Thanks in advance,
> > > Anil
> > > This email contains material that is confidential. The 
> > content of this email is for the sole use of the intended 
> > recipient(s). Any review or distribution by persons other 
> > than the intended recipient(s) without the express permission 
> > of NetScreen Technologies, Inc. is strictly prohibited. If 
> > you are not the intended recipient, please contact the sender 
> > and delete/destroy all copies of this email and any related 
> > attachments. NetScreen does not guarantee the accuracy or 
> > completeness of third party materials or information.
> > > 
> > > 
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > > 
> > 
> > 
> > 
> 
> 
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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Date: Wed, 18 Feb 2004 14:50:00 -0600
From: Alex Audu <alex.audu@alcatel.com>
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To: Henning Schulzrinne <hgs@cs.columbia.edu>
CC: sip@ietf.org, jmpolk@cisco.com
Subject: [Sip] draft-ietf-sip-resource-priority-02 - behavior
References: <402FF85F.4020202@cs.columbia.edu> <40325ADF.30508@cisco.com> <4032A0B2.50504@cs.columbia.edu>
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Subsection 4.1 (General Rules)  gave an example behavior of a SIP 
element supporting this
specification as:

  ... "For example, for SUBSCRIBE, a higher-priority request
   may get preferential treatment if storage or bandwidth for
   notifications are scarce, possibly displacing a lower-priority
   subscription."

Is this suggesting that resources allocated to a lower priority request
can be reclaimed for a higher priority one (say in the case of scarce
storage space)?  If that is the case, how are the clients of the lower
priority request notified of their misfortune?

Regards,
Alex.




Henning Schulzrinne wrote:

>
>> - Now that Resource-Priority is optional in all requests, shouldn't
>> Accept-Resource-Priority be permitted in 420 responses to all the same
>> messages?
>
>
> Agreed; except for ACK...
>
>>
>> - Shouldn't the R-P be optional in responses to any request where it was
>> permitted?
>
>
> Same quick cut-and-paste error; yes.
>
>>
>> - If A-R-P is mandatory in 417 responses to SUB, NOT, and MSG, shouldn't
>> it also be mandatory (rather than optional) in 417 response to INVITE?
>> And by logic just above, shouldn't it be mandatory in responses to any
>> message where R-P is optional?
>>
>> - So I think the table probably ought to look like:
>>
>>     Header field             where proxy INV ACK CAN BYE REG OPT PRA
>>     ----------------------------------------------------------------
>>     Resource-Priority        R     amd    o   o   o   o   o   o   o
>>     Resource-Priority        200   -      o   o   o   o   o   o   o
>>     Accept-Resource-Priority 200   -      o   o   o   o   o   o   o
>>     Accept-Resource-Priority 417   -      m   m   m   m   m   m   m
>>     Accept-Resource-Priority 420   -      o   o   o   o   o   o   o
>>
>>     Header field             where proxy SUB NOT UPD MSG REF INF PUB
>>     ----------------------------------------------------------------
>>     Resource-Priority        R     amd    o   o   o   o   o   o   o
>>     Resource-Priority        200   -      o   o   o   o   o   o   o
>>     Accept-Resource-Priority 200   -      o   o   o   o   o   o   o
>>     Accept-Resource-Priority 417   -      m   m   m   m   m   m   m
>>     Accept-Resource-Priority 420   -      o   o   o   o   o   o   o
>
>
> Yup, minus the responses for ACK.
>
>>
>> In section 4.1, why do you single out INVITE, MESSAGE, UPDATE, SUBSCRIBE
>> and NOTIFY, but omit ACK and PRACK? If INVITE is important, then ACK and
>> PRACK must be important too. This probably isn't an issue in a UAS,
>> because it considers those to be part of the INVITE. But it could make a
>> difference in a proxy.
>>
>>     Paul
>
>
> _______________________________________________
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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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Subject: Re: [Sip] draft-ietf-sip-resource-priority-02 - behavior
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SUBSCRIBE is never a guarantee or contract of delivery. The notifier can 
decide, for any reason whatsoever, to stop delivering notifications at 
any time. This is just one more, along with "I don't feel like it any 
more". It's unfortunate, but priority always involves stiffing the 
little guy in some way.

(The more likely behavior is that a system that is bandwidth-constrained 
might prioritize notifications resulting from a PUBLISH. It simply sends 
notifications in order of priority until the next PUBLISH arrives; the 
ones low on the totem pole might never get notified.)

Alex Audu wrote:

> Subsection 4.1 (General Rules)  gave an example behavior of a SIP 
> element supporting this
> specification as:
> 
>  ... "For example, for SUBSCRIBE, a higher-priority request
>   may get preferential treatment if storage or bandwidth for
>   notifications are scarce, possibly displacing a lower-priority
>   subscription."
> 
> Is this suggesting that resources allocated to a lower priority request
> can be reclaimed for a higher priority one (say in the case of scarce
> storage space)?  If that is the case, how are the clients of the lower
> priority request notified of their misfortune?
> 
> Regards,
> Alex.


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Subject: RE: [Sip] CallId question
Date: Wed, 18 Feb 2004 16:13:17 -0500
Message-ID: <4A3384433CE2AB46A63468CB207E209DB1A1AA@zoe.office.snowshore.com>
Thread-Topic: [Sip] CallId question
Thread-Index: AcP2XbNm60e0A+AeTAKXhuGCuTTjyQABTBdA
From: "Eric Burger" <eburger@snowshore.com>
To: "Robert Sparks" <rsparks@dynamicsoft.com>
Cc: "Paul Kyzivat" <pkyzivat@cisco.com>,
        "Anil Bollineni" <ABollineni@netscreen.com>, <sip@ietf.org>
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I concur (make it token).

> -----Original Message-----
> From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
> Sent: Wednesday, February 18, 2004 3:28 PM
> To: Eric Burger
> Cc: Paul Kyzivat; Anil Bollineni; sip@ietf.org
> Subject: RE: [Sip] CallId question
>=20
>=20
> I would like to see us revisit the BNF for Call-ID at
> some point and explicitly make its value a single bundle
> of opaque bits.
>=20
> It almost is already - it is defined as word @ word
> (NOT word @ host).=20
>=20
> There are implementations that have been mislead by
> the recommendation to use a host for the second
> word (as a source of probably-unique bits) into
> _parsing_ that field from received messages as=20
> a host (as defined in the BNF in 3261), and=20
> rejecting messages (or breaking) when the bits
> they found didn't line up.
>=20
> I'm willing to bet that a very large subset of
> the existing implementations in the world would
> not deal with two "@" signs show up gracefully.
> (Something to add to the torture tests maybe).
>=20
> We can make this less likely to occur in the future.
>=20
> RjS
>=20
> On Wed, 2004-02-18 at 10:09, Eric Burger wrote:
> > Good point -- if you are going to be a B2BUA, you have to=20
> follow the *UA* rules, e.g., generate new, unique Call-Id's.
> >=20
> > > -----Original Message-----
> > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > > Sent: Wednesday, February 18, 2004 10:59 AM
> > > To: Eric Burger
> > > Cc: Anil Bollineni; sip@ietf.org; Cullen Jennings
> > > Subject: Re: [Sip] CallId question
> > >=20
> > >=20
> > > I agree that you shouldn't muck with the call-id, but for=20
> different=20
> > > reasons. From 3261:
> > >=20
> > >     In a new request created by a UAC outside of any dialog,=20
> > > the Call-ID
> > >     header field MUST be selected by the UAC as a globally unique
> > >     identifier over space and time unless overridden by=20
> > > method-specific
> > >     behavior.  All SIP UAs must have a means to guarantee=20
> > > that the Call-
> > >     ID header fields they produce will not be inadvertently=20
> > > generated by
> > >     any other UA.
> > >=20
> > > A UA behind a NAT can't use its IP address in its call-id=20
> > > because that=20
> > > won't be guaranteed to be unique. It might be able to use its=20
> > > DNS name.=20
> > > If it doesn't have one, then it is still obligated to=20
> come up with=20
> > > something unique. Otherwise it is just broken.
> > >=20
> > > 	Paul
> > >=20
> > > Eric Burger wrote:
> > > > Upon reflection, this is an interesting theoretical problem=20
> > > that statistically is not a problem.
> > > >=20
> > > > What is a Call-Id?  It is an opaque, globally unique string=20
> > > generated by the UAC.  To promote uniqueness, 3261 requires a=20
> > > string identifying the UAC, either IP address or hostname. =20
> > > Note that NOTHING in SIP places semantics on the host part of=20
> > > the Call-Id.
> > > >=20
> > > > So, in reality, don't bother to mung the Call-Id.  What is=20
> > > the likelihood of two UAC's having the following properties=20
> > > AT THE SAME TIME:
> > > > 	same randomly generated LHS (unlikely, but possible)
> > > > 	same IP address (possible)
> > > > 	same destination (very unlikely)
> > > >=20
> > > > IIF all of these conditions are SIMULTANEOUSLY true, then=20
> > > there is a problem, in that there will be two dialogs with=20
> > > the same dialog identifier.  You've got bigger problems then=20
> > > what to change an IP address to :-)
> > > >=20
> > > >=20
> > > > That said, if the reason you are munging the Call-Id host=20
> > > part is for network hiding, then you really, really, really=20
> > > should be hacking the user part, as well.  I'll leave the=20
> > > security considerations section as an exercise for the reader.
> > > >=20
> > > > P.S., this whole thing breaks lots of stuff.  I'm sure=20
> > > Henry will revoke my License to SIP for even explaining=20
> some of it :-)
> > > >=20
> > > >=20
> > > > -----Original Message-----
> > > > From: Cullen Jennings [mailto:fluffy@cisco.com]
> > > > Sent: Tuesday, February 10, 2004 6:34 PM
> > > > To: Anil Bollineni; sip@ietf.org
> > > > Subject: Re: [Sip] CallId question
> > > >=20
> > > >=20
> > > >=20
> > > > The NAT/FW must not go changing this stuff. If it does, a=20
> > > S/MIME signed sipfrag will fail. There is no need for the NAT=20
> > > to go change call IDs
> > > >=20
> > > >=20
> > > >=20
> > > > On 2/10/04 12:27 PM, "Anil Bollineni"=20
> > > <ABollineni@netscreen.com> wrote:
> > > >=20
> > > >=20
> > > > Hello,
> > > > =20
> > > >             I have a question about CallId. Suppose two=20
> > > UA's generate callid's putting theire hostnames and send to=20
> > > firewall/NAT.
> > > >             UA1:INVITE
> > > >                  CallID: call1@192.10.10.10
> > > >             UA2: INVITE
> > > >                     CallId: call1@192.10.10.12
> > > >            =20
> > > >             Because those callid information contain IP=20
> > > addresses, the firewall/NAT should translate these IP=20
> > > addresses by putting it's own IP in callId. The callId's are=20
> > > changed to call1@firewall_ip, call1@firewall_ip. When=20
> > > responses come back from their respective UAS's the=20
> > > firewall/NAT translates the it's IP address in callId's to=20
> > > their original IP's. My question is should I change the=20
> > > string "call1" to something generated by firewall or Is it OK=20
> > > to leave like that. Even though they have same CallId, the=20
> > > proxy should discriminate the calls by from-tag, callid,=20
> and to-tag.
> > > > According to Spec, any UA should have a different means of=20
> > > generating callid's with those of other UA's.=20
> > > > =20
> > > > Anything to do correctly to the above procedure?
> > > > =20
> > > > Thanks in advance,
> > > > Anil
> > > > This email contains material that is confidential. The=20
> > > content of this email is for the sole use of the intended=20
> > > recipient(s). Any review or distribution by persons other=20
> > > than the intended recipient(s) without the express permission=20
> > > of NetScreen Technologies, Inc. is strictly prohibited. If=20
> > > you are not the intended recipient, please contact the sender=20
> > > and delete/destroy all copies of this email and any related=20
> > > attachments. NetScreen does not guarantee the accuracy or=20
> > > completeness of third party materials or information.
> > > >=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=20
> current sip
> > > > Use sipping@ietf.org for new developments on the=20
> application of sip
> > > >=20
> > >=20
> > >=20
> > >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>=20
>=20
>=20


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



From exim@www1.ietf.org  Wed Feb 18 16:52:40 2004
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Date: Wed, 18 Feb 2004 16:49:14 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Eric Burger <eburger@snowshore.com>
CC: Robert Sparks <rsparks@dynamicsoft.com>,
        Anil Bollineni <ABollineni@netscreen.com>, sip@ietf.org
Subject: Re: [Sip] CallId question
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Eric,

You can't make it a token an remain backward compatible with 3261.

I support tightening up the semantics. But you can't just say it must be 
  a globally unique value in space and time without giving some guidance 
on how to partition the namespace so that developers don't pick 
conflicting techniques.

Even changing to word@host would be slightly incompatible. But perhaps 
it could be specified that the form word@host should not be used unless 
the application generating the callid can assure that no other entity is 
using the same value of host to generate callids.

	Paul

Eric Burger wrote:
> I concur (make it token).
> 
> 
>>-----Original Message-----
>>From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
>>Sent: Wednesday, February 18, 2004 3:28 PM
>>To: Eric Burger
>>Cc: Paul Kyzivat; Anil Bollineni; sip@ietf.org
>>Subject: RE: [Sip] CallId question
>>
>>
>>I would like to see us revisit the BNF for Call-ID at
>>some point and explicitly make its value a single bundle
>>of opaque bits.
>>
>>It almost is already - it is defined as word @ word
>>(NOT word @ host). 
>>
>>There are implementations that have been mislead by
>>the recommendation to use a host for the second
>>word (as a source of probably-unique bits) into
>>_parsing_ that field from received messages as 
>>a host (as defined in the BNF in 3261), and 
>>rejecting messages (or breaking) when the bits
>>they found didn't line up.
>>
>>I'm willing to bet that a very large subset of
>>the existing implementations in the world would
>>not deal with two "@" signs show up gracefully.
>>(Something to add to the torture tests maybe).
>>
>>We can make this less likely to occur in the future.
>>
>>RjS
>>
>>On Wed, 2004-02-18 at 10:09, Eric Burger wrote:
>>
>>>Good point -- if you are going to be a B2BUA, you have to 
>>
>>follow the *UA* rules, e.g., generate new, unique Call-Id's.
>>
>>>>-----Original Message-----
>>>>From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>>Sent: Wednesday, February 18, 2004 10:59 AM
>>>>To: Eric Burger
>>>>Cc: Anil Bollineni; sip@ietf.org; Cullen Jennings
>>>>Subject: Re: [Sip] CallId question
>>>>
>>>>
>>>>I agree that you shouldn't muck with the call-id, but for 
>>>
>>different 
>>
>>>>reasons. From 3261:
>>>>
>>>>    In a new request created by a UAC outside of any dialog, 
>>>>the Call-ID
>>>>    header field MUST be selected by the UAC as a globally unique
>>>>    identifier over space and time unless overridden by 
>>>>method-specific
>>>>    behavior.  All SIP UAs must have a means to guarantee 
>>>>that the Call-
>>>>    ID header fields they produce will not be inadvertently 
>>>>generated by
>>>>    any other UA.
>>>>
>>>>A UA behind a NAT can't use its IP address in its call-id 
>>>>because that 
>>>>won't be guaranteed to be unique. It might be able to use its 
>>>>DNS name. 
>>>>If it doesn't have one, then it is still obligated to 
>>>
>>come up with 
>>
>>>>something unique. Otherwise it is just broken.
>>>>
>>>>	Paul
>>>>
>>>>Eric Burger wrote:
>>>>
>>>>>Upon reflection, this is an interesting theoretical problem 
>>>>
>>>>that statistically is not a problem.
>>>>
>>>>>What is a Call-Id?  It is an opaque, globally unique string 
>>>>
>>>>generated by the UAC.  To promote uniqueness, 3261 requires a 
>>>>string identifying the UAC, either IP address or hostname.  
>>>>Note that NOTHING in SIP places semantics on the host part of 
>>>>the Call-Id.
>>>>
>>>>>So, in reality, don't bother to mung the Call-Id.  What is 
>>>>
>>>>the likelihood of two UAC's having the following properties 
>>>>AT THE SAME TIME:
>>>>
>>>>>	same randomly generated LHS (unlikely, but possible)
>>>>>	same IP address (possible)
>>>>>	same destination (very unlikely)
>>>>>
>>>>>IIF all of these conditions are SIMULTANEOUSLY true, then 
>>>>
>>>>there is a problem, in that there will be two dialogs with 
>>>>the same dialog identifier.  You've got bigger problems then 
>>>>what to change an IP address to :-)
>>>>
>>>>>
>>>>>That said, if the reason you are munging the Call-Id host 
>>>>
>>>>part is for network hiding, then you really, really, really 
>>>>should be hacking the user part, as well.  I'll leave the 
>>>>security considerations section as an exercise for the reader.
>>>>
>>>>>P.S., this whole thing breaks lots of stuff.  I'm sure 
>>>>
>>>>Henry will revoke my License to SIP for even explaining 
>>>
>>some of it :-)
>>
>>>>>
>>>>>-----Original Message-----
>>>>>From: Cullen Jennings [mailto:fluffy@cisco.com]
>>>>>Sent: Tuesday, February 10, 2004 6:34 PM
>>>>>To: Anil Bollineni; sip@ietf.org
>>>>>Subject: Re: [Sip] CallId question
>>>>>
>>>>>
>>>>>
>>>>>The NAT/FW must not go changing this stuff. If it does, a 
>>>>
>>>>S/MIME signed sipfrag will fail. There is no need for the NAT 
>>>>to go change call IDs
>>>>
>>>>>
>>>>>
>>>>>On 2/10/04 12:27 PM, "Anil Bollineni" 
>>>>
>>>><ABollineni@netscreen.com> wrote:
>>>>
>>>>>
>>>>>Hello,
>>>>> 
>>>>>            I have a question about CallId. Suppose two 
>>>>
>>>>UA's generate callid's putting theire hostnames and send to 
>>>>firewall/NAT.
>>>>
>>>>>            UA1:INVITE
>>>>>                 CallID: call1@192.10.10.10
>>>>>            UA2: INVITE
>>>>>                    CallId: call1@192.10.10.12
>>>>>            
>>>>>            Because those callid information contain IP 
>>>>
>>>>addresses, the firewall/NAT should translate these IP 
>>>>addresses by putting it's own IP in callId. The callId's are 
>>>>changed to call1@firewall_ip, call1@firewall_ip. When 
>>>>responses come back from their respective UAS's the 
>>>>firewall/NAT translates the it's IP address in callId's to 
>>>>their original IP's. My question is should I change the 
>>>>string "call1" to something generated by firewall or Is it OK 
>>>>to leave like that. Even though they have same CallId, the 
>>>>proxy should discriminate the calls by from-tag, callid, 
>>>
>>and to-tag.
>>
>>>>>According to Spec, any UA should have a different means of 
>>>>
>>>>generating callid's with those of other UA's. 
>>>>
>>>>> 
>>>>>Anything to do correctly to the above procedure?
>>>>> 
>>>>>Thanks in advance,
>>>>>Anil
>>>>>This email contains material that is confidential. The 
>>>>
>>>>content of this email is for the sole use of the intended 
>>>>recipient(s). Any review or distribution by persons other 
>>>>than the intended recipient(s) without the express permission 
>>>>of NetScreen Technologies, Inc. is strictly prohibited. If 
>>>>you are not the intended recipient, please contact the sender 
>>>>and delete/destroy all copies of this email and any related 
>>>>attachments. NetScreen does not guarantee the accuracy or 
>>>>completeness of third party materials or information.
>>>>
>>>>>
>>>>>_______________________________________________
>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>This list is for NEW development of the core SIP 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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Cc: "Anil Bollineni" <ABollineni@netscreen.com>, <sip@ietf.org>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sip] Changing Via In NAT
Date: Wed, 18 Feb 2004 13:53:16 -0800
To: "Eric Burger" <eburger@snowshore.com>
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There is one critical difference between an ALG and a SIP proxy or =20
B2BUA.

ALGs don't insert Via headers, Contact, Record-Route, Route, etc..  =20
They just rewrite stuff going by on the wire, assuming that packets =20
will happen to go through this particular firewall of NAT on their way =20=

to the "outside".

SIP intermediaries insert some sort of SIP identifier so that requests =20=

and responses get routed to them.

In this way, ALGs are much, much worse.

thanks,
-rohan


On Feb 18, 2004, at 7:44 AM, Eric Burger wrote:

> An ALG is an ALG: a B2BUA.
> =A0
> Do whatever you want.=A0 It is not subject to standardization.=A0 It =
is =20
> not a good idea.=A0 Do so at your own peril.
> -----Original Message-----
> From: Anil Bollineni [mailto:ABollineni@netscreen.com]
> Sent: Friday, February 13, 2004 2:08 PM
> To: 'sip@ietf.org'
> Subject: [Sip] Changing Via In NAT
>
>
> Dear All,
>
> =A0
>
> In draft-ietf-sip-nat-01.txt, there are SIP extensions for NAT =20
> traversal. The received and rport parameter are used to send back the =20=

> responses correctly. However if we don't use these extensions, and =20
> change the via ip address and port with that of firewall/port then the =
=20
> SIP responses will come directly to firewall, in that case the =20
> firewall permits the responses. UA1 inside the network sends requests =20=

> to firewall/NAT.
>
> =A0
>
> UA1
>
>  INVITE sip:user@domain.com SIP/2.0 --------------=E0 Firewall/NAT =20
> =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 =
---------------------------------------------=E0UA2
>
> Via: SIP/2.0/UDP 10.1.1.1:4540=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 INVITE =20
> sip:user@domain.com SIP/2.0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0
>
>  Via: SIP/2.0/UDP firewall_ip:5060
>
> =
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=20
>  (Maintain mapping)
>
>  =A0
>
> the UA2 will send the responses to firewall/NAT and firewall/NAT will =20=

> write back the addresses in Via to original IP/port. I understand from =
=20
> the draft, if there are no SIP ALG functionality in firewall, then the =
=20
> extensions are used to make SIP traverse through NAT. However if we =20=

> implement SIP ALG in firewall then we can make SIP traverse through =20=

> NAT. Can anybody tell that this way is per SIP spec.
>
> =A0
>
> Thank you,
>
> Anil
>
> =A0
>
> This email contains material that is confidential. The content of this =
=20
> email is for the sole use of the intended recipient(s). Any review or =20=

> distribution by persons other than the intended recipient(s) without =20=

> the express permission of NetScreen Technologies, Inc. is strictly =20
> prohibited. If you are not the intended recipient, please contact the =20=

> sender and delete/destroy all copies of this email and any related =20
> attachments. NetScreen does not guarantee the accuracy or completeness =
=20
> of third party materials or information.
>
> =A0

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There is one critical difference between an ALG and a SIP proxy or
B2BUA.


ALGs don't insert Via headers, Contact, Record-Route, Route, etc..=20
They just rewrite stuff going by on the wire, assuming that packets
will happen to go through this particular firewall of NAT on their way
to the "outside".


SIP intermediaries insert some sort of SIP identifier so that requests
and responses get routed to them.


In this way, ALGs are much, much worse.


thanks,

-rohan



On Feb 18, 2004, at 7:44 AM, Eric Burger wrote:


=
<excerpt><fontfamily><param>Arial</param><color><param>0000,0000,FFFF</par=
am><smaller>An
ALG is an ALG: a B2BUA.</smaller></color></fontfamily>

=A0

=
<fontfamily><param>Arial</param><color><param>0000,0000,FFFF</param><small=
er>Do
whatever you want.=A0 It is not subject to standardization.=A0 It is not =
a
good idea.=A0 Do so at your own peril.</smaller></color></fontfamily>

<fontfamily><param>Tahoma</param><smaller>-----Original =
Message-----</smaller></fontfamily>

=
<bold><fontfamily><param>Tahoma</param><smaller>From:</smaller></fontfamil=
y></bold><fontfamily><param>Tahoma</param><smaller>
Anil Bollineni [mailto:ABollineni@netscreen.com]</smaller></fontfamily>

=
<bold><fontfamily><param>Tahoma</param><smaller>Sent:</smaller></fontfamil=
y></bold><fontfamily><param>Tahoma</param><smaller>
Friday, February 13, 2004 2:08 PM</smaller></fontfamily>

=
<bold><fontfamily><param>Tahoma</param><smaller>To:</smaller></fontfamily>=
</bold><fontfamily><param>Tahoma</param><smaller>
'sip@ietf.org'</smaller></fontfamily>

=
<bold><fontfamily><param>Tahoma</param><smaller>Subject:</smaller></fontfa=
mily></bold><fontfamily><param>Tahoma</param><smaller>
[Sip] Changing Via In NAT</smaller></fontfamily>



<fontfamily><param>Arial</param><x-tad-bigger>Dear =
All,</x-tad-bigger></fontfamily>


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


<fontfamily><param>Arial</param><x-tad-bigger>In
draft-ietf-sip-nat-01.txt, there are SIP extensions for NAT traversal.
The received and rport parameter are used to send back the responses
correctly. However if we don't use these extensions, and change the
via ip address and port with that of firewall/port then the SIP
responses will come directly to firewall, in that case the firewall
permits the responses. UA1 inside the network sends requests to
firewall/NAT.</x-tad-bigger></fontfamily>


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


=
<fontfamily><param>Arial</param><x-tad-bigger>UA1</x-tad-bigger></fontfami=
ly>


<fontfamily><param>Arial</param><x-tad-bigger> INVITE
sip:user@domain.com SIP/2.0
=
--------------</x-tad-bigger></fontfamily><x-tad-bigger>=E0</x-tad-bigger>=
<fontfamily><param>Arial</param><x-tad-bigger>
Firewall/NAT =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0
=
---------------------------------------------</x-tad-bigger></fontfamily><=
x-tad-bigger>=E0</x-tad-bigger><fontfamily><param>Arial</param><x-tad-bigg=
er>UA2</x-tad-bigger></fontfamily>


<fontfamily><param>Arial</param><x-tad-bigger>Via: SIP/2.0/UDP
10.1.1.1:4540=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0 INVITE sip:user@domain.com
SIP/2.0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0</x-tad-bigger></fontfamily>


<fontfamily><param>Arial</param><x-tad-bigger> Via: SIP/2.0/UDP
firewall_ip:5060</x-tad-bigger></fontfamily>


=
<fontfamily><param>Arial</param><x-tad-bigger>=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0
(Maintain mapping)</x-tad-bigger></fontfamily>


<fontfamily><param>Arial</param><x-tad-bigger>
</x-tad-bigger></fontfamily><fontfamily><param>Times New =
Roman</param><bigger><bigger>=A0</bigger></bigger></fontfamily>


<fontfamily><param>Arial</param><x-tad-bigger>the UA2 will send the
responses to firewall/NAT and firewall/NAT will write back the
addresses in Via to original IP/port. I understand from the draft, if
there are no SIP ALG functionality in firewall, then the extensions
are used to make SIP traverse through NAT. However if we implement SIP
ALG in firewall then we can make SIP traverse through NAT. Can anybody
tell that this way is per SIP spec.</x-tad-bigger></fontfamily>


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


<fontfamily><param>Arial</param><x-tad-bigger>Thank =
you,</x-tad-bigger></fontfamily>


=
<fontfamily><param>Arial</param><x-tad-bigger>Anil</x-tad-bigger></fontfam=
ily>


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


<fontfamily><param>Arial</param><x-tad-bigger>This email contains
material that is confidential. The content of this email is for the
sole use of the intended recipient(s). Any review or distribution by
persons other than the intended recipient(s) without the express
permission of NetScreen Technologies, Inc. is strictly prohibited. If
you are not the intended recipient, please contact the sender and
delete/destroy all copies of this email and any related attachments.
NetScreen does not guarantee the accuracy or completeness of third
party materials or information.</x-tad-bigger></fontfamily>


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

</excerpt>=

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From: Anil Bollineni <ABollineni@netscreen.com>
To: "'Rohan Mahy'" <rohan@cisco.com>
Cc: "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] Changing Via In NAT
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Hi Rohan,

=20

            Thanks for pointing me to right RFC. I am thinking that how =
many
UA's correctly support this STUN protocol. I am thinking that we need =
to add
SIP ALG in NAT that if we want to support UA's that don't support any =
SIP
NAT extensions and STUN protocol. I thought that this supporting all =
SIP
aware functionality in ALG is difficult as changes made to SIP protocol =
need
to accommodate in SIP ALG.=20

=20

Thanks,

Anil

=20

This email contains material that is confidential. The content of this =
email
is for the sole use of the intended recipient(s). Any review or =
distribution
by persons other than the intended recipient(s) without the express
permission of NetScreen Technologies, Inc. is strictly prohibited. If =
you
are not the intended recipient, please contact the sender and =
delete/destroy
all copies of this email and any related attachments. NetScreen does =
not
guarantee the accuracy or completeness of third party materials or
information.

-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]=20
Sent: Wednesday, February 18, 2004 4:46 AM
To: Anil Bollineni
Cc: 'sip@ietf.org'
Subject: Re: [Sip] Changing Via In NAT

=20

=20

On Feb 13, 2004, at 11:07 AM, Anil Bollineni wrote:=20

=20

Dear All,=20

=20

In draft-ietf-sip-nat-01.txt, there are SIP extensions for NAT =
traversal.
The received and rport parameter are used to send back the responses
correctly.=20

=20

This is an extremely old reference. The name was changed to
draft-ietf-sip-symmetric-response, and is now published as RFC 3581.=20

=20

However if we don't use these extensions, and change the via ip address =
and
port with that of firewall/port then the SIP responses will come =
directly to
firewall, in that case the firewall permits the responses. UA1 inside =
the
network sends requests to firewall/NAT.=20

=20

UA1=20

=20

INVITE sip:user@domain.com SIP/2.0 --------------=E0 Firewall/NAT
---------------------------------------------=E0UA2=20

=20

Via: SIP/2.0/UDP 10.1.1.1:4540                          INVITE
sip:user@domain.com SIP/2.0                            =20

=20

Via: SIP/2.0/UDP firewall_ip:5060=20

=20

=20
(Maintain mapping)=20

=20

the UA2 will send the responses to firewall/NAT and firewall/NAT will =
write
back the addresses in Via to original IP/port. I understand from the =
draft,
if there are no SIP ALG functionality in firewall, then the extensions =
are
used to make SIP traverse through NAT. However if we implement SIP ALG =
in
firewall then we can make SIP traverse through NAT. Can anybody tell =
that
this way is per SIP spec.=20

=20

The experience of the working group has shown that implementing a SIP =
ALG in
a firewall is "a bad thing". I have personal experience with the =
failings of
SIP ALGs at my own company, not the least of which is that they don't =
work
at all when SIP is integrity-protected using TLS.=20

=20

Many UAs now support RFC 3581 and can use the STUN protocol to get =
valid
public addresses and ports to advertise in SDP for their RTP traffic. =
This
has been relatively painless compared to working with the shortcomings =
of
ALGs.=20

=20

thanks,=20

-rohan=20

=20


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Hi Rohan,</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'>=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 =
Thanks for pointing me to right
RFC. I am thinking that how many UA's correctly support this STUN =
protocol.
I am thinking that we need to add SIP ALG in NAT that if we want to =
support UA's
that don't support any SIP NAT extensions and STUN protocol. I thought
that this supporting all SIP aware functionality in ALG is difficult as =
changes
made to SIP protocol need to accommodate in SIP </span></font><font =
size=3D2
  color=3Dnavy face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
  color:navy'>ALG.</span></font><font size=3D2 color=3Dnavy =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'> =
</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'>Thanks,</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'>Anil</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><font size=3D2 color=3Dnavy face=3DArial><span =
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<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'>-----Original =
Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> Rohan Mahy
[mailto:rohan@cisco.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, =
February 18, 2004
4:46 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> </span></font><font =
size=3D2
 face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>Anil =
Bollineni</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'>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'>Subject:</span></b> Re: [Sip] =
Changing Via In
NAT</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-right:0in;margin-bottom:12.0pt;margin-left:
.5in'><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<p style=3D'margin-left:.5in'><font size=3D3 face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt'>On Feb 13, 2004, at 11:07 AM, Anil Bollineni =
wrote: </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 style=3D'margin-left:.5in'><font size=3D3 face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial'>Dear All,</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 style=3D'margin-left:.5in'><font size=3D3 face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial'>In draft-ietf-sip-nat-01.txt, there are SIP
extensions for NAT traversal. The received and rport parameter are used =
to send
back the responses correctly. </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 style=3D'margin-left:.5in'><font size=3D3 face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt'>This is an extremely old reference. The name =
was
changed to draft-ietf-sip-symmetric-response, and is now published as =
RFC 3581.
</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 style=3D'margin-left:.5in'><font size=3D3 face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial'>However if we don't use these extensions, and =
change
the via ip address and port with that of firewall/port then the SIP =
responses
will come directly to firewall, in that case the firewall permits the
responses. UA1 inside the network sends requests to =
firewall/NAT.</span></font>
</p>

<p class=3DMsoNormal =
style=3D'margin-right:0in;margin-bottom:12.0pt;margin-left:
.5in'><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<p style=3D'margin-left:.5in'><font size=3D3 face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial'>UA1</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 style=3D'margin-left:.5in'><font size=3D3 face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial'>INVITE sip:user@domain.com SIP/2.0 =
--------------</span></font>=E0<font
face=3DArial><span style=3D'font-family:Arial'> Firewall/NAT
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;
---------------------------------------------</span></font>=E0<font =
face=3DArial><span
style=3D'font-family:Arial'>UA2</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 style=3D'margin-left:.5in'><font size=3D3 face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial'>Via: SIP/2.0/UDP
10.1.1.1:4540&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;
INVITE sip:user@domain.com SIP/2.0 =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&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>

<p style=3D'margin-left:.5in'><font size=3D3 face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial'>Via: SIP/2.0/UDP =
firewall_ip:5060</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 style=3D'margin-left:.5in'><font size=3D3 face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;
(Maintain mapping)</span></font> </p>

<p class=3DMsoNormal =
style=3D'margin-right:0in;margin-bottom:12.0pt;margin-left:
.5in'><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<p style=3D'margin-left:.5in'><font size=3D3 face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial'>the UA2 will send the responses to =
firewall/NAT and
firewall/NAT will write back the addresses in Via to original IP/port. =
I
understand from the draft, if there are no SIP ALG functionality in =
firewall,
then the extensions are used to make SIP traverse through NAT. However =
if we
implement SIP ALG in firewall then we can make SIP traverse through =
NAT. Can
anybody tell that this way is per SIP spec.</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 style=3D'margin-left:.5in'><font size=3D3 face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt'>The experience of the working group has =
shown that
implementing a SIP ALG in a firewall is &quot;a bad thing&quot;. I have
personal experience with the failings of SIP ALGs at my own company, =
not the
least of which is that they don't work at all when SIP is =
integrity-protected
using TLS. </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 style=3D'margin-left:.5in'><font size=3D3 face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt'>Many UAs now support RFC 3581 and can use =
the STUN
protocol to get valid public addresses and ports to advertise in SDP =
for their
RTP traffic. This has been relatively painless compared to working with =
the
shortcomings of ALGs. </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 style=3D'margin-left:.5in'><font size=3D3 face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt'>thanks, </span></font></p>

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

</body>

</html>

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From exim@www1.ietf.org  Wed Feb 18 18:13:34 2004
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From: Robert Sparks <rsparks@dynamicsoft.com>
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Subject: [Sip] Summary of observations from SIPIT 14
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SIPIT 14 took place Feb 9 to 13, 2004 in Mandelieu, France.
The event was hosted by ETSI Plugtests. 

If you aren't familiar with SIPITs, please review the
information at http://www.sipit.net . 

This is a summary of what I observed:

There were roughly 128 attendees from 55 organizations.
Participants came from the US, Canada, Israel, Japan, 
Taiwan, France, Germany, Belgium, UK, Turkey, India, Sweden,
Finland, and Norway. Roughly 1/5 of the implementations had
not been to a previous SIPIT.

Startup at this event was more stressful than normal - we had a 
shipment of switches go awry and there was a good deal of last minute
creative network engineering, leading to a good deal of support
work on Monday. We resolved that by Monday night and had a solid
network for the remainder of the event. Still - the infrastructure
volunteers were up ridiculous hours the whole week. I have plans
to improve that. We started training a "loading crew" to help
with future events, and we are refining our kit and our requirements
on the host. If you're a regular SIPIT attendee and are willing
to help, please send me a note.

Overall, the state of interoperability at the event continues to
improve. The biggest strides between this event and the last were
around correct support for TLS. The biggest weakness continues
to be proper construction of offers and answers. Folks know
where to put their SDP, but can't agree what to put in it.
The offer-answer-examples draft is helping, but we may need
to do more.

We found a few places to work on clarifying the specification:

 - The merge detection language covers initial requests, but
   not in-dialog requests. We don't expect in-dialog requests
   to diverge and merge, but there are edge cases where it
   can happen (such as through NAPTR/SRV failover). This became
   particularly important when elements "optimized" failover,
   shorting over to the next SRV record after 4 seconds of no
   response instead of waiting for a full transaction timeout.

 - We inherit HEXDIG from 2234 (I think, this isn't super clear).
   That RFC allows only uppercase letters for hex digits. We 
   defined LHEX to deal with that for digest challenges (where
   we allow only lower case in the syntax). But - ipv6 addresses
   can realistically take either case. We need something new.

 - There were several UAs that puked on Call-IDs that had an
   @ sign in them where the stuff after the @ sign was a
   quoted ([]) IPv6 address (which happened when a request went
   through a v6tov4 gateway). We really should deprecate the
   '@' form from the BNF, replacing the whole header field
   value with an opaque bitstring.

 - It was not clear to all implementors that an in-dialog
   REFER was a target refresh request (hence MUST have a
   single Contact HFV. Bug 750 captures this.

 - An optimization question came up: If you parallel fork
   over UDP to n legs, leg 1 answers, leg 2-n have not yet 
   responded (at all, so you can't send a CANCEL). Do you 
   still have to continue retransmitting the request on
   legs 2-n, or can you squelch them? (You would have to
   keep the transaction around to deal with a response if
   it came, of course).

 - Empty reason phrases broke lots of people - we will add
   these to the torture-test draft. Note that the BNF still
   requires a space after the Status-Code even if the phrase
   is empty.

 - We had another example of Table 2 causing more confusion
   than not - a new registrar decided it was ok not to include
   the current bindings in REGISTER responses since table 2
   listed Contact as optional in that context.

I was able to interview 46 of the 55 organizations. Here are
some observations from those notes - treat all the numbers
as approximations.

- 20 did not support TLS, but 5 of those plan to by the next
  event. (So we're running at about half the implementations
  supporting TLS.) All but one implementation was built on top
  of openssl. Only 6 implementations supported sips:

- We had two implementations of event-lists (and they interoperated)

- Only four implementations did not support loose-routing.

- around 10 implementations supported IPv6. 

- Only 9 implementations understood UPDATE

- I noted 12 implementations supported NAPTR+SRV, 4 SRV only, 8 do 
  A lookups only, and 4 did no DNS at all. My notes only captured this
  information for roughly half the teams, but I think the proportions
  are representative.

- 7 implementations supported S/MIME

- 18 implementations supported PRACK

- UA implementations (including proxies and b2buas) outnumbered 
  proxy/registrar implementations around 4 to 1.

- There were fewer SIMPLE implementations (and most of the ones 
  present were presence server only). Many teams left their SIMPLE
  work at home, waiting for a SIMPLEt.


The multiparty tests at this event were similar to those of
the past. The spiral test continues to have the highest 
participation and yields the most implementation bugs. We
refined the test scenario for REFER, setting up a star topology
that lets us configure once and run many scenarios. We ran a
NAPTR/SRV failover test that was intricate. Several implementations
performed correctly when it became necessary to fail-over. We
exposed one that was only taking the first returned record of
a given type, and others that got their weighting algorithms
backwards. We didn't have many STUN implementations, but those
that were there worked together. We only had one SIGCOMP 
implementation show up this time.


RjS



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From exim@www1.ietf.org  Wed Feb 18 18:27:37 2004
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Subject: [Fwd: [Sip] Summary of observations from SIPIT 14]
From: Robert Sparks <rsparks@dynamicsoft.com>
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I just sent:

-----Forwarded Message-----
> From: Robert Sparks <rsparks@dynamicsoft.com>
> To: sip@ietf.org, sip-implementors@ietf.org
> Subject: [Sip] Summary of observations from SIPIT 14
> Date: Wed, 18 Feb 2004 17:10:13 -0600

Of course, the second address is bogus. The correct
address for that list appears in the headers of this
message.

Apologies now to those who will "Reply-All" to the
original and get a bounce.

RjS


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From: Bryan Thale <bryan.thale@motorola.com>
Organization: Networks & Infrastructure Research, Motorola Labs
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Subject: Re: [Sip] Binary Encoding for SIP
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Hi Cullen,

I'd vote for 'b', not because the idea is stupid but because it is 
unnecessary.  Why restrict the ability of the end-point service to 
choose the most efficient encoding for its purposes?  If that happens to 
be 'binary' then great, but if not, that's OK, too.

I agree with this sentiment from your draft:

> ... it is reasonable to suggest that SIP implementations send one type 
> of encoding


However, this is at odds with the ultimate recommendation of your draft:

> Devices MUST use a content transfer encoding of "binary" for MIME body 
> parts in SIP messages they send.


If the problem is that implementors are inappropriately choosing Base64 
due to lack of sufficient guidance, I would prefer to see "Devices 
SHOULD use ..." instead.  It provides the needed guidance and yet does 
not arbitrarily close doors on possible future implementations.

Bryan.

-- 
Bryan Thale
Networks & Infrastructure Research, Motorola Labs
mailto:bryan.thale@motorola.com



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Subject: RE: [Sip] draft-khartabil-sip-auth-analysis-00.txt
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> -----Original Message-----
> From: ext Scott Lawrence [mailto:slawrence@pingtel.com]
> Sent: 18.February.2004 17:30
> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> Cc: sip@ietf.org
> Subject: Re: [Sip] draft-khartabil-sip-auth-analysis-00.txt
>=20
>=20
>=20
>=20
> > >   realm
> > >     A string to be displayed to users so they know which=20
> username and
> > >     password to use.
> >=20
> > I don't think this takes into consideration pre-configuration.
>=20
> That's not really the point; the realm needs to provide=20
> enough of a hint
> for the UA to determine which credentials it should use.  Setting up a
> configuration that requires different credentials for the=20
> same realm at
> different servers is dumb, and not something I think we need to spend
> much worry on.

It not about setting different credentials for the same realm. It is =
about pre-configuring the wrong credentials for a realm. If my phone is =
preconfigured with username: hisham, password: hishampass, realm: =
nokia.com and:

--> INVITE
<-- 407
--> INVITE
<-- 407 (credentials where wrong)

At this stage, the user should be prompted to enter new =
username/password. My objection is to the text quoted below from =
RFC3261:

"  A service provider that pre-configures UAs with credentials for its
   realm should be aware that users will not have the opportunity to
   present their own credentials for this realm when challenged at a
   pre-configured device".

I think the users should have the opportunity.

>=20
> > >                 ... This string should contain at least=20
> the name of
> > >    the host performing the authentication and might additionally
> > >    indicate the collection of users who might have=20
> access. An example
> > >    might be "registered_users@gotham.news.com".
> >=20
> > Can we mandate that for SIP??
>=20
> I don't think that we can mandate anything that is by nature an
> administrative decision - we already provide good advice. =20
> However, 3261
> does say (22.1):
>=20
>       o  Realm strings MUST be globally unique.  It is=20
> RECOMMENDED that
>          a realm string contain a hostname or domain name,=20
> following the
>          recommendation in Section 3.2.1 of RFC 2617 [17].

MUST be globally unique for every proxy in the same administrative =
domain? I don't think this is required.

>=20
> > > As a practical matter, if you have the same username and=20
> password in
> > > both domains it doesn't matter - if you don't, then it's=20
> an impossibly
> > > clumsy situation (this problem is part of the reason that=20
> we defined
> > > proxy authentication to by hop-by-hop in HTTP in the first place).
> >=20
> > Perhaps I was not clear in my text. Imagine this scenario:=20
> I have cached my credentials for an outbound proxy with realm=20
> proxy.nokia.com from an earlier transaction. Now I send a new=20
> INVITE routed through the outbound proxy that gets a 407 from=20
> a second proxy with the same realm. How do I know if the 407=20
> came from the outbound proxy (the username password might=20
> have changed in the meantime) or from the next hop proxy?
>=20
> What difference does it make?  You just use the credentials=20
> you have in
> either case, or you fail... =20

It matters because the UA doesn't know if the credentials supplied to =
the first proxy were wrong and therefore is sending another 407 or if =
the second proxy is challenging with the same realm. The qop ting you =
mention below should fix this.

>As I pointed out, you can tell the
> difference if the server is using qop.
>=20
>=20
> > > If the proxy is using the 2617 version of Digest (with the qop
> > > attribute) as opposed to the 1867 version, then the UA can tell
> > > the difference.  The message flow looks like this:
> > >=20
> > >         UA                    P1                  P2
> > >     a    +------REQ 1--------->|                   |
> > >     b    |<----407(1)----------+                   |
> > >     c    +------REQ 2 -------->+----REQ 2--------->|
> > >     e    |<---407(2)-----------+<-----407(2)-------|
> > >          +------REQ 3--------->+----REQ 3--------->|
> > >=20
> > >  a) initial request
> > >  b) challenge from P1, with the qop parameter specified
> > >  c) retried request, with a qop parameter, and therefor=20
> also a cnonce
> > >     chosen by the UA; it is passed on (now without=20
> > > authentication) to P2
> > >  d) P2 challenges, also specifying a qop parameter
> > >     P2 challenge is forwarded by P1, with the WWW-Authenticate
> > >     header as provided by P2, but also with an Authentication-Info
> > >     provided by P1.
> > >=20
> > > The UA can recognize that authentication to P1 was=20
> successful, because
> > > P1 returned an Authentication-Info header that contained=20
> the cnonce
> > > and nc values from message c (REQ 2), and a correct=20
> response hash to
> > > show that P1 knows the authentication secret.
> >=20
> > This is not mandated for SIP entities, maybe it aught to be.
>=20
> Actually, it is, but it's the usual unfortunate incorrect=20
> intrepretation
> of SHOULD that is biting us again.  RFC 2617 says:
>=20
>    qop-options
>      This directive is optional, but is made so only for backward
>      compatibility with RFC 2069 [6]; it SHOULD be used by all
>      implementations compliant with this version of the Digest scheme.
>=20
> Since SIP references only 2617, and backward compatibility=20
> with 1867 is
> not important, all SIP servers should be using qop-options.

This is all great, although I prefer for this to be called out =
explicitly for SIP, making the SHOULD a MUST. Also, there is nothing =
mandating a proxy to send the Authentication-info header. Perhaps that =
aught to be mandated also if a proxy server proxies a request it had =
previously challenged and another proxy downstream challenges the same =
request with the same realm.

>=20
> > > > 10. Authentication-Info
> > > >
> > > >  RFC 3261 allows the usage of Authentication-Info=20
> header. The BNF in
> > > >  RFC 3261 allows multiple authentication-info headers=20
> where RFC 2617
> > > >  allows only one. Is it only the terminating UAS that=20
> is allowed to
> > > >  insert this header. If so, why allow multiples to be=20
> > > present. If not,
> > > >  how does the UAC know which proxy added this header?=20
> It cannot know
> > > >  since there is no parameter indicating so, not even nonce.
> > >=20
> > > As I noted earlier, 2617 allows only one because proxy=20
> authentication
> > > is hop-by-hop, but the cnonce value allows the UA to=20
> disambiguate them
> > > (if they are carefully constructed, but then they should=20
> be anyway).
> >=20
> > SIP BNF allows multiples of those headers. Is this a BNF=20
> bug? If not, then some Authentication-Info parameters need to=20
> be mandated.
>=20
> The BNF is never the last word - the text MUST be read too, and the
> requirement there is, I think, already clear.

Ok, although it is only implied that message-qop MUST appear in =
Authentication-info header if the earlier challenge/response contained =
the qop.

>=20
> --=20
> Scott Lawrence       =20
>   Pingtel Corp.  =20
>=20
>=20
>=20

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CONFERENCE ANNOUNCEMENT & CALL FOR PRESENTATIONS 

------------------------------------------------------------------------------
      INTERNETWORKING 2004
Technical Program: May 13-14, 2004
          Las Vegas, Nevada
http://www.caitr.org/internetworking04/
In conjunction with NetWorld+Interop Las Vegas 2004
--------------------------------------------------------------------------------
The Internetworking 2004 Program Committee cordially invites you to submit proposals for original, unpublished presentations focusing on internetworking technologies in the IP, optical, and wireless domains. 

Summaries not exceeding 250 words should be submitted (in plain text format) to submissions at caitr.org for review and possible inclusion in the program, no later than February 25, 2004.

Topics of interest include, but are not limited to the following: 

   Voice over IP (VoIP) 
   Virtual Private Networks 
   Routing and Quality of Service (QoS) 
   Network Processors 
   Network Security 
   Service Integration 
   Operational Support Systems 
   Transition from IPv4 to IPv6 and Interworking 
   Network Survivability and Fault Management 
   Wireless Internet 
   Next Generation Mobile Networks 
   Internetworking IP and Optical Networks 
   Internetworking MPLS with Legacy ATM and Frame Relay Networks 
   Fiber To The Home (FTTH) 
   High Speed Transport Layer Protocols 
   Unicast and Multicast Routing and Convergence 
   Storage Area Networks (SANs) 
   Peer to Peer Networking 
   Pervasive Computing 
   Grid Computing 
   IP Video Conferencing 
   High Definition Video Distribution 

For additional information, please contact info at caitr.org, or the conference technical chair, Dr. Parviz Yegani (<pyegani at cisco.com>).





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<DIV>
<DIV>
<DIV>
<DIV>CONFERENCE ANNOUNCEMENT &amp; CALL FOR PRESENTATIONS <BR><BR>------------------------------------------------------------------------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; INTERNETWORKING 2004<BR>Technical Program: May 13-14, 2004<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Las Vegas, Nevada<BR><A href="http://www.caitr.org/internetworking04/" target=_blank>http://www.caitr.org/internetworking04/</A></DIV>
<DIV>In conjunction with NetWorld+Interop Las Vegas 2004</DIV>
<DIV>--------------------------------------------------------------------------------<BR>The Internetworking 2004 Program Committee cordially invites you to submit proposals for original, unpublished presentations focusing on internetworking technologies in the IP, optical, and wireless domains. <BR><BR>Summaries not exceeding 250 words should be submitted (in plain text format) to <A href="http://lists.cs.columbia.edu/mailman/listinfo/tccc" target=_blank><FONT color=#0000ff>submissions at caitr.org</FONT></A> for review and possible inclusion in the program, no later than February 25, 2004.<BR><BR>Topics of interest include, but are not limited to the following: </DIV>
<DIV><BR>&nbsp;&nbsp; Voice over IP (VoIP) <BR>&nbsp;&nbsp; Virtual Private Networks <BR>&nbsp;&nbsp; Routing and Quality of Service (QoS) <BR>&nbsp;&nbsp; Network Processors <BR>&nbsp;&nbsp; Network Security <BR>&nbsp;&nbsp; Service Integration <BR>&nbsp;&nbsp; Operational Support Systems <BR>&nbsp;&nbsp; Transition from IPv4 to IPv6 and Interworking <BR>&nbsp;&nbsp; Network Survivability and Fault Management <BR>&nbsp;&nbsp; Wireless Internet <BR>&nbsp;&nbsp; Next Generation Mobile Networks <BR>&nbsp;&nbsp; Internetworking IP and Optical Networks <BR>&nbsp;&nbsp; Internetworking MPLS with Legacy ATM and Frame Relay Networks <BR>&nbsp;&nbsp; Fiber To The Home (FTTH) <BR>&nbsp;&nbsp; High Speed Transport Layer Protocols <BR>&nbsp;&nbsp; Unicast and Multicast Routing and Convergence <BR>&nbsp;&nbsp; Storage Area Networks (SANs) <BR>&nbsp;&nbsp; Peer to Peer Networking <BR>&nbsp;&nbsp; Pervasive Computing <BR>&nbsp;&nbsp; Grid Computing <BR>&nbsp;&nbsp; IP Video Conferencing
 <BR>&nbsp;&nbsp; High Definition Video Distribution <BR><BR>For additional information, please contact <A href="http://lists.cs.columbia.edu/mailman/listinfo/tccc" target=_blank><FONT color=#0000ff>info at caitr.org</FONT></A>, or the conference technical chair, Dr. Parviz Yegani (&lt;<A href="http://lists.cs.columbia.edu/mailman/listinfo/tccc" target=_blank><FONT color=#0000ff>pyegani at cisco.com</FONT></A>&gt;).<BR></DIV></DIV></DIV></DIV>
--0-1905995372-1077205712=:94531--

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To: Rohan Mahy <rohan@cisco.com>, Anil Bollineni <ABollineni@netscreen.com>
From: Jiri Kuthan <jiri@iptel.org>
Subject: Re: [Sip] Changing Via In NAT
Cc: "'sip@ietf.org'" <sip@ietf.org>
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At 01:46 PM 2/18/2004, Rohan Mahy wrote:
>The experience of the working group has shown that implementing a SIP ALG in a firewall is "a bad thing".  I have personal experience with the failings of SIP ALGs at my own company, not the least of which is that they don't work at all when SIP is integrity-protected using TLS. 
>
>Many UAs now support RFC 3581 and can use the STUN protocol to get valid public addresses and ports to advertise in SDP for their RTP traffic. This has been relatively painless compared to working with the shortcomings of ALGs. 

Absolutely -- rport is very essential part of the story as it adds lot of
robustness to traversal of SIP transactions over NATs. Sending replies
back symmetricaly simply works. Not using rport leads to all kind of
instabilities such as missed SIP replies when STUN address in Via without
rport becomes invalid for whatever reason.

Besides that, adding rport to UAC is banal.

-jiri  


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Date: Thu, 19 Feb 2004 18:00:58 -0800
Subject: Re: [Sip] SIP instance identifiers
From: Cullen Jennings <fluffy@cisco.com>
To: Brian Stucker <bstucker@nortelnetworks.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Paul H Kyzivat <pkyzivat@cisco.com>
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On 2/17/04 11:26 PM, "Brian Stucker" <bstucker@nortelnetworks.com> wrote:

> . I also don't see a reason why we should expect
> other nodes in the network to be able to use the instance identifier for
> anything 
> other than a token (ie. if it's a mac address, and two UA instances do
> something 
> because it's a MAC address, that's OK, but other's may not take that tact).

I sort of agree and sort of don't. The only important thing about MAC is
that most phone vendors have it on the outside of the box and you can read
it with a barcode scanner. If a user goes to some configuration web page and
see that they have 2 devices registered, they have some hope of finding an
identifier on the back of the phone (the MAC) that they can match to one of
the two instances on the web page. 


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Date: Thu, 19 Feb 2004 18:01:05 -0800
From: Cullen Jennings <fluffy@cisco.com>
To: Dean Willis <dean.willis@softarmor.com>,
        Brian Stucker <bstucker@nortelnetworks.com>
CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, <sip@ietf.org>
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Subject: [Sip] Re: Abstraction:  SIP instance identifiers requirements
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I don't think this unique instance identifier thing had much to do with
billing or debugging. I'm imagine a case something like the following:

I have two offices, one on the west side of the building and one of the
east. They both have phones registered as fluffy. Because it is better for
lounging in the Sun, I spend the morning in the East office and afternoon in
the West. Right now I can't write a proxy that forward to the correct one.

If I was using SUBSCRIBE to support event notification of provisioning
changes, I would not be able to identify which phone was subscribing.


On 2/17/04 11:16 AM, "Dean Willis" <dean.willis@softarmor.com> wrote:

> 1) billing -- "Alice called Bob from phone ID000001 through gateway
> ID3344555 on Tuesday, March 1, 2011"
> 
> 2) Debugging -- "This request was responded to by node IDYIYIOHIHIUH,
> which is a Barfia model 9995 phone running Finux 21.05A and the Vocal
> user agent release 89.6"


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To: Cullen Jennings <fluffy@cisco.com>
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Just to be clear, I am not *advocating* that we need these identifiers 
any other place. I'm just saying that if there are other uses for a 
device id, that it won't hurt to use the same id elsewhere.

For the purpose we are discussing, it is important that the id be 
associated with a registered contact, and since REGISTER permits 
multiple contacts, it really needs to be a parameter on each contact.

That is really an entirely different application from identifying which 
device originated a particular message. If there is a need for that, 
then it needs to be in some header that can be in any message. I don't 
recommend we do that until we have a need for it.

	Paul

Cullen Jennings wrote:
> On 2/17/04 2:28 PM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
> 
> 
>>I can see 
>>validity in Brian's desire to have it in a header of its own,
>>potentially in any message. I don't see those usages as conflicting.
> 
> 
> Just to be clear, I don't want it to have multiple possible place it could
> be in a SIP message and I have to go traipsing around the message trying
> here then there unless it is a FOO message where you look over here. A
> header would work or caller pref but I don't see an argument for doing both.
> I want it just one place.
> 
> The case with PUBLISH not having contact header had concerned me (actually I
> think this was a mistake) but seems what happens in PIDF bodies could be
> different.
> 
> 


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Subject: Re: [Sip] SIP instance identifiers
From: Cullen Jennings <fluffy@cisco.com>
To: Paul H Kyzivat <pkyzivat@cisco.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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On 2/17/04 2:28 PM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:

> I can see 
> validity in Brian's desire to have it in a header of its own,
> potentially in any message. I don't see those usages as conflicting.

Just to be clear, I don't want it to have multiple possible place it could
be in a SIP message and I have to go traipsing around the message trying
here then there unless it is a FOO message where you look over here. A
header would work or caller pref but I don't see an argument for doing both.
I want it just one place.

The case with PUBLISH not having contact header had concerned me (actually I
think this was a mistake) but seems what happens in PIDF bodies could be
different.


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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, 19 Feb 2004 20:32:57 -0800
Subject: Re: [Sip] draft-khartabil-sip-policy-uri-call-info-purpose-00
From: Cullen Jennings <fluffy@cisco.com>
To: <hisham.khartabil@nokia.com>, <sip@ietf.org>
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I was hoping it would only be conferences. If it is more than that, then I
would need to know what I should expect to find when I load the document the
URI points at. 

Cullen


On 2/16/04 1:07 AM, "hisham.khartabil@nokia.com"
<hisham.khartabil@nokia.com> wrote:

> Thanks.
> 
> Do you think that it will only be limited to conferences?
> 
> /Hisham
> 
>> -----Original Message-----
>> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
>> Cullen Jennings
>> Sent: 13.February.2004 18:49
>> To: sip@ietf.org
>> Subject: [Sip]
>> 
>> 
>>     
>> I like this draft, just one little NIT, I think it would be
>> better to call
>> the tag conf-policy-info instead of just policy-info
>> 
>> Cullen
>> 
>> 
>> 
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>> 
> 


_______________________________________________
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Date: Thu, 19 Feb 2004 22:21:19 -0800
Subject: Re: [Sip] Binary Encoding for SIP
From: Cullen Jennings <fluffy@cisco.com>
To: Paul H Kyzivat <pkyzivat@cisco.com>
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Good point. 

I think that all they ways I can imagine getting indirect content, HTTP,
FTP, TFTP, NFS, SMB, file are all 8 bit safe so I think this argument
probably still holds.

Cullen


On 2/17/04 6:27 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:

> Cullen,
> 
> This is an attractive proposal because it has potential to make sip
> implementations simpler. But before I can answer I need clarification on
> one thing:
> 
> Does this apply to content referenced and fetched via content-indirection?
> 
> This proposal only has value if my mime processor *never* needs to deal
> with any content-transfer-encoding other than binary. If content
> referenced by content indirection still might use other encodings, then
> that value is lost. Unfortunately, given the potential range of
> transports that might be used with content indirection, I don't know if
> requiring binary can be done in general.
> 
> Paul
> 
> Cullen Jennings wrote:
>>     
>> I submitted a draft on binary encoding for SIP. This improves
>> interoperability, improves speed, decreases sizes of messages, and generally
>> makes life better for SIP which will no doubt eventually help with World
>> Hunger. I need feedback, please let me know:
>> 
>> a) Yah, lets do it
>> 
>> b) No this is a stupid ideas and here are the reasons why
>> 
>> c) There has not been many list comments, this draft involves some very
>> complex issues. I need a year or two to develop an opinion on the single
>> sentence inside it.
>> 
>> d) I agree with Cowboy Willis
>> 
>> Until it arrives in the archives, you can find it at
>> 
>> http://www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.html
>> 
>> or, if you prefer ASCII encodings of everything,
>> 
>> www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.txt
>> 
>> Thanks, Cullen
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>> 
> 
> 


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Date: Thu, 19 Feb 2004 22:18:51 -0800
Subject: Re: [Sip] Binary Encoding for SIP
From: Cullen Jennings <fluffy@cisco.com>
To: Marc Petit-Huguenin <petithug@acm.org>
CC: <sip@ietf.org>
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You are right - 3261 definitely allows binaries bodies.

I am just trying to suggest that it would be better to send MIME bodies with
a transfer encoding set to binary instead of base64 when sending them. Right
now SIP allows either, the examples are in base64, but binary is probably a
better choice. It seems like it should not be a big deal but it has been one
of the frustrating interoperability issues in getting S/MIME to work.

Cullen

On 2/15/04 9:03 AM, "Marc Petit-Huguenin" <petithug@acm.org> wrote:

> I thought that binary body was already supported: RFC3261 section 7.4.1,
> third paragraph.
> 
> Cullen Jennings wrote:
>>     
>> I submitted a draft on binary encoding for SIP. This improves
>> interoperability, improves speed, decreases sizes of messages, and generally
>> makes life better for SIP which will no doubt eventually help with World
>> Hunger. I need feedback, please let me know:
>> 
>> a) Yah, lets do it
>> 
>> b) No this is a stupid ideas and here are the reasons why
>> 
>> c) There has not been many list comments, this draft involves some very
>> complex issues. I need a year or two to develop an opinion on the single
>> sentence inside it.
>> 
>> d) I agree with Cowboy Willis
>> 
>> Until it arrives in the archives, you can find it at
>> 
>> http://www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.html
>> 
>> or, if you prefer ASCII encodings of everything,
>> 
>> www.employees.org/~fluffy/ietf/draft-jennings-sip-mime-01.txt
>> 
>> Thanks, Cullen
>> 
>> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


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Date: Thu, 19 Feb 2004 21:48:06 -0800
Subject: Re: [Sip] RE: [Sip-implementors] Via: processing
From: Cullen Jennings <fluffy@cisco.com>
To: <ranjit.avasarala@wipro.com>, <nataraju.alilaghatta@wipro.com>,
        <Paul.D.Smith@dataconnection.com>, <brett@broadsoft.com>
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No I don't think this is a good idea. Putting a FQDN that is behind a NAT is
not any more likely to be resolvable to a routeable IP address. A private IP
is much easier to recognize and debug. Many hosts think they know their
hostname but do not and they should not put in an address that is not
routable in any domain. For example. most things, be it windows or unix,
will report a hostname when really they got their IP address from DHCP and
the hostname they are reporting is not in DNS.

Here's my recommendation for IP phones, softphones - always use an IP in via
and contacts unless you are positive that your FQDN really is in the global
DNS and you have a specific reason to use DNS (like multiple NICs for
redundancy). 

Cullen


On 2/12/04 10:20 PM, "ranjit.avasarala@wipro.com"
<ranjit.avasarala@wipro.com> wrote:

> Hi
>  actually putting IP address in Via is not recommended when the SIP
> message has to traverse through NAT. So it is better to have FQDN
> instead of IP. 
> 
> Ranjit
> 
> 
> 
> 
> 
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
> Nataraju Alilaghatta (WT01 - TELECOM & INTER-NETWORKING SOLUTIONS)
> Sent: Friday, February 13, 2004 4:37 AM
> To: Paul.D.Smith@dataconnection.com; brett@broadsoft.com
> Cc: sip@ietf.org
> Subject: RE: [Sip] RE: [Sip-implementors] Via: processing
> 
> 
> I guess putting FQDN or IP addresses in the Via header driven by the
> tradeoff between overhead in DNS query for FQDN name but provides
> robustness or direct usage of IP address which lead to undeliverable in
> case intermediate entities failure.
> 
> It's my feeling, please correct me if I am wrong....
> 
> Thanks & Regards,
> Nataraju A.B. 
> 
> 
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Paul
> D.Smith
> Sent: Friday, January 30, 2004 9:56 PM
> To: 'brett@broadsoft.com'
> Cc: SIP WG
> Subject: [Sip] RE: [Sip-implementors] Via: processing
> 
> Brett, Arlie,
> 
> RFC3261 clearly allows a response to be "rerouted" if it can be
> determined that the response did not reach the first-attempt target.
> For example, if the request is received over TCP, and the connection has
> gone and cannot be reestablished by the time the response is ready to be
> sent back, RFC3261 states the following
> 
> --- from RFC3261, section 18.2.2, 1st bullet.
> 
> If that connection attempt fails, the server SHOULD use the procedures
> in [4] for servers in order to determine the IP address and port to open
> the connection and send the response to.
> 
> --- end of cut
> 
> RFC3263 then gives an algorithm for attempting to determine alternate IP
> addresses to which the response might be sent.  Clearly having an FQDN
> in the sent-by will give a chance of success which having an IP address
> will not, especially if the sent-by and received IP addresses are the
> same!
> 
> 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
> 
> 
> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]
> Sent: 29 January 2004 00:27
> To: SIP (E-mail)
> Subject: RE: [Sip-implementors] Via: processing
> 
> 
>> I'm looking at the spec for Via: processing.
>> RFC 3261 clearly shows that a Via: line can
>> contain DNS names, as well as textual
>> representations of network addresses.
>> 
>> How many products actually implement support
>> for resolving DNS names provided in Via: lines?
> 
> Not very many support DNS resolution of a
> Via entry.  However many support the concept
> of adding the Via's received parameter when
> it does not match the Via's sent-by address.
> 
>> This seems...  well, it seems unnecessary,
>> potentially insecure (allowing for easy DDoS
>> and potentially amplification attacks), and
>> just a general nuisance.  Is this kind of
>> thing ever actually used for a good purpose?
> 
> Some people have toyed with the idea of
> advancing response addresses based upon
> rfc3263 during failure situations.
> 
>> Does any product do anything *but* just put
>> real network addresses in Via: lines in
>> requests?
> 
> Yes.
> 
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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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This list is for NEW development of the core SIP Protocol
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Date: Thu, 19 Feb 2004 22:30:44 -0800
Subject: Re: [Sip] Binary Encoding for SIP
From: Cullen Jennings <fluffy@cisco.com>
To: Bryan Thale <bryan.thale@motorola.com>, <sip@ietf.org>,
        Henning Schulzrinne <hgs@cs.columbia.edu>
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I sat for a long time looking at that one MUST and trying to decide if it
should be a SHOULD. I also avoided mentioning what Henning points is the
more interesting question of what must all UAs be willing to receive.

So I take if you would be happy to see is say that things SHOULD send binary
but need to be prepared to receive anything.

I can live with that. Fits a certain stringent in what you send, lenient in
what you receive model that I like. I doubt that implementers will be busy
to go off and implement receiving multiple different things when there is no
apparent value to doing that. It would solve my one really key issues which
is that I would like to give enough guidance on the minimum to implement
that things like S/MIME had a better chance of working together.

Cullen



On 2/17/04 9:44 AM, "Bryan Thale" <bryan.thale@motorola.com> wrote:

> Hi Cullen,
> 
> I'd vote for 'b', not because the idea is stupid but because it is
> unnecessary.  Why restrict the ability of the end-point service to
> choose the most efficient encoding for its purposes?  If that happens to
> be 'binary' then great, but if not, that's OK, too.
> 
> I agree with this sentiment from your draft:
> 
>> ... it is reasonable to suggest that SIP implementations send one type
>> of encoding
> 
> 
> However, this is at odds with the ultimate recommendation of your draft:
> 
>> Devices MUST use a content transfer encoding of "binary" for MIME body
>> parts in SIP messages they send.
> 
> 
> If the problem is that implementors are inappropriately choosing Base64
> due to lack of sufficient guidance, I would prefer to see "Devices
> SHOULD use ..." instead.  It provides the needed guidance and yet does
> not arbitrarily close doors on possible future implementations.
> 
> Bryan.


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



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Date: Fri, 20 Feb 2004 18:53:02 +0900
From: Kumiko Ono <ono.kumiko@lab.ntt.co.jp>
Subject: Re: [Sip] Binary Encoding for SIP
To: Cullen Jennings <fluffy@cisco.com>, Bryan Thale <bryan.thale@motorola.com>,
        sip@ietf.org, Henning Schulzrinne <hgs@cs.columbia.edu>
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Hi Cullen,

I am for 'a'. However, there may be some proxy severs who have the 
capability in handling 8-bit (text) data, not binary data. There may be 
an impact on all SIP entities including proxy servers that need not 
handle S/MIME data. 

Regards,
Kumiko 

>I sat for a long time looking at that one MUST and trying to decide if it
>should be a SHOULD. I also avoided mentioning what Henning points is the
>more interesting question of what must all UAs be willing to receive.
>
>So I take if you would be happy to see is say that things SHOULD send binary
>but need to be prepared to receive anything.
>
>I can live with that. Fits a certain stringent in what you send, lenient in
>what you receive model that I like. I doubt that implementers will be busy
>to go off and implement receiving multiple different things when there is no
>apparent value to doing that. It would solve my one really key issues which
>is that I would like to give enough guidance on the minimum to implement
>that things like S/MIME had a better chance of working together.
>
>Cullen
>
>
>
>On 2/17/04 9:44 AM, "Bryan Thale" <bryan.thale@motorola.com> wrote:
>
>> Hi Cullen,
>> 
>> I'd vote for 'b', not because the idea is stupid but because it is
>> unnecessary.  Why restrict the ability of the end-point service to
>> choose the most efficient encoding for its purposes?  If that happens to
>> be 'binary' then great, but if not, that's OK, too.
>> 
>> I agree with this sentiment from your draft:
>> 
>>> ... it is reasonable to suggest that SIP implementations send one type
>>> of encoding
>> 
>> 
>> However, this is at odds with the ultimate recommendation of your draft:
>> 
>>> Devices MUST use a content transfer encoding of "binary" for MIME body
>>> parts in SIP messages they send.
>> 
>> 
>> If the problem is that implementors are inappropriately choosing Base64
>> due to lack of sufficient guidance, I would prefer to see "Devices
>> SHOULD use ..." instead.  It provides the needed guidance and yet does
>> not arbitrarily close doors on possible future implementations.
>> 
>> Bryan.
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip

--
Kumiko Ono
NTT Network Service Systems Labs.
Tel +81 422 59 4508/Fax +81 422 60 4012

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Folks,

it seems that some people in the ITU-T are having some issues figuring 
out whether or not certain requests can be sent by the UAS within an 
early dialog. The confusion comes from statements like this one (taken 
from the INFO spec):

"     Unless stated otherwise, the protocol rules for the INFO request
       governing the usage of tags, Route and Record-Route,
       retransmission and reliability, CSeq incrementing and message
       formatting follow those in [1] as defined for the BYE request."

Since a UAS does not send BYEs within an early dialog, but a non-2xx 
final response instead, to terminate the dialog, people assume that INFO 
cannot be sent in this situation either.

Let's try to keep this in mind and clarify this point when/if we define 
new SIP methods.

In any event, the clarification to this issue can be found in RFC 3398:

"From the callee to the caller, if a message is received by a gateway
    before the call has been answered (i.e., ANM is received) it SHOULD
    be encapsulated in an INFO, provided that this will not be the first
    SIP message sent in the backwards direction (in which case it SHOULD
    be encapsulated in a provisional 1xx response)."

Regards,

Gonzalo

 

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To: "Gonzalo Camarillo (JO/LMF)" <gonzalo.camarillo@ericsson.com>,
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Hi Gonzalo,

Thanks for your reply!

>it seems that some people in the ITU-T are having some issues 
>figuring out whether or not certain requests can be sent by the UAS within an 
>early dialog. The confusion comes from statements like this 
>one (taken from the INFO spec):
> 
> "     Unless stated otherwise, the protocol rules for the INFO request
>        governing the usage of tags, Route and Record-Route,
>        retransmission and reliability, CSeq incrementing and message
>        formatting follow those in [1] as defined for the BYE request."
> 
>Since a UAS does not send BYEs within an early dialog, but a non-2xx 
>final response instead, to terminate the dialog, people assume that INFO 
>cannot be sent in this situation either.
> 
>Let's try to keep this in mind and clarify this point when/if 
>we define new SIP methods.
> 
>In any event, the clarification to this issue can be found in 
>RFC 3398:
> 
>"From the callee to the caller, if a message is received by a gateway
>before the call has been answered (i.e., ANM is received) 
>it SHOULD be encapsulated in an INFO, provided that this will not 
>be the first SIP message sent in the backwards direction (in which 
>case it SHOULD be encapsulated in a provisional 1xx response)."

The problem is that the ITU-T interworking spec doesn't reference RFC3398. The spec reference the INFO RFC, and that is also where I think the usage rules should be defined (not in another spec, RFC3398 in this case, simply using the method).

The INFO RFC, as you said, currently says that INFO "follows the rules of BYE" (whatever that means is another question :), and the rules for BYE specifically says that it must not be sent backward before the INVITE has finished. If the INFO RFC would say "follow the rules of UPDATE" we would not have this problem, since the UPDATE spec specifically allows the use of the method before the INVITE has finished.

Thanks!

Christer Holmberg
Ericsson Finland

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Christer,

would the ITU-T be happier if we log an errata against the INFO RFC 
saying that UASs can send INFO within early dialogs?

Gonzalo

Christer Holmberg (JO/LMF) wrote:

> Hi Gonzalo,
> 
> Thanks for your reply!
> 
> 
>>it seems that some people in the ITU-T are having some issues 
>>figuring out whether or not certain requests can be sent by the UAS
> 
> within an 
> 
>>early dialog. The confusion comes from statements like this 
>>one (taken from the INFO spec):
>>
>>"     Unless stated otherwise, the protocol rules for the INFO request
>>       governing the usage of tags, Route and Record-Route,
>>       retransmission and reliability, CSeq incrementing and message
>>       formatting follow those in [1] as defined for the BYE request."
>>
>>Since a UAS does not send BYEs within an early dialog, but a non-2xx 
>>final response instead, to terminate the dialog, people assume that
> 
> INFO 
> 
>>cannot be sent in this situation either.
>>
>>Let's try to keep this in mind and clarify this point when/if 
>>we define new SIP methods.
>>
>>In any event, the clarification to this issue can be found in 
>>RFC 3398:
>>
>>"From the callee to the caller, if a message is received by a gateway
>>before the call has been answered (i.e., ANM is received) 
>>it SHOULD be encapsulated in an INFO, provided that this will not 
>>be the first SIP message sent in the backwards direction (in which 
>>case it SHOULD be encapsulated in a provisional 1xx response)."
> 
> 
> The problem is that the ITU-T interworking spec doesn't reference
> RFC3398. The spec reference the INFO RFC, and that is also where I think
> the usage rules should be defined (not in another spec, RFC3398 in this
> case, simply using the method).
> 
> The INFO RFC, as you said, currently says that INFO "follows the rules
> of BYE" (whatever that means is another question :), and the rules for
> BYE specifically says that it must not be sent backward before the
> INVITE has finished. If the INFO RFC would say "follow the rules of
> UPDATE" we would not have this problem, since the UPDATE spec
> specifically allows the use of the method before the INVITE has
> finished.
> 
> Thanks!
> 
> Christer Holmberg
> Ericsson Finland


 

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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "Gonzalo Camarillo (JO/LMF)" <gonzalo.camarillo@ericsson.com>
Cc: sip <sip@ietf.org>, Rohan Mahy <rohan@cisco.com>,
        Dean Willis
	 <dean.willis@softarmor.com>
Date: Fri, 20 Feb 2004 13:06:23 +0100
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Subject: [Sip] RE: Defining new requests
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I don't know how happy it would be, but at least it wouldn't have to spend time on this relatively minor issue :)

Regards,

Christer

> -----Original Message-----
> From: Gonzalo Camarillo (JO/LMF) 
> Sent: 20. helmikuuta 2004 13:55
> To: Christer Holmberg (JO/LMF)
> Cc: sip; Rohan Mahy; Dean Willis
> Subject: Re: Defining new requests
> 
> 
> Christer,
> 
> would the ITU-T be happier if we log an errata against the INFO RFC 
> saying that UASs can send INFO within early dialogs?
> 
> Gonzalo
> 
> Christer Holmberg (JO/LMF) wrote:
> 
> > Hi Gonzalo,
> > 
> > Thanks for your reply!
> > 
> > 
> >>it seems that some people in the ITU-T are having some issues 
> >>figuring out whether or not certain requests can be sent by the UAS
> > 
> > within an 
> > 
> >>early dialog. The confusion comes from statements like this 
> >>one (taken from the INFO spec):
> >>
> >>"     Unless stated otherwise, the protocol rules for the 
> INFO request
> >>       governing the usage of tags, Route and Record-Route,
> >>       retransmission and reliability, CSeq incrementing and message
> >>       formatting follow those in [1] as defined for the 
> BYE request."
> >>
> >>Since a UAS does not send BYEs within an early dialog, but 
> a non-2xx 
> >>final response instead, to terminate the dialog, people assume that
> > 
> > INFO 
> > 
> >>cannot be sent in this situation either.
> >>
> >>Let's try to keep this in mind and clarify this point when/if 
> >>we define new SIP methods.
> >>
> >>In any event, the clarification to this issue can be found in 
> >>RFC 3398:
> >>
> >>"From the callee to the caller, if a message is received by 
> a gateway
> >>before the call has been answered (i.e., ANM is received) 
> >>it SHOULD be encapsulated in an INFO, provided that this will not 
> >>be the first SIP message sent in the backwards direction (in which 
> >>case it SHOULD be encapsulated in a provisional 1xx response)."
> > 
> > 
> > The problem is that the ITU-T interworking spec doesn't reference
> > RFC3398. The spec reference the INFO RFC, and that is also 
> where I think
> > the usage rules should be defined (not in another spec, 
> RFC3398 in this
> > case, simply using the method).
> > 
> > The INFO RFC, as you said, currently says that INFO 
> "follows the rules
> > of BYE" (whatever that means is another question :), and 
> the rules for
> > BYE specifically says that it must not be sent backward before the
> > INVITE has finished. If the INFO RFC would say "follow the rules of
> > UPDATE" we would not have this problem, since the UPDATE spec
> > specifically allows the use of the method before the INVITE has
> > finished.
> > 
> > Thanks!
> > 
> > Christer Holmberg
> > Ericsson Finland
> 
> 

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Subject: RE: [Sip] SIP instance identifiers
Date: Fri, 20 Feb 2004 10:39:54 -0800
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Thread-Topic: [Sip] SIP instance identifiers
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I tend to agree with Paul (if I read him correctly).

=20

I think that GUID and the original GRID parameter are very different
things because of their properties (and obviously the resultant usage
cases).

Below is my view on the topic.

=20

1. Global Unique ID: GUID IS a very well-known concept. It is used by
many applications outside SIP, but I don't see its direct relation to
SIP. I am not sure whether we need a dedicated SIP placeholder for it.
For PIDF applications, GRUU is enough, in my opinion (see below).

=20

2. GRUU format: GRUU (not GRID or GUID) is a SIP URI and is very SIPish
concept with the properties as defined in the GRUU draft. It IS the "SIP
Instance Identifier". Being a SIP URI, it can be used in any Presence
(e.g. PIDF) applications without changing the defined XML schemas. It
may be beneficial to define a URI parameter that says: "This SIP URI has
GRUU properties". It will eliminate the need for Supported: gruu in
every message and for potential expanding the PIDF documents to include
this clarification.=20

=20

3. GRUU signaling and usage: GRUU can be allocated either by Registrar,
or by UA, or by different means. My point is GRUU identification (i.e.
placement and signaling) during session establishment and usage
(especially outside the Registrar domain) MUST be identical for all
cases. Moreover, it MUST be possible for the Registrar domain to conceal
the information about the allocation method being used.

=20

4. Registration Procedure: We need to standardize the registration
procedure for both basic cases: GRUU allocated by Server (done) and GRUU
allocated by Client. In my opinion, both cases need to be under the
Registration/contact umbrella - which makes them SIP concepts! Whether
GRUU value is preserved during client's/server's reboots - it doesn't
change the concept. From usability point of view, it would be very nice
to keep the value as long as possible. From the outside user point of
view - it doesn't need to know (and doesn't want to know) what the
registration durability of its peers is.

=20

5. Saving Registration Cycles and DB size: By using GRUU we have the
ability to contact a specific instance of SIP AOR. Once we point to a
specific entity, we MAY also need to point to specific sub-entities
inside this GRUU. This will save the registration burden from each of
these sub-entities, since they will be covered by GRUU's registration
and routing. The original GRID parameters allowed for exactly this
functionality. The usage example is a conference factory which generates
foci and doesn't need to dynamically REGISTER each of them in order to
achieve proper routing. BTW, the concept of "internal routing" to
specific logical module inside SIP entity can be generalized, decoupled
from GRUU and be defined as a dedicated SIP header.

=20

Orit.

=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Paul
Kyzivat
Sent: Thursday, February 19, 2004 9:37 PM
To: Cullen Jennings
Cc: Jonathan Rosenberg; sip@ietf.org
Subject: Re: [Sip] SIP instance identifiers

=20

Just to be clear, I am not *advocating* that we need these identifiers=20

any other place. I'm just saying that if there are other uses for a=20

device id, that it won't hurt to use the same id elsewhere.

=20

For the purpose we are discussing, it is important that the id be=20

associated with a registered contact, and since REGISTER permits=20

multiple contacts, it really needs to be a parameter on each contact.

=20

That is really an entirely different application from identifying which=20

device originated a particular message. If there is a need for that,=20

then it needs to be in some header that can be in any message. I don't=20

recommend we do that until we have a need for it.

=20

      Paul


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<div class=3DSection1>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>I tend to agree with Paul (if I read him =
correctly).</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>I think that GUID and the original GRID parameter are very =
different
things because of their properties (and obviously the resultant usage =
cases).</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Below is my view on the topic.</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>1. <u>Global Unique ID:</u> GUID IS a very well-known concept. =
It is
used by many applications outside SIP, but I don't see its direct =
relation to
SIP. I am not sure whether we need a dedicated SIP placeholder for it. =
For PIDF
applications, GRUU is enough, in my opinion (see =
below).</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>2. <u>GRUU format:</u> GRUU (not GRID or GUID) is a SIP URI and =
is very
SIPish concept with the properties as defined in the GRUU draft. It IS =
the
&quot;SIP Instance Identifier&quot;. Being a SIP URI, it can be used in =
any Presence
(e.g. PIDF) applications without changing the defined XML schemas. It =
may be beneficial
to define a URI parameter that says: &quot;This SIP URI has GRUU
properties&quot;. It will eliminate the need for Supported: gruu in =
every
message and for potential expanding the PIDF documents to include this =
clarification.
</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>3. <u>GRUU signaling and usage:</u> GRUU can be allocated either =
by
Registrar, or by UA, or by different means. My point is GRUU =
identification (i.e.
placement and signaling) during session establishment and usage =
(especially
outside the Registrar domain) MUST be identical for all cases. Moreover, =
it
MUST be possible for the Registrar domain to conceal the information =
about the
allocation method being used.</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>4. <u>Registration Procedure:</u> We need to standardize the
registration procedure for both basic cases: GRUU allocated by Server =
(done)
and GRUU allocated by Client. In my opinion, both cases need to be under =
the Registration/contact
umbrella - which makes them SIP concepts! Whether GRUU value is =
preserved during
client's/server's reboots - it doesn't change the concept. From =
usability point
of view, it would be very nice to keep the value as long as possible. =
From the outside
user point of view - it doesn't need to know (and doesn't want to know) =
what
the registration durability of its peers is.</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>5. <u>Saving Registration Cycles and DB size</u>: By using GRUU =
we have
the ability to contact a specific instance of SIP AOR. Once we point to =
a
specific entity, we MAY also need to point to specific sub-entities =
inside this
GRUU. This will save the registration burden from each of these =
sub-entities,
since they will be covered by GRUU&#8217;s registration and routing. The
original GRID parameters allowed for exactly this functionality. The =
usage
example is a conference factory which generates foci and doesn&#8217;t =
need to dynamically
REGISTER each of them in order to achieve proper routing. BTW, the =
concept of &#8220;internal
routing&#8221; to specific logical module inside SIP entity can be =
generalized,
decoupled from GRUU and be defined as a dedicated SIP =
header.</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Orit.</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>-----Original Message-----<br>
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Paul =
Kyzivat<br>
Sent: Thursday, February 19, 2004 9:37 PM<br>
To: Cullen Jennings<br>
Cc: Jonathan Rosenberg; sip@ietf.org<br>
Subject: Re: [Sip] SIP instance identifiers</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Just to be clear, I am not *advocating* that we need these =
identifiers </span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>any other place. I'm just saying that if there are other uses =
for a </span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>device id, that it won't hurt to use the same id =
elsewhere.</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>For the purpose we are discussing, it is important that the id =
be </span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>associated with a registered contact, and since REGISTER permits =
</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>multiple contacts, it really needs to be a parameter on each =
contact.</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>That is really an entirely different application from =
identifying which
</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>device originated a particular message. If there is a need for =
that, </span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>then it needs to be in some header that can be in any message. I =
don't </span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>recommend we do that until we have a need for =
it.</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;</span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul</span></font></p>

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_______________________________________________
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From exim@www1.ietf.org  Fri Feb 20 16:04:45 2004
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From: Anil Bollineni <ABollineni@netscreen.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Date: Fri, 20 Feb 2004 13:00:44 -0800
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Hi All,

            For any SIP message, if UA or Proxy sending message than other
than from port 5060, then it should be included in SIP from URI. For
example, if the Register message comes from 202.10.10.20:2218, the From
header should look like From: sip:joe@202.10.10.20:2218. Is it MUST if the
port is different than port 5060 then it should appear in the From header?

 

Thanks in Advance,

Anil

This email contains material that is confidential. The content of this email
is for the sole use of the intended recipient(s). Any review or distribution
by persons other than the intended recipient(s) without the express
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are not the intended recipient, please contact the sender and delete/destroy
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; For
any SIP message, if UA or Proxy sending message than other than from port 5060,
then it should be included in SIP from URI. For example, if the Register
message comes from 202.10.10.20:2218, the From header should look like From:
sip:joe@202.10.10.20:2218. Is it MUST if the port is different than port 5060
then it should appear in the From header?</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Thanks in Advance,</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Anil</span></font></p>

<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>This
email contains material that is confidential. The content of this email is for
the sole use of the intended recipient(s). Any review or distribution by
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Hi All,

 

      I understand from the following statement in RFC, that From header
SHOULD not contain IP addresses/ports from whom the request has come. If it
is so, I can't assume that the information is not used to assume from where
(IP address/port) the request has come. Appreciate your response to clarify
the point.

 

  As such, it is very important that the From URI not contain IP addresses
or the FQDN

   of the host on which the UA is running, since these are not logical

   names.

 

Thank you,

Anil

 

This email contains material that is confidential. The content of this email
is for the sole use of the intended recipient(s). Any review or distribution
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are not the intended recipient, please contact the sender and delete/destroy
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<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>Hi All,</span></font></p>

<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;</span></font></p>

<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I
understand from the following statement in RFC, that From header SHOULD not
contain IP addresses/ports from whom the request has come. If it is so, I can't
assume that the information is not used to assume from where (IP address/port)
the request has come. Appreciate your response to clarify the point.</span></font></p>

<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;</span></font></p>

<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; As such, it is very
important that the From URI not contain IP addresses or the FQDN</span></font></p>

<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp; of the host on
which the UA is running, since these are not logical</span></font></p>

<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp; names.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Thank you,</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Anil</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>This
email contains material that is confidential. The content of this email is for
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accuracy or completeness of third party materials or information.</span></font></p>

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To: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>,
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        sip
	 <sip@ietf.org>
Cc: Rohan Mahy <rohan@cisco.com>, Dean Willis <dean.willis@softarmor.com>
Subject: RE: [Sip] RE: Defining new requests
Date: Sun, 22 Feb 2004 23:16:29 -0000
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While RFC 3398 may appear to clarify this, it is not in the scope of RFC 3398 to do so.

RFC 3398 defines only the interworking, not the SIP protocol associated with it.

I would suggest a bug fix is tables for the INFO RFC.

regards

Keith


> -----Original Message-----
> From: Christer Holmberg (JO/LMF) 
> [mailto:christer.holmberg@ericsson.com]
> Sent: 20 February 2004 11:40
> To: Gonzalo Camarillo (JO/LMF); sip
> Cc: Rohan Mahy; Dean Willis
> Subject: [Sip] RE: Defining new requests
> 
> 
> 
> Hi Gonzalo,
> 
> Thanks for your reply!
> 
> >it seems that some people in the ITU-T are having some issues 
> >figuring out whether or not certain requests can be sent by 
> the UAS within an 
> >early dialog. The confusion comes from statements like this 
> >one (taken from the INFO spec):
> > 
> > "     Unless stated otherwise, the protocol rules for the 
> INFO request
> >        governing the usage of tags, Route and Record-Route,
> >        retransmission and reliability, CSeq incrementing and message
> >        formatting follow those in [1] as defined for the 
> BYE request."
> > 
> >Since a UAS does not send BYEs within an early dialog, but a non-2xx 
> >final response instead, to terminate the dialog, people 
> assume that INFO 
> >cannot be sent in this situation either.
> > 
> >Let's try to keep this in mind and clarify this point when/if 
> >we define new SIP methods.
> > 
> >In any event, the clarification to this issue can be found in 
> >RFC 3398:
> > 
> >"From the callee to the caller, if a message is received by a gateway
> >before the call has been answered (i.e., ANM is received) 
> >it SHOULD be encapsulated in an INFO, provided that this will not 
> >be the first SIP message sent in the backwards direction (in which 
> >case it SHOULD be encapsulated in a provisional 1xx response)."
> 
> The problem is that the ITU-T interworking spec doesn't 
> reference RFC3398. The spec reference the INFO RFC, and that 
> is also where I think the usage rules should be defined (not 
> in another spec, RFC3398 in this case, simply using the method).
> 
> The INFO RFC, as you said, currently says that INFO "follows 
> the rules of BYE" (whatever that means is another question 
> :), and the rules for BYE specifically says that it must not 
> be sent backward before the INVITE has finished. If the INFO 
> RFC would say "follow the rules of UPDATE" we would not have 
> this problem, since the UPDATE spec specifically allows the 
> use of the method before the INVITE has finished.
> 
> Thanks!
> 
> Christer Holmberg
> Ericsson Finland
> 
> This communication is confidential and intended solely for 
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> been sent to you in error, please notify the sender by 
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> 
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> basis that we are not liable for any such corruption, 
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> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
CC: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>,
        sip <sip@ietf.org>, Rohan Mahy <rohan@cisco.com>,
        Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Re: Defining new requests
References: <F8EFC4B4A8C016428BC1F589296D4FBF07B07A69@esealnt630.al.sw.ericsson.se> <4035F586.1000405@ericsson.com>
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The ITU-T (Study Group 11, to be exact) had a different issue with INFO: that the 
RFC depends on RFC 2543 rather than 3261.  As a result, they picked up the content 
of RFC 2976 as Annex C.3 of the ISUP interworking specification Q.1912.5.  If RFC 
2976 ever gets updated Annex C.3 can be deprecated.  In the meantime, this INFO 
question means we have some maintenance to perform.  I think the best solution 
will be to take Gonzalo's suggestion and say that INFO follows the rules for 
UPDATE.  We already have a normative dependency on the UPDATE RFC.

Gonzalo Camarillo wrote:

> Christer,
> 
> would the ITU-T be happier if we log an errata against the INFO RFC 
> saying that UASs can send INFO within early dialogs?
> 
> Gonzalo
> 
> Christer Holmberg (JO/LMF) wrote:
> 
>> Hi Gonzalo,
>>
>> Thanks for your reply!
>>
>>
>>> it seems that some people in the ITU-T are having some issues 
>>> figuring out whether or not certain requests can be sent by the UAS
>>
>>
>> within an
>>
>>> early dialog. The confusion comes from statements like this one 
>>> (taken from the INFO spec):
>>>
>>> "     Unless stated otherwise, the protocol rules for the INFO request
>>>       governing the usage of tags, Route and Record-Route,
>>>       retransmission and reliability, CSeq incrementing and message
>>>       formatting follow those in [1] as defined for the BYE request."
>>>
>>> Since a UAS does not send BYEs within an early dialog, but a non-2xx 
>>> final response instead, to terminate the dialog, people assume that
>>
>>
>> INFO
>>
>>> cannot be sent in this situation either.
>>>
>>> Let's try to keep this in mind and clarify this point when/if we 
>>> define new SIP methods.
>>>
>>> In any event, the clarification to this issue can be found in RFC 3398:
>>>
>>> "From the callee to the caller, if a message is received by a gateway
>>> before the call has been answered (i.e., ANM is received) it SHOULD 
>>> be encapsulated in an INFO, provided that this will not be the first 
>>> SIP message sent in the backwards direction (in which case it SHOULD 
>>> be encapsulated in a provisional 1xx response)."
>>
>>
>>
>> The problem is that the ITU-T interworking spec doesn't reference
>> RFC3398. The spec reference the INFO RFC, and that is also where I think
>> the usage rules should be defined (not in another spec, RFC3398 in this
>> case, simply using the method).
>>
>> The INFO RFC, as you said, currently says that INFO "follows the rules
>> of BYE" (whatever that means is another question :), and the rules for
>> BYE specifically says that it must not be sent backward before the
>> INVITE has finished. If the INFO RFC would say "follow the rules of
>> UPDATE" we would not have this problem, since the UPDATE spec
>> specifically allows the use of the method before the INVITE has
>> finished.
>>
>> Thanks!
>>
>> Christer Holmberg
>> Ericsson Finland
> 
> 
> 
> 
> 
> This communication is confidential and intended solely for the 
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> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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> Use sipping@ietf.org for new developments on the application of sip
> 

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From: Anil Bollineni <ABollineni@netscreen.com>
To: "'xiao'tong liang'" <boyliangxt@yahoo.com>
Cc: "'sip@ietf.org'" <sip@ietf.org>,
        "'sip-implementors@cs.columbia.edu'"
	 <sip-implementors@cs.columbia.edu>
Subject: RE: [Sip] From Header
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The Via header tells that IP address/port where the response should be sent.
It does not specify from where the request has been come, as the ports for
sending request and receiving responses can be different. I think the only
way we can tell is by seeing the sending ip and port in IP header of the
packet. Please correct if I am wrong.

Thanks,
Anil


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-----Original Message-----
From: xiao'tong liang [mailto:boyliangxt@yahoo.com] 
Sent: Sunday, February 22, 2004 6:07 PM
To: Anil Bollineni
Subject: Re: [Sip] From Header

Via header.

--
xiaotong


--- Anil Bollineni <ABollineni@netscreen.com> wrote:
> Hi All,
> 
>  
> 
>       I understand from the following statement in
> RFC, that From header
> SHOULD not contain IP addresses/ports from whom the
> request has come. If it
> is so, I can't assume that the information is not
> used to assume from where
> (IP address/port) the request has come. Appreciate
> your response to clarify
> the point.
> 
>  
> 
>   As such, it is very important that the From URI
> not contain IP addresses
> or the FQDN
> 
>    of the host on which the UA is running, since
> these are not logical
> 
>    names.
> 
>  
> 
> Thank you,
> 
> Anil
> 
>  
> 
> This email contains material that is confidential.
> The content of this email
> is for the sole use of the intended recipient(s).
> Any review or distribution
> by persons other than the intended recipient(s)
> without the express
> permission of NetScreen Technologies, Inc. is
> strictly prohibited. If you
> are not the intended recipient, please contact the
> sender and delete/destroy
> all copies of this email and any related
> attachments. NetScreen does not
> guarantee the accuracy or completeness of third
> party materials or
> information.
> 
>  
> 
> 


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Orit Levin wrote:
> I tend to agree with Paul (if I read him correctly).

I think maybe you don't read me as intended, because I don't entirely=20
agree with you here. (At least if I understand you.)

> I think that GUID and the original GRID parameter are very different=20
> things because of their properties (and obviously the resultant usage=20
> cases).

If you are talking about the grid=3Dxxx parameter that Jonathan has=20
proposed be used with GRUUs, then I don't think it has any signficance=20
to this discussion at all.

> Below is my view on the topic.
>=20
> 1. _Global Unique ID:_ GUID IS a very well-known concept. It is used by=
=20
> many applications outside SIP, but I don't see its direct relation to=20
> SIP. I am not sure whether we need a dedicated SIP placeholder for it.=20
> For PIDF applications, GRUU is enough, in my opinion (see below).

I don't know that anyone is talking about this. But we are talking about=20
a slightly more restricted concept - a Device ID. Conceivably a device=20
could use a guid as its device id, but that is largely irrelevant to=20
this discussion.

> 2. _GRUU format:_ GRUU (not GRID or GUID) is a SIP URI and is very=20
> SIPish concept with the properties as defined in the GRUU draft. It IS=20
> the "SIP Instance Identifier".

The last sentence is debatable. It is *an* endpoint instance identifier=20
that is sip routable. But it isn't necessarily defined for as long as=20
the device(s) it references are. And it isn't necessarily bound to the=20
same device at all times.

I think you may have missed the point that the device id is distinct=20
from the GRUU, though dependent upon it. Many different GRUUs may share=20
the same device id. I am proposing that the device id be represented as=20
a URI, but not necessarily (generally not) a sip uri. Generally I expect=20
a URN to be used.

> 3. _GRUU signaling and usage:_ GRUU can be allocated either by=20
> Registrar, or by UA, or by different means. My point is GRUU=20
> identification (i.e. placement and signaling) during session=20
> establishment and usage (especially outside the Registrar domain) MUST=20
> be identical for all cases. Moreover, it MUST be possible for the=20
> Registrar domain to conceal the information about the allocation method=
=20
> being used.
>=20
> 4. _Registration Procedure:_ We need to standardize the registration=20
> procedure for both basic cases: GRUU allocated by Server (done) and GRU=
U=20
> allocated by Client. In my opinion, both cases need to be under the=20
> Registration/contact umbrella - which makes them SIP concepts! Whether=20
> GRUU value is preserved during client's/server's reboots - it doesn't=20
> change the concept. From usability point of view, it would be very nice=
=20
> to keep the value as long as possible. >From the outside user point of=20
> view - it doesn't need to know (and doesn't want to know) what the=20
> registration durability of its peers is.

Jonathan should probably chime in here. I believe he argues that=20
typically a UA can't tell whether it can generate its own GRUU, so they=20
should always be assigned by registrars.

> 5. _Saving Registration Cycles and DB size_: By using GRUU we have the=20
> ability to contact a specific instance of SIP AOR. Once we point to a=20
> specific entity, we MAY also need to point to specific sub-entities=20
> inside this GRUU. This will save the registration burden from each of=20
> these sub-entities, since they will be covered by GRUU=92s registration=
=20
> and routing. The original GRID parameters allowed for exactly this=20
> functionality. The usage example is a conference factory which generate=
s=20
> foci and doesn=92t need to dynamically REGISTER each of them in order t=
o=20
> achieve proper routing. BTW, the concept of =93internal routing=94 to=20
> specific logical module inside SIP entity can be generalized, decoupled=
=20
> from GRUU and be defined as a dedicated SIP header.

Maybe it can be decoupled from GRUU, but I think it must be coupled to a=20
URL, not a message. So it must be a URL formatting convention and/or a=20
url parameter. So maybe you could generalize the grid parameter to apply=20
to all sip/sips urls, rather than coupling it to gruu. I haven't thought=20
thru where that might be useful.

	Paul


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From exim@www1.ietf.org  Mon Feb 23 16:10:41 2004
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I find the newest version of this draft troubling. It feels to me like 
it has selected a slippery slope and is already accelerating rapidly 
downward.

My major concern is with the approach described in section 4:

    Alternatively, the Refer-To URI could be a Universal Resource Name
    (URN) [21] which could describe a particular operation such as Hold
    or Retrieve.  Combined with the dialog-identifiers of an existing
    session conveyed as parameters of the Refer-To header, this would
    permit explicit operations which do not need additional parameters or
    handle more than a single session.  For example, the following could
    represent a Hold operation of a session with the Call-ID "123":

    	Refer-To: <urn:ietf:params:sip:remotecc:hold>
    	 ;call-id=123;remote-tag=aaa;local-tag=bbb

Further examples are given abstractly, for example:

     --------------------------------
     | Remote Call Control Body     |
     | transfer                     |
     |   FirstCall=                 |
     |     sip:line1@192.168.0.5    |
     |     call-id:456              |
     |     remote-tag=uvw           |
     |     local-tag=xyz            |
     |   SecondCall=                |
     |     sip:cathy@example.net    |
     |   other parameters           |
     --------------------------------

My concern is that this appears to be defining an explicit feature 
invocation model, with specific features that have names and parameters. 
While it could conceivably be in scope for IETF to define things to this 
level, in the past the sip wg has avoided doing so. If that philosophy 
is to change, then I think we must consider all the implications of 
doing so.

I think every feature of this sort will require a considerable effort to 
specify in sufficient detail so that interoperation is assured. It may 
be that we are already in that situation with REFER and just haven't 
realized it yet, but I am inclined to think this makes that situation worse.

I don't think the features as described go far enough to define their 
semantics. In particular, I think it is difficult to define the meaning 
of these features without addressing media handling and the management 
of explicit UI components of the endpoint. For instance, what does 
MakeCall imply about the UI and media handling on the device that is to 
make a call? Is the user alerted first, with the device awaiting an 
off-hook before completing the call? Or are the microphone and speaker 
in the handset activated for media even though the phone is on-hook? If 
the device is capable of using multiple media, which are enabled in the 
call? The questions abound.

More generally, must all endpoints that support extended-refer support 
every feature that might be named in a Remote Call Control Body? If so, 
how do we evolve the set of features? If not, how does a UA determine 
what features it can invoke, and how?

The approach of simply enhancing Refer-To with header parameters that 
simplify the construction of complex referred operations is more 
bounded, and less likely to present issues in the future. Of course it 
may or may not be sufficient for the kinds of call control people may 
want to do remotely.

	Paul


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Subject: RE: [Sip] SIP instance identifiers
Date: Mon, 23 Feb 2004 13:19:15 -0800
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From: "Orit Levin" <oritl@microsoft.com>
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Whatever the overlap in our views is, what is in the scope of this
discussion and what is out of the scope - is a point of view based on
the applications each of us is familiar with.

If we=20
- have both the GRUU concept and the GRID concept as URI parameters
- allow a user to allocated its own device identifier (which needs to be
"globally" unique within its AOR scope only)
- allow a Registrar to override the user allocated GRUU and replace it
with Registrar generated value (if required according to local policies)

it will satisfy the requirements of the applications that I am familiar
with.

Orit.=20

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Monday, February 23, 2004 11:36 AM
To: Orit Levin
Cc: sip@ietf.org
Subject: Re: [Sip] SIP instance identifiers



Orit Levin wrote:
> I tend to agree with Paul (if I read him correctly).

I think maybe you don't read me as intended, because I don't entirely
agree with you here. (At least if I understand you.)

> I think that GUID and the original GRID parameter are very different=20
> things because of their properties (and obviously the resultant usage=20
> cases).

If you are talking about the grid=3Dxxx parameter that Jonathan has
proposed be used with GRUUs, then I don't think it has any signficance
to this discussion at all.

> Below is my view on the topic.
>=20
> 1. _Global Unique ID:_ GUID IS a very well-known concept. It is used=20
> by many applications outside SIP, but I don't see its direct relation=20
> to SIP. I am not sure whether we need a dedicated SIP placeholder for
it.
> For PIDF applications, GRUU is enough, in my opinion (see below).

I don't know that anyone is talking about this. But we are talking about
a slightly more restricted concept - a Device ID. Conceivably a device
could use a guid as its device id, but that is largely irrelevant to
this discussion.

> 2. _GRUU format:_ GRUU (not GRID or GUID) is a SIP URI and is very=20
> SIPish concept with the properties as defined in the GRUU draft. It IS

> the "SIP Instance Identifier".

The last sentence is debatable. It is *an* endpoint instance identifier
that is sip routable. But it isn't necessarily defined for as long as
the device(s) it references are. And it isn't necessarily bound to the
same device at all times.

I think you may have missed the point that the device id is distinct
from the GRUU, though dependent upon it. Many different GRUUs may share
the same device id. I am proposing that the device id be represented as
a URI, but not necessarily (generally not) a sip uri. Generally I expect
a URN to be used.

> 3. _GRUU signaling and usage:_ GRUU can be allocated either by=20
> Registrar, or by UA, or by different means. My point is GRUU=20
> identification (i.e. placement and signaling) during session=20
> establishment and usage (especially outside the Registrar domain) MUST

> be identical for all cases. Moreover, it MUST be possible for the=20
> Registrar domain to conceal the information about the allocation=20
> method being used.
>=20
> 4. _Registration Procedure:_ We need to standardize the registration=20
> procedure for both basic cases: GRUU allocated by Server (done) and=20
> GRUU allocated by Client. In my opinion, both cases need to be under=20
> the Registration/contact umbrella - which makes them SIP concepts!=20
> Whether GRUU value is preserved during client's/server's reboots - it=20
> doesn't change the concept. From usability point of view, it would be=20
> very nice to keep the value as long as possible. >From the outside=20
> user point of view - it doesn't need to know (and doesn't want to=20
> know) what the registration durability of its peers is.

Jonathan should probably chime in here. I believe he argues that
typically a UA can't tell whether it can generate its own GRUU, so they
should always be assigned by registrars.

> 5. _Saving Registration Cycles and DB size_: By using GRUU we have the

> ability to contact a specific instance of SIP AOR. Once we point to a=20
> specific entity, we MAY also need to point to specific sub-entities=20
> inside this GRUU. This will save the registration burden from each of=20
> these sub-entities, since they will be covered by GRUU's registration=20
> and routing. The original GRID parameters allowed for exactly this=20
> functionality. The usage example is a conference factory which=20
> generates foci and doesn't need to dynamically REGISTER each of them=20
> in order to achieve proper routing. BTW, the concept of "internal=20
> routing" to specific logical module inside SIP entity can be=20
> generalized, decoupled from GRUU and be defined as a dedicated SIP
header.

Maybe it can be decoupled from GRUU, but I think it must be coupled to a
URL, not a message. So it must be a URL formatting convention and/or a
url parameter. So maybe you could generalize the grid parameter to apply
to all sip/sips urls, rather than coupling it to gruu. I haven't thought
thru where that might be useful.

	Paul


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Subject: RE: [Sip] From Header
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Hi Anil,

If I understand you correctly then I think you want to know the sender's
IP and port number. This information can be obtaine from the SDP. Via
header gives information of Proxy not of sender's.

Ranjit

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Anil
Bollineni
Sent: Tuesday, February 24, 2004 12:43 AM
To: 'xiao'tong liang'
Cc: 'sip@ietf.org'; 'sip-implementors@cs.columbia.edu'
Subject: RE: [Sip] From Header


The Via header tells that IP address/port where the response should be
sent. It does not specify from where the request has been come, as the
ports for sending request and receiving responses can be different. I
think the only way we can tell is by seeing the sending ip and port in
IP header of the packet. Please correct if I am wrong.

Thanks,
Anil


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-----Original Message-----
From: xiao'tong liang [mailto:boyliangxt@yahoo.com]=20
Sent: Sunday, February 22, 2004 6:07 PM
To: Anil Bollineni
Subject: Re: [Sip] From Header

Via header.

--
xiaotong


--- Anil Bollineni <ABollineni@netscreen.com> wrote:
> Hi All,
>=20
> =20
>=20
>       I understand from the following statement in
> RFC, that From header
> SHOULD not contain IP addresses/ports from whom the
> request has come. If it
> is so, I can't assume that the information is not
> used to assume from where
> (IP address/port) the request has come. Appreciate
> your response to clarify
> the point.
>=20
> =20
>=20
>   As such, it is very important that the From URI
> not contain IP addresses
> or the FQDN
>=20
>    of the host on which the UA is running, since
> these are not logical
>=20
>    names.
>=20
> =20
>=20
> Thank you,
>=20
> Anil
>=20
> =20
>=20
> This email contains material that is confidential.
> The content of this email
> is for the sole use of the intended recipient(s).
> Any review or distribution
> by persons other than the intended recipient(s)
> without the express
> permission of NetScreen Technologies, Inc. is
> strictly prohibited. If you
> are not the intended recipient, please contact the
> sender and delete/destroy
> all copies of this email and any related
> attachments. NetScreen does not
> guarantee the accuracy or completeness of third
> party materials or
> information.
>=20
> =20
>=20
>=20


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Date: Tue, 24 Feb 2004 09:32:52 +0200
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Tom Taylor <taylor@nortelnetworks.com>
CC: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>,
        sip <sip@ietf.org>, Rohan Mahy <rohan@cisco.com>,
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Subject: Re: [Sip] Re: Defining new requests
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Tom,

what is the timeframe here? As you know, we have had discussions in the 
past about either re-defining INFO so that it can only carry PSTN 
telephony signalling, or defining INFO packages. We could re-start those 
discussions, and once we reach a conclusion, clarify the rules to send 
INFO from the UAS within early dialogs in the same spec...

If the ITU is in a hurry, other solutions may be better, like logging an 
errata against the INFO RFC.

Gonzalo

Tom Taylor wrote:
> The ITU-T (Study Group 11, to be exact) had a different issue with INFO: 
> that the RFC depends on RFC 2543 rather than 3261.  As a result, they 
> picked up the content of RFC 2976 as Annex C.3 of the ISUP interworking 
> specification Q.1912.5.  If RFC 2976 ever gets updated Annex C.3 can be 
> deprecated.  In the meantime, this INFO question means we have some 
> maintenance to perform.  I think the best solution will be to take 
> Gonzalo's suggestion and say that INFO follows the rules for UPDATE.  We 
> already have a normative dependency on the UPDATE RFC.
> 
> Gonzalo Camarillo wrote:
> 
>> Christer,
>>
>> would the ITU-T be happier if we log an errata against the INFO RFC 
>> saying that UASs can send INFO within early dialogs?
>>
>> Gonzalo
>>
>> Christer Holmberg (JO/LMF) wrote:
>>
>>> Hi Gonzalo,
>>>
>>> Thanks for your reply!
>>>
>>>
>>>> it seems that some people in the ITU-T are having some issues 
>>>> figuring out whether or not certain requests can be sent by the UAS
>>>
>>>
>>>
>>> within an
>>>
>>>> early dialog. The confusion comes from statements like this one 
>>>> (taken from the INFO spec):
>>>>
>>>> "     Unless stated otherwise, the protocol rules for the INFO request
>>>>       governing the usage of tags, Route and Record-Route,
>>>>       retransmission and reliability, CSeq incrementing and message
>>>>       formatting follow those in [1] as defined for the BYE request."
>>>>
>>>> Since a UAS does not send BYEs within an early dialog, but a non-2xx 
>>>> final response instead, to terminate the dialog, people assume that
>>>
>>>
>>>
>>> INFO
>>>
>>>> cannot be sent in this situation either.
>>>>
>>>> Let's try to keep this in mind and clarify this point when/if we 
>>>> define new SIP methods.
>>>>
>>>> In any event, the clarification to this issue can be found in RFC 3398:
>>>>
>>>> "From the callee to the caller, if a message is received by a gateway
>>>> before the call has been answered (i.e., ANM is received) it SHOULD 
>>>> be encapsulated in an INFO, provided that this will not be the first 
>>>> SIP message sent in the backwards direction (in which case it SHOULD 
>>>> be encapsulated in a provisional 1xx response)."
>>>
>>>
>>>
>>>
>>> The problem is that the ITU-T interworking spec doesn't reference
>>> RFC3398. The spec reference the INFO RFC, and that is also where I think
>>> the usage rules should be defined (not in another spec, RFC3398 in this
>>> case, simply using the method).
>>>
>>> The INFO RFC, as you said, currently says that INFO "follows the rules
>>> of BYE" (whatever that means is another question :), and the rules for
>>> BYE specifically says that it must not be sent backward before the
>>> INVITE has finished. If the INFO RFC would say "follow the rules of
>>> UPDATE" we would not have this problem, since the UPDATE spec
>>> specifically allows the use of the method before the INVITE has
>>> finished.
>>>
>>> Thanks!
>>>
>>> Christer Holmberg
>>> Ericsson Finland
>>


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From exim@www1.ietf.org  Tue Feb 24 02:56:38 2004
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Thread-Topic: RPORT and TCP
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From: "Sarit Galanos" <Sarit@radvision.com>
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Subject: [Sip] RPORT and TCP
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Hi,=20
RFC 3581-Symmetric Response Routing specifies that the rport parameter =
should be used only if the transport is unreliable.

"When a server attempts to send a response, it examines the topmost
   Via header field value of that response.  If the "sent-protocol"
   component indicates an unreliable unicast transport protocol, such as
   UDP, and there is no "maddr" parameter, but there is both a
   "received" parameter and an "rport" parameter, the response MUST be
   sent to the IP address listed in the "received" parameter, and the
   port in the "rport" parameter.  The response MUST be sent from the
   same address and port that the corresponding request was received on.
   This effectively adds a new processing step between bullets two and
   three in Section 18.2.2 of SIP [1]."

As I see it, the same problem that rport solves in UDP can happen in =
TCP.
This can happen if the connection is broken between the request and the =
response.

If for example the Via header in the request is:

Via: SIP/2.0/TCP proxy.example.com;branch=3Dz9hG4bKkjsh77

and proxy.example.com was resolved to a port different then 5060, the =
server will not be able to=20
establish a new connection in order to send the response since 5060 will =
be used.

Why do we have the UDP limitation in the standard?
How can such a problem be solved.

Regards,
Sarit.

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Subject: RE: [Sip-implementors] RE: [Sip] From Header
Date: Tue, 24 Feb 2004 10:42:26 +0100
Message-ID: <FE03AFC4B33E7447979123987BD65F4501883936@exchange.hotsip.com>
Thread-Topic: [Sip-implementors] RE: [Sip] From Header
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From: "Christian Jansson" <christian.jansson@hotsip.com>
To: "Anil Bollineni" <ABollineni@netscreen.com>,
        "xiao'tong liang" <boyliangxt@yahoo.com>
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> I think the only way we can tell=20
> is by seeing the sending ip and port in IP header of the=20
> packet. Please correct if I am wrong.

Correct.

/ Christian Jansson, Hotsip R&D

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Subject: RE: [Sip-implementors] RE: [Sip] From Header
Date: Tue, 24 Feb 2004 10:49:07 +0100
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From: "Christian Jansson" <christian.jansson@hotsip.com>
To: <ranjit.avasarala@wipro.com>, <ABollineni@netscreen.com>,
        <boyliangxt@yahoo.com>
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Ranjit wrote:
> If I understand you correctly then I think you want to know=20
> the sender's IP and port number. This information can be=20
> obtaine from the SDP.

No the sender address and port should not be obtained from the SDP.
The IP and port found in the SDP only tells the receiver where to
send media, nothing else. The IP in the SDP does not have to be the
same IP as from where the SIP signaling is done. The port in the SDP
will probably always differ from the port from where the SIP
signaling is done (I don't think any implementation would like
to demux SIP and media on the same port)

/ Christian Jansson, Hotsip R&D



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To: Cullen Jennings <fluffy@cisco.com>
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Subject: Re: [Sip] Binary Encoding for SIP
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>
>
>So I take if you would be happy to see is say that things SHOULD send binary
>but need to be prepared to receive anything.
>  
>

Yep.

Bryan.

-- 
Bryan Thale
Networks & Infrastructure Research, Motorola Labs
mailto:bryan.thale@motorola.com



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Subject: [Sip] On creating SIP services using Java applet encoding.
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Hi
  I am recent to the group and have been reading and getting interested 
about encodings in SIP (e.g
MIME encoding proposal postings by Cullen).

I also noted that Henning in one of his papers (Oct 2000, IEEE Comm 
magazine),
refers to a work where Java applets were proposed to be included in SIP 
requests for
service creation.

 I am interested if there is any progress made in this specific aspect? 
In fact, encoding
executable content on SIP requests can be powerful abstraction and can 
be usefully utized for
a lot of other things (not only service creation). But it can also 
create otentially serious security issues...
(This is also somewhat similar to binary MIME encodings in SIP, in which 
case the question arises how
such encodings are interpreted.)

 I am just wondering if any work is done on this..

Thanks & Regards, Gokul

Pl post reply to the MG and not to me. Thx.


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Subject: [Sip] SIP Agenda, IETF 59
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Hyperlinked version posted at:
http://www.softarmor.com/sipwg/meets/ietf59/agenda.html

Please try and get any slides in well in advance.

Note that GRUU-related stuff appears on both SIPPING and SIP agendas. 
The idea is to discuss requirements in SIPPING and mechanism in SIP.


Monday, March 1, 2002 1300-1500 Crystal 1/2

1300 Status and Charter Milestones, Working Model
    Chairs
1320 Request History
    Mary Barnes
    draft-ietf-sip-history-info-02.txt
1330 VoiceMail URIs
     Cullen Jennings
     draft-jennings-sip-voicemail-uri-01.txt
     draft-jennings-sip-voicemail-uri-01.html
1340 Wrapping up MIME Body Part Usage
     Cullen Jennings
     draft-jennings-sip-mime-01.txt
     draft-jennings-sip-mime-01.html
1350 Policy URI and Purpose
     Hisham Khartabil
     draft-khartabil-sip-policy-uri-call-info-purpose-00
1400 Authentication Issues
     draft-khartabil-sip-auth-analysis-00.txt
  1410 Non-Invite Transactions
     Robert Sparks
    draft-sparks-sip-nit-problems-00.txt
    draft-sparks-sip-nit-actions-00.txt
    draft-sparks-sip-nit-future-00.txt
1425 GRUU and Device ID Mechanisms
     Jonathan Rosenberg, Brian Stucker and Cullen Jennings
     draft-ietf-sip-gruu-01.txt
    draft-stucker-sip-guid-00.txt
    draft-jennings-sipping-instance-id-00.txt [html]
1500 Close

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Subject: [Sip] changes in session timer
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Folks,

I submitted a revision of the session timer, based on comments received 
from IESG and subsequent list discussion. For some reason, this 
submission was not processed by the I-D editor (I'm working to correct 
that). In the mean time, you can pick it up at:

http://www.jdrosen.net/papers/draft-ietf-sip-session-timer-14.txt


There are a few significant changes:

* Formerly, if a session refresh timed out, the UA waited until 1/3 of 
the session interval remained before sending a BYE. This contradicts 
RFC3261 which basically says to send a BYE immediately. Session timer 
now references that behavior.

* There is now an absolute minimum session timer of 90 seconds. This is 
basically enough time for completion of a SIP BYE transaction after half 
of the session has expired. The Min-SE header field now has a default 
value of 90 seconds when not present. This will preclude usage of 
session timer for really fine grained refreshes, say on the order of 5 
to 10 seconds. I fear that some people may be trying to do some kind of 
nat traversal or something with session timer - this would rule that out 
formally (it was never the intended usage).

* The default for session-expires, when not present, used to be 
"infinity". However, thats not really right. Really, it means 
"undefined", and session expiration is then up to local policy of the 
proxy. Using infinity would mean that a proxy could never clean up state.


Please comment ASAP if you have any problems with these changes. We'd 
like to get session timer moving along.

-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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Hi Rohan,



ext Rohan Mahy wrote:
> Hi Folks,
> 
> I just noticed a minor problem with publish ; it defines a new response 
> code:
> 
>     412 Precondition Failed
> 
> which is confusingly similar to the existing response code:
> 
>     580 Precondition Failure
> 
> I propose changing The reason phrase for 412 to "Publication Condition 
> Failed"

Good point. A question though: since we're borrowing this response code 
from HTTP, how strictly do we need to keep the reason phrases aligned? I 
assume it's enough that the semantics are the same, and the default 
reason phrase can therefore be different.

Also, to generalize, would:

	412 (Conditional Request Failed)

be better?

Cheers,
Aki

> 
> thanks,
> -rohan
> 
> 
> On Feb 16, 2004, at 12:41 PM, Internet-Drafts@ietf.org wrote:
> 
>> A New Internet-Draft is available from the on-line Internet-Drafts 
>> directories.
>> This draft is a work item of the Session Initiation Protocol Working 
>> Group of the IETF.
>>
>>     Title        : Session Initiation Protocol (SIP) Extension for 
>> Event State
>>                               Publication
>>     Author(s)    : A. Niemi
>>     Filename    : draft-ietf-sip-publish-03.txt
>>     Pages        : 40
>>     Date        : 2004-2-16
>>     
>> This document describes an extension to the Session Initiation
>> Protocol (SIP) for publishing event state used within the framework
>> for SIP Event Notification. The first application of this extension
>> is targeted at the publication of presence information.
>> The mechanism described in this document can be extended to support
>> publication of any event state, for which there exists an appropriate
>> event package. It is not intended to be a general-purpose mechanism
>> for transport of arbitrary data, as there are better-suited
>> mechanisms for this purpose (FTP, HTTP, etc.)
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-ietf-sip-publish-03.txt
>>
>> To remove yourself from the IETF Announcement list, send a message to
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>>
>> Internet-Drafts are also available by anonymous FTP. Login with the 
>> username
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>> type "cd internet-drafts" and then
>>     "get draft-ietf-sip-publish-03.txt".
>>
>> A list of Internet-Drafts directories can be found in
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>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>
>>
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>>
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> 

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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: "'hisham.khartabil@nokia.com'" <hisham.khartabil@nokia.com>
CC: Jonathan Rosemberg <jdrosen@dynamicsoft.com>, sip <sip@ietf.org>,
        aki.niemi@nokia.com
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Subject: [Sip] Call-Info purpose
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Hisham,

in your draft below, you mention that Jonathan pointed out that:

" It was also discovered that there was no IANA registry set up for the
    "purpose" parameter in the Call-info header of the Session Initiation
    Protocol [1]. This document sets up the IANA registry."

http://www.ietf.org/internet-drafts/draft-khartabil-sip-policy-uri-call-info-purpose-00

When I check the registry, defined in the draft below, I can see that 
"purpose" was already registered (page 6).

http://www.ietf.org/internet-drafts/draft-ietf-sip-parameter-registry-01.txt

Header Field                   Parameter Name     Reference
____________________________________________________________

[...]

Call-Info                      purpose            RFC 3261

Regards,

Gonzalo


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Thread-Topic: [Sip] changes in session timer
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From: "Bharrat, Shaun" <SBharrat@sonusnet.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>, <sip@ietf.org>
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Subject: [Sip] Question re: session timer
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Hi:

No problems with the following changes. However, a different
question. Can someone explain the rational for not allowing
a proxy/UAS to _increase_ the session expiration before=20
accepting it? I understand that the proxy/UAS can effect this
increase by responding with the 422 but then every call
(from that UAC) requires multiple INVITE exchanges. Is it=20
really the case that we expect the UAC to get a 422 and decide=20
it doesn't want the call after all because the session expiration=20
would be too big? In any event, if this were the case, couldn't=20
the UAC just BYE the call at that point?

Thanks in advance for the info.

Cheers,
Shaun


Shaun Bharrat                                  Sonus Networks
www.sonusnet.com          The Voice of the New Public Network  =20


>-----Original Message-----
>From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
>Jonathan Rosenberg
>Sent: Wednesday, February 25, 2004 12:45 AM
>To: sip@ietf.org
>Subject: [Sip] changes in session timer
>
>
>Folks,
>
>I submitted a revision of the session timer, based on comments=20
>received=20
>from IESG and subsequent list discussion. For some reason, this=20
>submission was not processed by the I-D editor (I'm working to correct=20
>that). In the mean time, you can pick it up at:
>
>http://www.jdrosen.net/papers/draft-ietf-sip-session-timer-14.txt
>
>
>There are a few significant changes:
>
>* Formerly, if a session refresh timed out, the UA waited until 1/3 of=20
>the session interval remained before sending a BYE. This contradicts=20
>RFC3261 which basically says to send a BYE immediately. Session timer=20
>now references that behavior.
>
>* There is now an absolute minimum session timer of 90=20
>seconds. This is=20
>basically enough time for completion of a SIP BYE transaction=20
>after half=20
>of the session has expired. The Min-SE header field now has a default=20
>value of 90 seconds when not present. This will preclude usage of=20
>session timer for really fine grained refreshes, say on the order of 5=20
>to 10 seconds. I fear that some people may be trying to do=20
>some kind of=20
>nat traversal or something with session timer - this would=20
>rule that out=20
>formally (it was never the intended usage).
>
>* The default for session-expires, when not present, used to be=20
>"infinity". However, thats not really right. Really, it means=20
>"undefined", and session expiration is then up to local policy of the=20
>proxy. Using infinity would mean that a proxy could never=20
>clean up state.
>
>
>Please comment ASAP if you have any problems with these changes. We'd=20
>like to get session timer moving along.
>
>-Jonathan R.
>
>--=20
>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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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] changes in session timer
Date: Wed, 25 Feb 2004 08:55:20 -0500
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> * There is now an absolute minimum session timer of 90 
> seconds. This is 
> basically enough time for completion of a SIP BYE transaction 
> after half 
> of the session has expired. The Min-SE header field now has a default 
> value of 90 seconds when not present. This will preclude usage of 
> session timer for really fine grained refreshes, say on the 
> order of 5 
> to 10 seconds. I fear that some people may be trying to do 
> some kind of 
> nat traversal or something with session timer - this would 
> rule that out 
> formally (it was never the intended usage).
I know whenever we bring up billing people turn off, but one of the things
session timer is useful for is to tear down calls that got hosed somewhere.
Making the min 90 seconds means you could not use the mechanism -- you would
have to have some other mechanism that didn't distort your billing record
so much.  90 seconds is an eternity in human time. 

Somehow, precluding use for NAT binding keepalive doesn't sound like
a good enough reason to have such a high min.  I frankly don't understand
why there has to be a min at all; surely it depends on the characteristics
of the network the session traverses.  

Brian 

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To: ext Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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        Jonathan Rosemberg <jdrosen@dynamicsoft.com>, sip <sip@ietf.org>
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Hi Gonzalo,

You're right - the parameter registration is handled by the below draft. 
But what is missing is a registry for the 'purpose' values. That's what 
we're porposing to set up, and populate with the already defined set of 
{icon, info, card} per RFC3261, and the new proposed 'policy-uri' value.

Cheers,
Aki

ext Gonzalo Camarillo wrote:
> Hisham,
> 
> in your draft below, you mention that Jonathan pointed out that:
> 
> " It was also discovered that there was no IANA registry set up for the
>    "purpose" parameter in the Call-info header of the Session Initiation
>    Protocol [1]. This document sets up the IANA registry."
> 
> http://www.ietf.org/internet-drafts/draft-khartabil-sip-policy-uri-call-info-purpose-00 
> 
> 
> When I check the registry, defined in the draft below, I can see that 
> "purpose" was already registered (page 6).
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sip-parameter-registry-01.txt 
> 
> 
> Header Field                   Parameter Name     Reference
> ____________________________________________________________
> 
> [...]
> 
> Call-Info                      purpose            RFC 3261
> 
> Regards,
> 
> Gonzalo
> 
> 
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Jonathan Rosenberg wrote:
...
> * The default for session-expires, when not present, used to be 
> "infinity". However, thats not really right. Really, it means 
> "undefined", and session expiration is then up to local policy of the 
> proxy. Using infinity would mean that a proxy could never clean up state.

I see a problem with this change. What you say makes sense for proxies, 
but it doesn't make sense for a UAC or UAS. For them, infinity is the 
right default, because in this case they don't refresh. In the absence 
of a session-expires the UAC and UAS can't know the proxy's policy. 
Perhaps the wording needs separated out for proxy and UA.

The new wording is:

    The default value of the Session-Expires header field is undefined.
    This means that absence of the Session-Expires header field implies
    no expiration of the session using the mechanism defined in this
    specification. Note that other mechanisms not defined in this
    specification, such as locally configured timers, may apply.

while the old wording was:

    The default value of the Session-Expires header field, when not
    present, is infinity. This means that absence of the Session-Expires
    header field implies no expiration.

How about the following as a compromise:

    The default value of the Session-Expires header field, when not
    present, is infinity. For UAC and UAS this means that absence of
    the Session-Expires header field implies the session timer will
    never expire, so no BYE will be sent on expiration, and no refreshes
    will be performed. This also implies that proxies must make their
    own determination of how long to retain state in the absence of
    any messages within the dialog.

	Paul


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        Jonathan Rosemberg <jdrosen@dynamicsoft.com>, sip <sip@ietf.org>
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Hi,

yes, when we were creating the registries for header field and URI 
parameters I thought of creating another registry for the values of some 
parameters (e.g., the one you are dealing with)... but given the huge 
amount of discussions we had on the other registries, I gave up for the 
time being...

In any case, I will check what is going on with the IANA and get back to 
you...

Gonzalo

Aki Niemi wrote:
> Hi Gonzalo,
> 
> You're right - the parameter registration is handled by the below draft. 
> But what is missing is a registry for the 'purpose' values. That's what 
> we're porposing to set up, and populate with the already defined set of 
> {icon, info, card} per RFC3261, and the new proposed 'policy-uri' value.
> 
> Cheers,
> Aki
> 
> ext Gonzalo Camarillo wrote:
> 
>> Hisham,
>>
>> in your draft below, you mention that Jonathan pointed out that:
>>
>> " It was also discovered that there was no IANA registry set up for the
>>    "purpose" parameter in the Call-info header of the Session Initiation
>>    Protocol [1]. This document sets up the IANA registry."
>>
>> http://www.ietf.org/internet-drafts/draft-khartabil-sip-policy-uri-call-info-purpose-00 
>>
>>
>> When I check the registry, defined in the draft below, I can see that 
>> "purpose" was already registered (page 6).
>>
>> http://www.ietf.org/internet-drafts/draft-ietf-sip-parameter-registry-01.txt 
>>
>>
>> Header Field                   Parameter Name     Reference
>> ____________________________________________________________
>>
>> [...]
>>
>> Call-Info                      purpose            RFC 3261
>>
>> Regards,
>>
>> Gonzalo
>>
>>
>> This communication is confidential and intended solely for the 
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-- 
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Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
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Gonzalo Camarillo wrote:

> Hi,
> 
> yes, when we were creating the registries for header field and URI 
> parameters I thought of creating another registry for the values of some 
> parameters (e.g., the one you are dealing with)... but given the huge 
> amount of discussions we had on the other registries, I gave up for the 
> time being...
> 
> In any case, I will check what is going on with the IANA and get back to 
> you...

Check for what? The absence of this registry is a bug in RFC3261; it 
should have been created there (indeed, its mentioned in the spec, but 
there are no IANA considerations associated with it). So, we need to set 
it up.

-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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Jonathan,

 > Check for what?

as you know, we both agree on this, and that's why we set both 
registries up... nevertheless, I have heard from the IANA (and from the 
IESG about the IANA) that they are overloaded and that we should not 
hold our breath waiting for the RFCs-to-be defining the registries to be 
published as RFCs...

Since we will have *many* RFCs-to-be that will depend on these 
registries and on any new registry we may need to create, I want to make 
sure that the IANA understands that we are in a hurry. That's whay I am 
going to check...

Gonzalo

Jonathan Rosenberg wrote:

> 
> 
> Gonzalo Camarillo wrote:
> 
>> Hi,
>>
>> yes, when we were creating the registries for header field and URI 
>> parameters I thought of creating another registry for the values of 
>> some parameters (e.g., the one you are dealing with)... but given the 
>> huge amount of discussions we had on the other registries, I gave up 
>> for the time being...
>>
>> In any case, I will check what is going on with the IANA and get back 
>> to you...
> 
> 
> Check for what? The absence of this registry is a bug in RFC3261; it 
> should have been created there (indeed, its mentioned in the spec, but 
> there are no IANA considerations associated with it). So, we need to set 
> it up.
> 
> -Jonathan R.
> 

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


This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.

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Cc: "'sip@ietf.org'" <sip@ietf.org>, Rohan Mahy <rohan@cisco.com>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-publish-03.txt
Date: Wed, 25 Feb 2004 09:18:32 -0800
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Hi Aki,

I'm fine with your proposal.

thx,
-r


On Feb 25, 2004, at 1:58 AM, Aki Niemi wrote:

> Hi Rohan,
>
>
>
> ext Rohan Mahy wrote:
>> Hi Folks,
>> I just noticed a minor problem with publish ; it defines a new 
>> response code:
>>     412 Precondition Failed
>> which is confusingly similar to the existing response code:
>>     580 Precondition Failure
>> I propose changing The reason phrase for 412 to "Publication 
>> Condition Failed"
>
> Good point. A question though: since we're borrowing this response 
> code from HTTP, how strictly do we need to keep the reason phrases 
> aligned? I assume it's enough that the semantics are the same, and the 
> default reason phrase can therefore be different.
>
> Also, to generalize, would:
>
> 	412 (Conditional Request Failed)
>
> be better?
>
> Cheers,
> Aki
>
>> thanks,
>> -rohan
>> On Feb 16, 2004, at 12:41 PM, Internet-Drafts@ietf.org wrote:
>>> A New Internet-Draft is available from the on-line Internet-Drafts 
>>> directories.
>>> This draft is a work item of the Session Initiation Protocol Working 
>>> Group of the IETF.
>>>
>>>     Title        : Session Initiation Protocol (SIP) Extension for 
>>> Event State
>>>                               Publication
>>>     Author(s)    : A. Niemi
>>>     Filename    : draft-ietf-sip-publish-03.txt
>>>     Pages        : 40
>>>     Date        : 2004-2-16
>>>     This document describes an extension to the Session Initiation
>>> Protocol (SIP) for publishing event state used within the framework
>>> for SIP Event Notification. The first application of this extension
>>> is targeted at the publication of presence information.
>>> The mechanism described in this document can be extended to support
>>> publication of any event state, for which there exists an appropriate
>>> event package. It is not intended to be a general-purpose mechanism
>>> for transport of arbitrary data, as there are better-suited
>>> mechanisms for this purpose (FTP, HTTP, etc.)
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-ietf-sip-publish-03.txt
>>>
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>>>
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>>> username
>>> "anonymous" and a password of your e-mail address. After logging in,
>>> type "cd internet-drafts" and then
>>>     "get draft-ietf-sip-publish-03.txt".
>>>
>>> A list of Internet-Drafts directories can be found in
>>> http://www.ietf.org/shadow.html
>>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>>
>>>
>>> Internet-Drafts can also be obtained by e-mail.
>>>
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>
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Subject: [Sip] INFO Documentation (was Re: Defining new requests)
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Q.1912.5 is in the process of final approval.  Logging errata may be satisfactory 
-- I'd have to check.  It would certainly be more comfortable than reproducing the 
RFC with modifications in the ITU document.  If the IETF acts within the next two 
weeks we should be able to remove the Annex from the Recommendation.

Here is the current list of changes as reported at the beginning of the Annex.

"The SIP INFO method is documented in RFC 2976. However, that RFC refers to RFC 
2543, the previous version of the SIP specification, rather than to RFC 3261. The 
text of this Annex reproduces RFC 2976 with changes to reflect the updated reference.

"The changes made are as follows:

"Paragraph C.3.2.1: The references to the tables of RFC2543 were changed to 
reflect the new table organization in RFC 3261.

"Paragraph C.3.2: The tables 1 and 2 were replaced by new tables that show the 
header fields defined as presented in RFC3261. All additional headers (additional 
to RFC 2543) used within RFC3261 were here set to optional with two exceptions. 
The Reply-To (only optional for the INVITE) and Call-Info Header (only used for 
INVITE, OPTIONS, REGISTER to give additional information about the caller or 
callee) are not used with the INFO method.

"Paragraph C.3.6: The reference was changed to the current SIP specification, RFC 
3261.


Gonzalo Camarillo wrote:

> Tom,
> 
> what is the timeframe here? As you know, we have had discussions in the 
> past about either re-defining INFO so that it can only carry PSTN 
> telephony signalling, or defining INFO packages. We could re-start those 
> discussions, and once we reach a conclusion, clarify the rules to send 
> INFO from the UAS within early dialogs in the same spec...
> 
> If the ITU is in a hurry, other solutions may be better, like logging an 
> errata against the INFO RFC.
> 
> Gonzalo
> 
> Tom Taylor wrote:
> 
>> The ITU-T (Study Group 11, to be exact) had a different issue with 
>> INFO: that the RFC depends on RFC 2543 rather than 3261.  As a result, 
>> they picked up the content of RFC 2976 as Annex C.3 of the ISUP 
>> interworking specification Q.1912.5.  If RFC 2976 ever gets updated 
>> Annex C.3 can be deprecated.  In the meantime, this INFO question 
>> means we have some maintenance to perform.  I think the best solution 
>> will be to take Gonzalo's suggestion and say that INFO follows the 
>> rules for UPDATE.  We already have a normative dependency on the 
>> UPDATE RFC.
>>
>> Gonzalo Camarillo wrote:
>>
>>> Christer,
>>>
>>> would the ITU-T be happier if we log an errata against the INFO RFC 
>>> saying that UASs can send INFO within early dialogs?
>>>
>>> Gonzalo
>>>
>>> Christer Holmberg (JO/LMF) wrote:
>>>
>>>> Hi Gonzalo,
>>>>
>>>> Thanks for your reply!
>>>>
>>>>
>>>>> it seems that some people in the ITU-T are having some issues 
>>>>> figuring out whether or not certain requests can be sent by the UAS
>>>>
>>>>
>>>>
>>>>
>>>> within an
>>>>
>>>>> early dialog. The confusion comes from statements like this one 
>>>>> (taken from the INFO spec):
>>>>>
>>>>> "     Unless stated otherwise, the protocol rules for the INFO request
>>>>>       governing the usage of tags, Route and Record-Route,
>>>>>       retransmission and reliability, CSeq incrementing and message
>>>>>       formatting follow those in [1] as defined for the BYE request."
>>>>>
>>>>> Since a UAS does not send BYEs within an early dialog, but a 
>>>>> non-2xx final response instead, to terminate the dialog, people 
>>>>> assume that
>>>>
>>>>
>>>>
>>>>
>>>> INFO
>>>>
>>>>> cannot be sent in this situation either.
>>>>>
>>>>> Let's try to keep this in mind and clarify this point when/if we 
>>>>> define new SIP methods.
>>>>>
>>>>> In any event, the clarification to this issue can be found in RFC 
>>>>> 3398:
>>>>>
>>>>> "From the callee to the caller, if a message is received by a gateway
>>>>> before the call has been answered (i.e., ANM is received) it SHOULD 
>>>>> be encapsulated in an INFO, provided that this will not be the 
>>>>> first SIP message sent in the backwards direction (in which case it 
>>>>> SHOULD be encapsulated in a provisional 1xx response)."
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> The problem is that the ITU-T interworking spec doesn't reference
>>>> RFC3398. The spec reference the INFO RFC, and that is also where I 
>>>> think
>>>> the usage rules should be defined (not in another spec, RFC3398 in this
>>>> case, simply using the method).
>>>>
>>>> The INFO RFC, as you said, currently says that INFO "follows the rules
>>>> of BYE" (whatever that means is another question :), and the rules for
>>>> BYE specifically says that it must not be sent backward before the
>>>> INVITE has finished. If the INFO RFC would say "follow the rules of
>>>> UPDATE" we would not have this problem, since the UPDATE spec
>>>> specifically allows the use of the method before the INVITE has
>>>> finished.
>>>>
>>>> Thanks!
>>>>
>>>> Christer Holmberg
>>>> Ericsson Finland
>>>
>>>
> 
> 
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Date: Wed, 25 Feb 2004 15:20:00 -0500
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        "'sip-implementors@cs.columbia.edu'" <sip-implementors@cs.columbia.edu>
Subject: Re: [Sip] From Header
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Responses are sent to the Via address, though more commonly to the 
source IP/port of the request when TCP is used, or when rport is used 
(RFC 3581) with UDP. The IP/port in the Via effectively provide a 
fallback, to be used for the server to re-initiate a connection to the 
client in the event of failure to send the response.

Subsequent requests in a dialog are sent to the Contact URI.

 From is a logical identifier, used for display and policy purposes. 
Please, please don't muck with it.

Let me take this opportunity to preach a little on the role of ALGs in a 
network.

As a general rule, ALGs are quite problematic. They don't work in 
conjunction with most of the SIP security mechanisms, including S/MIME 
but more importantly sips and TLS. Their spotty presence in a network, 
generally undetectable to application entities in the network, cannot be 
depended on, and therefore can never serve as a complete solution for an 
application service provider. Worse yet, when (and I say when, not if) 
one of these ALGs does something wrong that breaks a call flow, it 
becomes almost impossible to diagnose and correct. Generally speaking, 
application layer intelligence does not belong in the core of the 
network; that brings us back to the days of single-service networks 
where the whole network needs to be upgraded to support a new 
application. ALGs are contrary to the hourglass model that is one of the 
defining characteristics of IP.

I much prefer solutions like ICE, STUN, TURN and MIDCOM, which separate 
application layer intelligence from firewalls and NATs, and allow for 
applications to do the right thing for themselves for the purposes of 
NAT traversal.

As such, the right model is to NOT build an ALG, but if you must, build 
a full protocol-compliant entity that just so happens to ship with and 
be co-resident in your firewall/nat. That entity must be directly 
addressed by other SIP entities, rather than being a transparent element 
in the network. It is only through this model that we can create robust 
and secure SIP networks.

-Jonathan R.


Anil Bollineni wrote:

> The Via header tells that IP address/port where the response should be sent.
> It does not specify from where the request has been come, as the ports for
> sending request and receiving responses can be different. I think the only
> way we can tell is by seeing the sending ip and port in IP header of the
> packet. Please correct if I am wrong.
> 
> Thanks,
> Anil
> 
> 
> This email contains material that is confidential. The content of this email
> is for the sole use of the intended recipient(s). Any review or distribution
> by persons other than the intended recipient(s) without the express
> permission of NetScreen Technologies, Inc. is strictly prohibited. If you
> are not the intended recipient, please contact the sender and delete/destroy
> all copies of this email and any related attachments. NetScreen does not
> guarantee the accuracy or completeness of third party materials or
> information.
> 
> 
> -----Original Message-----
> From: xiao'tong liang [mailto:boyliangxt@yahoo.com] 
> Sent: Sunday, February 22, 2004 6:07 PM
> To: Anil Bollineni
> Subject: Re: [Sip] From Header
> 
> Via header.
> 
> --
> xiaotong
> 
> 
> --- Anil Bollineni <ABollineni@netscreen.com> wrote:
> 
>>Hi All,
>>
>> 
>>
>>      I understand from the following statement in
>>RFC, that From header
>>SHOULD not contain IP addresses/ports from whom the
>>request has come. If it
>>is so, I can't assume that the information is not
>>used to assume from where
>>(IP address/port) the request has come. Appreciate
>>your response to clarify
>>the point.
>>
>> 
>>
>>  As such, it is very important that the From URI
>>not contain IP addresses
>>or the FQDN
>>
>>   of the host on which the UA is running, since
>>these are not logical
>>
>>   names.
>>
>> 
>>
>>Thank you,
>>
>>Anil
>>
>> 
>>
>>This email contains material that is confidential.
>>The content of this email
>>is for the sole use of the intended recipient(s).
>>Any review or distribution
>>by persons other than the intended recipient(s)
>>without the express
>>permission of NetScreen Technologies, Inc. is
>>strictly prohibited. If you
>>are not the intended recipient, please contact the
>>sender and delete/destroy
>>all copies of this email and any related
>>attachments. NetScreen does not
>>guarantee the accuracy or completeness of third
>>party materials or
>>information.
>>
>> 
>>
>>
> 
> 
> 
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-- 
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Subject: [Sip] Attention: SIP draft authors
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Hi Folks,

If you are writing a draft using the xml2rfc tool, please make sure 
that the following line appears just after the line which begins <?rfc 
toc=

	<?rfc compact="yes"?>

if you don't do this, the tool will generate each top-level section on 
its own page, so you end up wasting a whole page to print for example:

	11. IANA Considerations

	   This document requires no action by IANA.

many thanks for saving trees, etc.. and making my reading list lighter 
(at least by weight).

thanks,
-rohan



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Cc: Rohan Mahy <rohan@cisco.com>, Dean Willis <dean.willis@softarmor.com>,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>
From: Rohan Mahy <rohan@cisco.com>
Date: Wed, 25 Feb 2004 17:28:14 -0800
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Subject: [Sip] SIP WG Reading List PDF now available
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Hi Folks,

A single, properly-paginated, suitable for double-sided printing, PDF 
file of the entire reading list for the SIP WGs upcoming meeting is now 
available on the supplemental website. Links are available from the 
agenda page:

http://www.softarmor.com/sipwg/meets/ietf59/agenda.html

There are actually 4 PDFs available.  You can select from PDFs for A4 
or US-Letter paper, and one-page per side or two-pages per side 
layouts.

Enjoy and see many of you in Korea.
thanks,
-rohan


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Subject: [Sip] Re: INFO Documentation (was Re: Defining new requests)
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Tom.

it seems that what you want is an update of the INFO spec, rather than 
fixing a few errata. I believe that we cannot meet your timeframe. In 
any event, it would be great we could start working on a solution for 
INFO in SIPPING, so that the ITU could reference it in the future.

Thanks,

Gonzalo

Tom Taylor wrote:

> Q.1912.5 is in the process of final approval.  Logging errata may be 
> satisfactory -- I'd have to check.  It would certainly be more 
> comfortable than reproducing the RFC with modifications in the ITU 
> document.  If the IETF acts within the next two weeks we should be able 
> to remove the Annex from the Recommendation.
> 
> Here is the current list of changes as reported at the beginning of the 
> Annex.
> 
> "The SIP INFO method is documented in RFC 2976. However, that RFC refers 
> to RFC 2543, the previous version of the SIP specification, rather than 
> to RFC 3261. The text of this Annex reproduces RFC 2976 with changes to 
> reflect the updated reference.
> 
> "The changes made are as follows:
> 
> "Paragraph C.3.2.1: The references to the tables of RFC2543 were changed 
> to reflect the new table organization in RFC 3261.
> 
> "Paragraph C.3.2: The tables 1 and 2 were replaced by new tables that 
> show the header fields defined as presented in RFC3261. All additional 
> headers (additional to RFC 2543) used within RFC3261 were here set to 
> optional with two exceptions. The Reply-To (only optional for the 
> INVITE) and Call-Info Header (only used for INVITE, OPTIONS, REGISTER to 
> give additional information about the caller or callee) are not used 
> with the INFO method.
> 
> "Paragraph C.3.6: The reference was changed to the current SIP 
> specification, RFC 3261.
> 
> 
> Gonzalo Camarillo wrote:
> 
>> Tom,
>>
>> what is the timeframe here? As you know, we have had discussions in 
>> the past about either re-defining INFO so that it can only carry PSTN 
>> telephony signalling, or defining INFO packages. We could re-start 
>> those discussions, and once we reach a conclusion, clarify the rules 
>> to send INFO from the UAS within early dialogs in the same spec...
>>
>> If the ITU is in a hurry, other solutions may be better, like logging 
>> an errata against the INFO RFC.
>>
>> Gonzalo
>>
>> Tom Taylor wrote:
>>
>>> The ITU-T (Study Group 11, to be exact) had a different issue with 
>>> INFO: that the RFC depends on RFC 2543 rather than 3261.  As a 
>>> result, they picked up the content of RFC 2976 as Annex C.3 of the 
>>> ISUP interworking specification Q.1912.5.  If RFC 2976 ever gets 
>>> updated Annex C.3 can be deprecated.  In the meantime, this INFO 
>>> question means we have some maintenance to perform.  I think the best 
>>> solution will be to take Gonzalo's suggestion and say that INFO 
>>> follows the rules for UPDATE.  We already have a normative dependency 
>>> on the UPDATE RFC.
>>>
>>> Gonzalo Camarillo wrote:
>>>
>>>> Christer,
>>>>
>>>> would the ITU-T be happier if we log an errata against the INFO RFC 
>>>> saying that UASs can send INFO within early dialogs?
>>>>
>>>> Gonzalo
>>>>
>>>> Christer Holmberg (JO/LMF) wrote:
>>>>
>>>>> Hi Gonzalo,
>>>>>
>>>>> Thanks for your reply!
>>>>>
>>>>>
>>>>>> it seems that some people in the ITU-T are having some issues 
>>>>>> figuring out whether or not certain requests can be sent by the UAS
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> within an
>>>>>
>>>>>> early dialog. The confusion comes from statements like this one 
>>>>>> (taken from the INFO spec):
>>>>>>
>>>>>> "     Unless stated otherwise, the protocol rules for the INFO 
>>>>>> request
>>>>>>       governing the usage of tags, Route and Record-Route,
>>>>>>       retransmission and reliability, CSeq incrementing and message
>>>>>>       formatting follow those in [1] as defined for the BYE request."
>>>>>>
>>>>>> Since a UAS does not send BYEs within an early dialog, but a 
>>>>>> non-2xx final response instead, to terminate the dialog, people 
>>>>>> assume that
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> INFO
>>>>>
>>>>>> cannot be sent in this situation either.
>>>>>>
>>>>>> Let's try to keep this in mind and clarify this point when/if we 
>>>>>> define new SIP methods.
>>>>>>
>>>>>> In any event, the clarification to this issue can be found in RFC 
>>>>>> 3398:
>>>>>>
>>>>>> "From the callee to the caller, if a message is received by a gateway
>>>>>> before the call has been answered (i.e., ANM is received) it 
>>>>>> SHOULD be encapsulated in an INFO, provided that this will not be 
>>>>>> the first SIP message sent in the backwards direction (in which 
>>>>>> case it SHOULD be encapsulated in a provisional 1xx response)."
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> The problem is that the ITU-T interworking spec doesn't reference
>>>>> RFC3398. The spec reference the INFO RFC, and that is also where I 
>>>>> think
>>>>> the usage rules should be defined (not in another spec, RFC3398 in 
>>>>> this
>>>>> case, simply using the method).
>>>>>
>>>>> The INFO RFC, as you said, currently says that INFO "follows the rules
>>>>> of BYE" (whatever that means is another question :), and the rules for
>>>>> BYE specifically says that it must not be sent backward before the
>>>>> INVITE has finished. If the INFO RFC would say "follow the rules of
>>>>> UPDATE" we would not have this problem, since the UPDATE spec
>>>>> specifically allows the use of the method before the INVITE has
>>>>> finished.
>>>>>
>>>>> Thanks!
>>>>>
>>>>> Christer Holmberg
>>>>> Ericsson Finland
>>>>
>>>>
>>>>
>>
>>
>> This communication is confidential and intended solely for the 
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>> delete the message without disclosing it. Thank you.
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-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


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Subject: [Sip] Re: INFO Documentation (was Re: Defining new requests)
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As a first step, I'll submit the Q.1912.5 annex as an I-D.  I expect we'll correct 
it to use UPDATE rather than BYE as the model in Geneva in the next couple of 
weeks, so what I submit will reflect that.

Gonzalo Camarillo wrote:

> Tom.
> 
> it seems that what you want is an update of the INFO spec, rather than 
> fixing a few errata. I believe that we cannot meet your timeframe. In 
> any event, it would be great we could start working on a solution for 
> INFO in SIPPING, so that the ITU could reference it in the future.
> 
> Thanks,
> 
> Gonzalo
> 
> Tom Taylor wrote:
> 
>> Q.1912.5 is in the process of final approval.  Logging errata may be 
>> satisfactory -- I'd have to check.  It would certainly be more 
>> comfortable than reproducing the RFC with modifications in the ITU 
>> document.  If the IETF acts within the next two weeks we should be 
>> able to remove the Annex from the Recommendation.
>>
>> Here is the current list of changes as reported at the beginning of 
>> the Annex.
>>
>> "The SIP INFO method is documented in RFC 2976. However, that RFC 
>> refers to RFC 2543, the previous version of the SIP specification, 
>> rather than to RFC 3261. The text of this Annex reproduces RFC 2976 
>> with changes to reflect the updated reference.
>>
>> "The changes made are as follows:
>>
>> "Paragraph C.3.2.1: The references to the tables of RFC2543 were 
>> changed to reflect the new table organization in RFC 3261.
>>
>> "Paragraph C.3.2: The tables 1 and 2 were replaced by new tables that 
>> show the header fields defined as presented in RFC3261. All additional 
>> headers (additional to RFC 2543) used within RFC3261 were here set to 
>> optional with two exceptions. The Reply-To (only optional for the 
>> INVITE) and Call-Info Header (only used for INVITE, OPTIONS, REGISTER 
>> to give additional information about the caller or callee) are not 
>> used with the INFO method.
>>
>> "Paragraph C.3.6: The reference was changed to the current SIP 
>> specification, RFC 3261.
>>
>>
>> Gonzalo Camarillo wrote:
>>
>>> Tom,
>>>
>>> what is the timeframe here? As you know, we have had discussions in 
>>> the past about either re-defining INFO so that it can only carry PSTN 
>>> telephony signalling, or defining INFO packages. We could re-start 
>>> those discussions, and once we reach a conclusion, clarify the rules 
>>> to send INFO from the UAS within early dialogs in the same spec...
>>>
>>> If the ITU is in a hurry, other solutions may be better, like logging 
>>> an errata against the INFO RFC.
>>>
>>> Gonzalo
>>>
>>> Tom Taylor wrote:
>>>
>>>> The ITU-T (Study Group 11, to be exact) had a different issue with 
>>>> INFO: that the RFC depends on RFC 2543 rather than 3261.  As a 
>>>> result, they picked up the content of RFC 2976 as Annex C.3 of the 
>>>> ISUP interworking specification Q.1912.5.  If RFC 2976 ever gets 
>>>> updated Annex C.3 can be deprecated.  In the meantime, this INFO 
>>>> question means we have some maintenance to perform.  I think the 
>>>> best solution will be to take Gonzalo's suggestion and say that INFO 
>>>> follows the rules for UPDATE.  We already have a normative 
>>>> dependency on the UPDATE RFC.
>>>>
>>>> Gonzalo Camarillo wrote:
>>>>
>>>>> Christer,
>>>>>
>>>>> would the ITU-T be happier if we log an errata against the INFO RFC 
>>>>> saying that UASs can send INFO within early dialogs?
>>>>>
>>>>> Gonzalo
>>>>>
>>>>> Christer Holmberg (JO/LMF) wrote:
>>>>>
>>>>>> Hi Gonzalo,
>>>>>>
>>>>>> Thanks for your reply!
>>>>>>
>>>>>>
>>>>>>> it seems that some people in the ITU-T are having some issues 
>>>>>>> figuring out whether or not certain requests can be sent by the UAS
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> within an
>>>>>>
>>>>>>> early dialog. The confusion comes from statements like this one 
>>>>>>> (taken from the INFO spec):
>>>>>>>
>>>>>>> "     Unless stated otherwise, the protocol rules for the INFO 
>>>>>>> request
>>>>>>>       governing the usage of tags, Route and Record-Route,
>>>>>>>       retransmission and reliability, CSeq incrementing and message
>>>>>>>       formatting follow those in [1] as defined for the BYE 
>>>>>>> request."
>>>>>>>
>>>>>>> Since a UAS does not send BYEs within an early dialog, but a 
>>>>>>> non-2xx final response instead, to terminate the dialog, people 
>>>>>>> assume that
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> INFO
>>>>>>
>>>>>>> cannot be sent in this situation either.
>>>>>>>
>>>>>>> Let's try to keep this in mind and clarify this point when/if we 
>>>>>>> define new SIP methods.
>>>>>>>
>>>>>>> In any event, the clarification to this issue can be found in RFC 
>>>>>>> 3398:
>>>>>>>
>>>>>>> "From the callee to the caller, if a message is received by a 
>>>>>>> gateway
>>>>>>> before the call has been answered (i.e., ANM is received) it 
>>>>>>> SHOULD be encapsulated in an INFO, provided that this will not be 
>>>>>>> the first SIP message sent in the backwards direction (in which 
>>>>>>> case it SHOULD be encapsulated in a provisional 1xx response)."
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> The problem is that the ITU-T interworking spec doesn't reference
>>>>>> RFC3398. The spec reference the INFO RFC, and that is also where I 
>>>>>> think
>>>>>> the usage rules should be defined (not in another spec, RFC3398 in 
>>>>>> this
>>>>>> case, simply using the method).
>>>>>>
>>>>>> The INFO RFC, as you said, currently says that INFO "follows the 
>>>>>> rules
>>>>>> of BYE" (whatever that means is another question :), and the rules 
>>>>>> for
>>>>>> BYE specifically says that it must not be sent backward before the
>>>>>> INVITE has finished. If the INFO RFC would say "follow the rules of
>>>>>> UPDATE" we would not have this problem, since the UPDATE spec
>>>>>> specifically allows the use of the method before the INVITE has
>>>>>> finished.
>>>>>>
>>>>>> Thanks!
>>>>>>
>>>>>> Christer Holmberg
>>>>>> Ericsson Finland
>>>>>
>>>>>
>>>>>
>>>>>
>>>
>>>
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> 

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From: "Ashir Ahmed" <a.ahmed@ntt.com>
Subject: Re: [Sip] From Header
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Anil Bollineni" <ABollineni@netscreen.com>
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-Both To and From headers contain logical addresses. These headers are not
used for routing purposes.
-Contact header in general contains a SIP(S) URI for receiving subsequent
requests. However, this header is NOT used in PUBLISH and MESSAGE requests
as they do not expect any subsequent requests.

Please find a general guideline on these message headers and URI scheme
usage at
http://www.softfront.co.jp/tech/ietfdoc/draft/draft-ashir-simple-message-guideline-01.txt


Ashir

----- Original Message ----- 
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Anil Bollineni" <ABollineni@netscreen.com>
Cc: "'xiao'tong liang'" <boyliangxt@yahoo.com>; <sip@ietf.org>;
<sip-implementors@cs.columbia.edu>
Sent: Thursday, February 26, 2004 5:20 AM
Subject: Re: [Sip] From Header


> Responses are sent to the Via address, though more commonly to the
> source IP/port of the request when TCP is used, or when rport is used
> (RFC 3581) with UDP. The IP/port in the Via effectively provide a
> fallback, to be used for the server to re-initiate a connection to the
> client in the event of failure to send the response.
>
> Subsequent requests in a dialog are sent to the Contact URI.
>
>  From is a logical identifier, used for display and policy purposes.
> Please, please don't muck with it.
>
> Let me take this opportunity to preach a little on the role of ALGs in a
> network.
>
> As a general rule, ALGs are quite problematic. They don't work in
> conjunction with most of the SIP security mechanisms, including S/MIME
> but more importantly sips and TLS. Their spotty presence in a network,
> generally undetectable to application entities in the network, cannot be
> depended on, and therefore can never serve as a complete solution for an
> application service provider. Worse yet, when (and I say when, not if)
> one of these ALGs does something wrong that breaks a call flow, it
> becomes almost impossible to diagnose and correct. Generally speaking,
> application layer intelligence does not belong in the core of the
> network; that brings us back to the days of single-service networks
> where the whole network needs to be upgraded to support a new
> application. ALGs are contrary to the hourglass model that is one of the
> defining characteristics of IP.
>
> I much prefer solutions like ICE, STUN, TURN and MIDCOM, which separate
> application layer intelligence from firewalls and NATs, and allow for
> applications to do the right thing for themselves for the purposes of
> NAT traversal.
>
> As such, the right model is to NOT build an ALG, but if you must, build
> a full protocol-compliant entity that just so happens to ship with and
> be co-resident in your firewall/nat. That entity must be directly
> addressed by other SIP entities, rather than being a transparent element
> in the network. It is only through this model that we can create robust
> and secure SIP networks.
>
> -Jonathan R.
>
>
> Anil Bollineni wrote:
>
> > The Via header tells that IP address/port where the response should be
sent.
> > It does not specify from where the request has been come, as the ports
for
> > sending request and receiving responses can be different. I think the
only
> > way we can tell is by seeing the sending ip and port in IP header of the
> > packet. Please correct if I am wrong.
> >
> > Thanks,
> > Anil
> >
> >
> > This email contains material that is confidential. The content of this
email
> > is for the sole use of the intended recipient(s). Any review or
distribution
> > by persons other than the intended recipient(s) without the express
> > permission of NetScreen Technologies, Inc. is strictly prohibited. If
you
> > are not the intended recipient, please contact the sender and
delete/destroy
> > all copies of this email and any related attachments. NetScreen does not
> > guarantee the accuracy or completeness of third party materials or
> > information.
> >
> >
> > -----Original Message-----
> > From: xiao'tong liang [mailto:boyliangxt@yahoo.com]
> > Sent: Sunday, February 22, 2004 6:07 PM
> > To: Anil Bollineni
> > Subject: Re: [Sip] From Header
> >
> > Via header.
> >
> > --
> > xiaotong
> >
> >
> > --- Anil Bollineni <ABollineni@netscreen.com> wrote:
> >
> >>Hi All,
> >>
> >>
> >>
> >>      I understand from the following statement in
> >>RFC, that From header
> >>SHOULD not contain IP addresses/ports from whom the
> >>request has come. If it
> >>is so, I can't assume that the information is not
> >>used to assume from where
> >>(IP address/port) the request has come. Appreciate
> >>your response to clarify
> >>the point.
> >>
> >>
> >>
> >>  As such, it is very important that the From URI
> >>not contain IP addresses
> >>or the FQDN
> >>
> >>   of the host on which the UA is running, since
> >>these are not logical
> >>
> >>   names.
> >>
> >>
> >>
> >>Thank you,
> >>
> >>Anil
> >>
> >>
> >>
> >>This email contains material that is confidential.
> >>The content of this email
> >>is for the sole use of the intended recipient(s).
> >>Any review or distribution
> >>by persons other than the intended recipient(s)
> >>without the express
> >>permission of NetScreen Technologies, Inc. is
> >>strictly prohibited. If you
> >>are not the intended recipient, please contact the
> >>sender and delete/destroy
> >>all copies of this email and any related
> >>attachments. NetScreen does not
> >>guarantee the accuracy or completeness of third
> >>party materials or
> >>information.
> >>
> >>
> >>
> >>
> >
> >
> >
> > __________________________________
> > Do you Yahoo!?
> > Yahoo! Mail SpamGuard - Read only the mail you want.
> > http://antispam.yahoo.com/tools
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP 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
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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] rfc3265  Handling of forked requests
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Hi Folks,

rfc3265 says

4.4.9. Handling of forked requests

   Each event package MUST specify whether forked SUBSCRIBE requests are
   allowed to install multiple subscriptions.

   If such behavior is not allowed, the first potential dialog-
   establishing message will create a dialog.  All subsequent NOTIFY
   messages which correspond to the SUBSCRIBE message (i.e., match "To",
   "From", "From" header "tag" parameter, "Call-ID", "CSeq", "Event",
   and "Event" header "id" parameter) but which do not match the dialog
   would be rejected with a 481 response.  Note that the 200-class
   response to the SUBSCRIBE can arrive after a matching NOTIFY has been
   received; such responses might not correlate to the same dialog
   established by the NOTIFY.  Except as required to complete the
   SUBSCRIBE transaction, such non-matching 200-class responses are
   ignored.


The question is, should the "CSeq" field of NOTIFY message participate in corresponding with SUBSCRIBE message?

-- 
Best regards,
 Simon                          mailto:simon.asryan@epygilab.am



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



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From: Simon Asryan <simon.asryan@epygilab.am>
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Subject: [Sip] rfc3265  Handling of forked requests
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Hi Folks,

rfc3265 says

4.4.9. Handling of forked requests

   Each event package MUST specify whether forked SUBSCRIBE requests are
   allowed to install multiple subscriptions.

   If such behavior is not allowed, the first potential dialog-
   establishing message will create a dialog.  All subsequent NOTIFY
   messages which correspond to the SUBSCRIBE message (i.e., match "To",
   "From", "From" header "tag" parameter, "Call-ID", "CSeq", "Event",
   and "Event" header "id" parameter) but which do not match the dialog
   would be rejected with a 481 response.  Note that the 200-class
   response to the SUBSCRIBE can arrive after a matching NOTIFY has been
   received; such responses might not correlate to the same dialog
   established by the NOTIFY.  Except as required to complete the
   SUBSCRIBE transaction, such non-matching 200-class responses are
   ignored.


The question is, should the "CSeq" field of NOTIFY message participate in corresponding with SUBSCRIBE message?

-- 
Best regards,
 Simon                          mailto:simon.asryan@epygilab.am



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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  Sat Feb 28 02:52:48 2004
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Subject: [Sip] Re: Abstraction:  SIP instance identifiers requirements
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Cullen Jennings wrote:

> I don't think this unique instance identifier thing had much to do with
> billing or debugging. I'm imagine a case something like the following:
> 
> I have two offices, one on the west side of the building and one of the
> east. They both have phones registered as fluffy. Because it is better for
> lounging in the Sun, I spend the morning in the East office and afternoon in
> the West. Right now I can't write a proxy that forward to the correct one.

Sure you can! Thats what callerprefs is for. You would define a new 
callee-caps parameter, "exposure" with values of east, west, north and 
south. The two devices would each include a +sip.exposure contact header 
field parmeter indicating which one is on which side. Then, a caller 
wanting to reach the east side would do this:

INVITE sip:fluffy@cisco.com
Accept-Contact: *;+sip.exposure="east"

which would do preferential routing to the east side.

Note, of course, that caller prefs provides no guarantees. Things still 
work if you fork the call to both. It is this unreliability for routing 
tha made it unsuitable for the problem of getting a request to a 
specific instance.

-Jonathan R.

> 
> If I was using SUBSCRIBE to support event notification of provisioning
> changes, I would not be able to identify which phone was subscribing.
> 
> 
> On 2/17/04 11:16 AM, "Dean Willis" <dean.willis@softarmor.com> wrote:
> 
> 
>>1) billing -- "Alice called Bob from phone ID000001 through gateway
>>ID3344555 on Tuesday, March 1, 2011"
>>
>>2) Debugging -- "This request was responded to by node IDYIYIOHIHIUH,
>>which is a Barfia model 9995 phone running Finux 21.05A and the Vocal
>>user agent release 89.6"
> 
> 

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



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CC: Orit Levin <oritl@microsoft.com>, sip@ietf.org
Subject: Re: [Sip] SIP instance identifiers
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inline.

Paul Kyzivat wrote:

>=20
>=20
> Orit Levin wrote:
>=20
>> I tend to agree with Paul (if I read him correctly).
>=20
>=20
> I think maybe you don't read me as intended, because I don't entirely=20
> agree with you here. (At least if I understand you.)
>=20
>> I think that GUID and the original GRID parameter are very different=20
>> things because of their properties (and obviously the resultant usage=20
>> cases).
>=20
>=20
> If you are talking about the grid=3Dxxx parameter that Jonathan has=20
> proposed be used with GRUUs, then I don't think it has any signficance=20
> to this discussion at all.

I agree, I dont see the relationship.

>> 2. _GRUU format:_ GRUU (not GRID or GUID) is a SIP URI and is very=20
>> SIPish concept with the properties as defined in the GRUU draft. It IS=
=20
>> the "SIP Instance Identifier".
>=20
>=20
> The last sentence is debatable. It is *an* endpoint instance identifier=
=20
> that is sip routable. But it isn't necessarily defined for as long as=20
> the device(s) it references are. And it isn't necessarily bound to the=20
> same device at all times.

Right. The GRUU is not a device identifier. Its a URI whose property is=20
that it tends to route to a device with a particular device ID. The UML=20
in the GRUU spec tries to explain the relationship between them.

>> 3. _GRUU signaling and usage:_ GRUU can be allocated either by=20
>> Registrar, or by UA, or by different means. My point is GRUU=20
>> identification (i.e. placement and signaling) during session=20
>> establishment and usage (especially outside the Registrar domain) MUST=
=20
>> be identical for all cases. Moreover, it MUST be possible for the=20
>> Registrar domain to conceal the information about the allocation=20
>> method being used.
>>
>> 4. _Registration Procedure:_ We need to standardize the registration=20
>> procedure for both basic cases: GRUU allocated by Server (done) and=20
>> GRUU allocated by Client. In my opinion, both cases need to be under=20
>> the Registration/contact umbrella - which makes them SIP concepts!=20
>> Whether GRUU value is preserved during client's/server's reboots - it=20
>> doesn't change the concept. From usability point of view, it would be=20
>> very nice to keep the value as long as possible. >From the outside=20
>> user point of view - it doesn't need to know (and doesn't want to=20
>> know) what the registration durability of its peers is.
>=20
>=20
> Jonathan should probably chime in here. I believe he argues that=20
> typically a UA can't tell whether it can generate its own GRUU, so they=
=20
> should always be assigned by registrars.

Correct. If a UA happens to be able to be able to generate a URI that it=20
KNOWS has the properties of a GRUU, this is going to be because it is=20
effectively co-located with a registrar responsible for a particular doma=
in.

>=20
>> 5. _Saving Registration Cycles and DB size_: By using GRUU we have the=
=20
>> ability to contact a specific instance of SIP AOR. Once we point to a=20
>> specific entity, we MAY also need to point to specific sub-entities=20
>> inside this GRUU. This will save the registration burden from each of=20
>> these sub-entities, since they will be covered by GRUU=92s registratio=
n=20
>> and routing. The original GRID parameters allowed for exactly this=20
>> functionality. The usage example is a conference factory which=20
>> generates foci and doesn=92t need to dynamically REGISTER each of them=
=20
>> in order to achieve proper routing. BTW, the concept of =93internal=20
>> routing=94 to specific logical module inside SIP entity can be=20
>> generalized, decoupled from GRUU and be defined as a dedicated SIP=20
>> header.
>=20
>=20
> Maybe it can be decoupled from GRUU, but I think it must be coupled to =
a=20
> URL, not a message. So it must be a URL formatting convention and/or a=20
> url parameter. So maybe you could generalize the grid parameter to appl=
y=20
> to all sip/sips urls, rather than coupling it to gruu. I haven't though=
t=20
> thru where that might be useful.

Sounds like ISDN subaddresssing.

I think that this is a fair analogy to grid. I'm not sure I see cases=20
where it is useful to have the grid functionality separate from GRUUs.=20
Orit, can you give a specific example?

-Jonathan R.
--=20
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: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: <sip@ietf.org>, "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>
Cc: "Toip list" <toip@snowshore.com>, "'Sipping@Ietf. Org'" <sipping@ietf.org>
Subject: [sip]  Comment to draft-agrawal-sip-h323-interworking-reqs-06.txt about text support
Date: Fri, 27 Feb 2004 22:22:48 +0100
Message-ID: <BHEHLFPKIPMLPFNFAHJKMEMLDLAA.gunnar.hellstrom@omnitor.se>
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In section 6.1.5 of the sip-h323-interworking spec, there is a sentence:

"The IWF SHOULD be able to map the common audio, video and application
   format names supported in H.323 to and from the equivalent RTP/AVP
   [RFC3550] names."

It is good practice to always mention how the three main media of video,
audio and text shall be handled in real time applications.

audio and video are mentioned explicitly.

It may not be evident to everybody that real time interactive text is
capability declared in H.323 Annex G / H.245 by
DataApplicationCapability.application =3D t140

So, in fact it is covered in the sentence requiring that "application"
format namnes shall be mapped.

The mapping should correspond to the RTP/AVP  name text/t140 and
corresponding names for inclusion of protection against  loss.

I seek your guidance how it can be made more clear that "text" capabiliti=
es
can be expressed in both H.323 and SIP systems and should be mapped.

( the spec is at
http://www.ietf.org/internet-drafts/draft-agrawal-sip-h323-interworking-r=
eqs
-06.txt )

Regards
Gunnar
-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------




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Date: Sun, 29 Feb 2004 05:38:40 +0900
Subject: Re: [Sip] From Port
From: Cullen Jennings <fluffy@cisco.com>
To: Anil Bollineni <ABollineni@netscreen.com>, <sip@ietf.org>
Message-ID: <BC672B60.335C5%fluffy@cisco.com>
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> This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

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The From does not have to contain anything useful at all. In general it
should have something that is more like the address of record for the user
instead of the the IP address and port that can be used to reach the device=
.
The contact is used to reach the device. So mine mike look like

From: =B3Cullen Jennings=B2 <sip:fluffy@cisco.com>

If the only way to reach the proxy was on port 1234 then I might have to
include that in my AOR but I would usually not put the port that the UA was
using in the from.=20


On 2/20/04 1:00 PM, "Anil Bollineni" <ABollineni@netscreen.com> wrote:

> Hi All,
>             For any SIP message, if UA or Proxy sending message than othe=
r
> than from port 5060, then it should be included in SIP from URI. For exam=
ple,
> if the Register message comes from 202.10.10.20:2218, the From header sho=
uld
> look like From: sip:joe@202.10.10.20:2218. Is it MUST if the port is diff=
erent
> than port 5060 then it should appear in the From header?
> =20
> Thanks in Advance,
> Anil
> This email contains material that is confidential. The content of this em=
ail
> is for the sole use of the intended recipient(s). Any review or distribut=
ion
> by persons other than the intended recipient(s) without the express permi=
ssion
> of NetScreen Technologies, Inc. is strictly prohibited. If you are not th=
e
> intended recipient, please contact the sender and delete/destroy all copi=
es of
> this email and any related attachments. NetScreen does not guarantee the
> accuracy or completeness of third party materials or information.
>=20
> =20
>=20



--B_3160916361_3906624
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<HTML>
<HEAD>
<TITLE>Re: [Sip] From Port</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><BR>
The From does not have to contain anything useful at all. In general it sho=
uld have something that is more like the address of record for the user inst=
ead of the the IP address and port that can be used to reach the device. The=
 contact is used to reach the device. So mine mike look like <BR>
<BR>
From: &#8220;Cullen Jennings&#8221; &lt;sip:fluffy@cisco.com&gt;<BR>
<BR>
If the only way to reach the proxy was on port 1234 then I might have to in=
clude that in my AOR but I would usually not put the port that the UA was us=
ing in the from. <BR>
<BR>
<BR>
On 2/20/04 1:00 PM, &quot;Anil Bollineni&quot; &lt;ABollineni@netscreen.com=
&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Arial"=
>Hi All,<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;For=
 any SIP message, if UA or Proxy sending message than other than from port 5=
060, then it should be included in SIP from URI. For example, if the Registe=
r message comes from 202.10.10.20:2218, the From header should look like Fro=
m: sip:joe@202.10.10.20:2218. Is it MUST if the port is different than port =
5060 then it should appear in the From header?<BR>
&nbsp;<BR>
Thanks in Advance,<BR>
Anil<BR>
This email contains material that is confidential. The content of this emai=
l is for the sole use of the intended recipient(s). Any review or distributi=
on by persons other than the intended recipient(s) without the express permi=
ssion of NetScreen Technologies, Inc. is strictly prohibited. If you are not=
 the intended recipient, please contact the sender and delete/destroy all co=
pies of this email and any related attachments. NetScreen does not guarantee=
 the accuracy or completeness of third party materials or information.<BR>
</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></BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0=
px'><BR>
</SPAN></FONT>
</BODY>
</HTML>


--B_3160916361_3906624--


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



From exim@www1.ietf.org  Sun Feb 29 03:16:44 2004
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Date: Sun, 29 Feb 2004 17:08:40 +0900
From: Cullen Jennings <fluffy@cisco.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: Dean Willis <dean.willis@softarmor.com>,
        Brian Stucker <bstucker@nortelnetworks.com>, <sip@ietf.org>
Message-ID: <BC67CD18.33820%fluffy@cisco.com>
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Subject: [Sip] Re: Abstraction:  SIP instance identifiers requirements
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This just pushes the problem around but does not solve it. When both the
phone boot, how do they know if they registers with the exposure set to east
or west? We need something that makes one phone different from the other
phone and configuration does not really help much because when the phones
first boot they have no configuration.


On 2/28/04 3:48 PM, "Jonathan Rosenberg" <jdrosen@dynamicsoft.com> wrote:

> 
> 
> Cullen Jennings wrote:
> 
>> I don't think this unique instance identifier thing had much to do with
>> billing or debugging. I'm imagine a case something like the following:
>> 
>> I have two offices, one on the west side of the building and one of the
>> east. They both have phones registered as fluffy. Because it is better for
>> lounging in the Sun, I spend the morning in the East office and afternoon in
>> the West. Right now I can't write a proxy that forward to the correct one.
> 
> Sure you can! Thats what callerprefs is for. You would define a new
> callee-caps parameter, "exposure" with values of east, west, north and
> south. The two devices would each include a +sip.exposure contact header
> field parmeter indicating which one is on which side. Then, a caller
> wanting to reach the east side would do this:
> 
> INVITE sip:fluffy@cisco.com
> Accept-Contact: *;+sip.exposure="east"
> 
> which would do preferential routing to the east side.
> 
> Note, of course, that caller prefs provides no guarantees. Things still
> work if you fork the call to both. It is this unreliability for routing
> tha made it unsuitable for the problem of getting a request to a
> specific instance.
> 
> -Jonathan R.
> 
>> 
>> If I was using SUBSCRIBE to support event notification of provisioning
>> changes, I would not be able to identify which phone was subscribing.
>> 
>> 
>> On 2/17/04 11:16 AM, "Dean Willis" <dean.willis@softarmor.com> wrote:
>> 
>> 
>>> 1) billing -- "Alice called Bob from phone ID000001 through gateway
>>> ID3344555 on Tuesday, March 1, 2011"
>>> 
>>> 2) Debugging -- "This request was responded to by node IDYIYIOHIHIUH,
>>> which is a Barfia model 9995 phone running Finux 21.05A and the Vocal
>>> user agent release 89.6"
>> 
>> 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: "Drage, Keith (Keith)" <drage@lucent.com>
CC: "'Tom Taylor'" <taylor@nortelnetworks.com>,
        "'Christer Holmberg (JO/LMF)'" <christer.holmberg@ericsson.com>,
        "'sip'" <sip@ietf.org>, "'Rohan Mahy'" <rohan@cisco.com>,
        "'Dean Willis'" <dean.willis@softarmor.com>
Subject: Re: [Sip] Re: INFO Documentation (was Re: Defining new requests)
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Keith,

> This is an revision requirement to INFO, not to an interworking
> specification, so it should be in SIP, not in SIPPING. It is the core
> INFO RFC that is itself not complete enough in this area.

we are already discussing this in *SIP*, not in *SIPPING*. What I am 
missing from your comment?

Thanks,

Gonzalo


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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: "Drage, Keith (Keith)" <drage@lucent.com>
CC: "'Tom Taylor'" <taylor@nortelnetworks.com>,
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Subject: Re: [Sip] Re: INFO Documentation (was Re: Defining new requests)
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Keith,

OK, I re-read the thread and I have noticed that you were referencing my 
last email. What I actually meant was that we should see whether we 
deprecate INFO so that it only transports PSTN stuff, or, on the other 
hand, we define INFO packages. We had some discussions about that some 
time ago.

So, we need to discuss which requirements we need to fulfill, and that 
should be done in SIPPING. The actual document updating INFO will be 
done in SIP, as you suggest.

Thanks,

Gonzalo

Drage, Keith (Keith) wrote:

> This is an revision requirement to INFO, not to an interworking specification, so it should be in SIP, not in SIPPING. It is the core INFO RFC that is itself not complete enough in this area.
> 
> Note that part of that revision should also be to update the RFC 2543 references to RFC 3261.
> 
> regards
> 
> Keith
> 
> 
>>-----Original Message-----
>>From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
>>Sent: 26 February 2004 12:26
>>To: Tom Taylor
>>Cc: Christer Holmberg (JO/LMF); sip; Rohan Mahy; Dean Willis
>>Subject: [Sip] Re: INFO Documentation (was Re: Defining new requests)
>>
>>
>>Tom.
>>
>>it seems that what you want is an update of the INFO spec, 
>>rather than 
>>fixing a few errata. I believe that we cannot meet your timeframe. In 
>>any event, it would be great we could start working on a solution for 
>>INFO in SIPPING, so that the ITU could reference it in the future.
>>
>>Thanks,
>>
>>Gonzalo
>>
>>Tom Taylor wrote:
>>
>>
>>>Q.1912.5 is in the process of final approval.  Logging 
>>
>>errata may be 
>>
>>>satisfactory -- I'd have to check.  It would certainly be more 
>>>comfortable than reproducing the RFC with modifications in the ITU 
>>>document.  If the IETF acts within the next two weeks we 
>>
>>should be able 
>>
>>>to remove the Annex from the Recommendation.
>>>
>>>Here is the current list of changes as reported at the 
>>
>>beginning of the 
>>
>>>Annex.
>>>
>>>"The SIP INFO method is documented in RFC 2976. However, 
>>
>>that RFC refers 
>>
>>>to RFC 2543, the previous version of the SIP specification, 
>>
>>rather than 
>>
>>>to RFC 3261. The text of this Annex reproduces RFC 2976 
>>
>>with changes to 
>>
>>>reflect the updated reference.
>>>
>>>"The changes made are as follows:
>>>
>>>"Paragraph C.3.2.1: The references to the tables of RFC2543 
>>
>>were changed 
>>
>>>to reflect the new table organization in RFC 3261.
>>>
>>>"Paragraph C.3.2: The tables 1 and 2 were replaced by new 
>>
>>tables that 
>>
>>>show the header fields defined as presented in RFC3261. All 
>>
>>additional 
>>
>>>headers (additional to RFC 2543) used within RFC3261 were 
>>
>>here set to 
>>
>>>optional with two exceptions. The Reply-To (only optional for the 
>>>INVITE) and Call-Info Header (only used for INVITE, 
>>
>>OPTIONS, REGISTER to 
>>
>>>give additional information about the caller or callee) are 
>>
>>not used 
>>
>>>with the INFO method.
>>>
>>>"Paragraph C.3.6: The reference was changed to the current SIP 
>>>specification, RFC 3261.
>>>
>>>
>>>Gonzalo Camarillo wrote:
>>>
>>>
>>>>Tom,
>>>>
>>>>what is the timeframe here? As you know, we have had 
>>
>>discussions in 
>>
>>>>the past about either re-defining INFO so that it can only 
>>
>>carry PSTN 
>>
>>>>telephony signalling, or defining INFO packages. We could re-start 
>>>>those discussions, and once we reach a conclusion, clarify 
>>
>>the rules 
>>
>>>>to send INFO from the UAS within early dialogs in the same spec...
>>>>
>>>>If the ITU is in a hurry, other solutions may be better, 
>>
>>like logging 
>>
>>>>an errata against the INFO RFC.
>>>>
>>>>Gonzalo
>>>>
>>>>Tom Taylor wrote:
>>>>
>>>>
>>>>>The ITU-T (Study Group 11, to be exact) had a different 
>>
>>issue with 
>>
>>>>>INFO: that the RFC depends on RFC 2543 rather than 3261.  As a 
>>>>>result, they picked up the content of RFC 2976 as Annex 
>>
>>C.3 of the 
>>
>>>>>ISUP interworking specification Q.1912.5.  If RFC 2976 ever gets 
>>>>>updated Annex C.3 can be deprecated.  In the meantime, this INFO 
>>>>>question means we have some maintenance to perform.  I 
>>
>>think the best 
>>
>>>>>solution will be to take Gonzalo's suggestion and say that INFO 
>>>>>follows the rules for UPDATE.  We already have a 
>>
>>normative dependency 
>>
>>>>>on the UPDATE RFC.
>>>>>
>>>>>Gonzalo Camarillo wrote:
>>>>>
>>>>>
>>>>>>Christer,
>>>>>>
>>>>>>would the ITU-T be happier if we log an errata against 
>>
>>the INFO RFC 
>>
>>>>>>saying that UASs can send INFO within early dialogs?
>>>>>>
>>>>>>Gonzalo
>>>>>>
>>>>>>Christer Holmberg (JO/LMF) wrote:
>>>>>>
>>>>>>
>>>>>>>Hi Gonzalo,
>>>>>>>
>>>>>>>Thanks for your reply!
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>it seems that some people in the ITU-T are having some issues 
>>>>>>>>figuring out whether or not certain requests can be 
>>
>>sent by the UAS
>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>within an
>>>>>>>
>>>>>>>
>>>>>>>>early dialog. The confusion comes from statements like 
>>
>>this one 
>>
>>>>>>>>(taken from the INFO spec):
>>>>>>>>
>>>>>>>>"     Unless stated otherwise, the protocol rules for the INFO 
>>>>>>>>request
>>>>>>>>      governing the usage of tags, Route and Record-Route,
>>>>>>>>      retransmission and reliability, CSeq 
>>
>>incrementing and message
>>
>>>>>>>>      formatting follow those in [1] as defined for 
>>
>>the BYE request."
>>
>>>>>>>>Since a UAS does not send BYEs within an early dialog, but a 
>>>>>>>>non-2xx final response instead, to terminate the 
>>
>>dialog, people 
>>
>>>>>>>>assume that
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>INFO
>>>>>>>
>>>>>>>
>>>>>>>>cannot be sent in this situation either.
>>>>>>>>
>>>>>>>>Let's try to keep this in mind and clarify this point 
>>
>>when/if we 
>>
>>>>>>>>define new SIP methods.
>>>>>>>>
>>>>>>>>In any event, the clarification to this issue can be 
>>
>>found in RFC 
>>
>>>>>>>>3398:
>>>>>>>>
>>>>>>>>"From the callee to the caller, if a message is 
>>
>>received by a gateway
>>
>>>>>>>>before the call has been answered (i.e., ANM is received) it 
>>>>>>>>SHOULD be encapsulated in an INFO, provided that this 
>>
>>will not be 
>>
>>>>>>>>the first SIP message sent in the backwards direction 
>>
>>(in which 
>>
>>>>>>>>case it SHOULD be encapsulated in a provisional 1xx response)."
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>The problem is that the ITU-T interworking spec doesn't 
>>
>>reference
>>
>>>>>>>RFC3398. The spec reference the INFO RFC, and that is 
>>
>>also where I 
>>
>>>>>>>think
>>>>>>>the usage rules should be defined (not in another spec, 
>>
>>RFC3398 in 
>>
>>>>>>>this
>>>>>>>case, simply using the method).
>>>>>>>
>>>>>>>The INFO RFC, as you said, currently says that INFO 
>>
>>"follows the rules
>>
>>>>>>>of BYE" (whatever that means is another question :), 
>>
>>and the rules for
>>
>>>>>>>BYE specifically says that it must not be sent backward 
>>
>>before the
>>
>>>>>>>INVITE has finished. If the INFO RFC would say "follow 
>>
>>the rules of
>>
>>>>>>>UPDATE" we would not have this problem, since the UPDATE spec
>>>>>>>specifically allows the use of the method before the INVITE has
>>>>>>>finished.
>>>>>>>
>>>>>>>Thanks!
>>>>>>>
>>>>>>>Christer Holmberg
>>>>>>>Ericsson Finland
>>>>>>
>>>>>>
>>>>>>
>>>>
>>>>This communication is confidential and intended solely for the 
>>>>addressee(s). Any unauthorized review, use, disclosure or 
>>
>>distribution 
>>
>>>>is prohibited. If you believe this message has been sent to you in 
>>>>error, please notify the sender by replying to this 
>>
>>transmission and 
>>
>>>>delete the message without disclosing it. Thank you.
>>>>
>>>>E-mail including attachments is susceptible to data corruption, 
>>>>interruption, unauthorized amendment, tampering and 
>>
>>viruses, and we 
>>
>>>>only send and receive e-mails on the basis that we are not 
>>
>>liable for 
>>
>>>>any such corruption, interception, amendment, tampering or 
>>
>>viruses or 
>>
>>>>any consequences thereof.
>>>>
>>>>
>>
>>-- 
>>Gonzalo Camarillo         Phone :  +358  9 299 33 71
>>Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
>>Telecom R&D               Fax   :  +358  9 299 30 52
>>FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
>>Finland                   http://www.hut.fi/~gonzalo
>>
>>
>>This communication is confidential and intended solely for 
>>the addressee(s). Any unauthorized review, use, disclosure or 
>>distribution is prohibited. If you believe this message has 
>>been sent to you in error, please notify the sender by 
>>replying to this transmission and delete the message without 
>>disclosing it. Thank you.
>>
>>E-mail including attachments is susceptible to data 
>>corruption, interruption, unauthorized amendment, tampering 
>>and viruses, and we only send and receive e-mails on the 
>>basis that we are not liable for any such corruption, 
>>interception, amendment, tampering or viruses or any 
>>consequences thereof.
>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>>

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


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Thread-Topic: sip-history-info-02: counter starter
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From: "Eric Burger" <eburger@snowshore.com>
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Subject: [Sip] sip-history-info-02: counter starter
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In section 4.3.3.1, the draft states "The index for this entry is =
RECOMMENDED to start at 1."

On the one hand, why would the first proxy start at anything other than =
1?

On the other hand, I could see doing it for debugging purposes: I set =
proxy A to start at 1000 and proxy B to start at 100000, and then I get =
an extra level of (non-interoperable, but yet still useful for debugging =
purposes) history information.

I would either state some reasons why it is only recommended (or =
should), or make it a must.  Personally, I like debugging stuff, so I =
would leave it as is.

rjs: I would also plan on a torture test where a proxy receives a =
message with a history-info index of, say, 32767 :-)


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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Robert Sparks <rsparks@dynamicsoft.com>
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Subject: Re: [Sip] Moving forward on non-invite issues
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Its late, I'm tired, so there is a good chance that what I am saying 
below makes no sense.

It occurs to me that there is another solution (and I dont recall if we 
discussed it), which can alleviate many of the problems in the problems 
draft. The idea is that we define two separate NIT timeouts - one for a 
client or proxy, and one for a server that generates its own responses 
(as opposed to forwarding ones generated by others). The server one is 
1/2 the size of the client one. We could, for example, keep the client 
one at 32 and make the server one 16.

In this case, a UAS would HAVE to send a 408 if it cant complete the 
transaction in 16s. However, this 408 will not arrive too late at any 
upstream elements, since all of them have still 16s remaining.

AFAICT, this approach has the following impacts on the problems in your 
problems draft:

1.1 is solved. You reply immediately, if not, a UAS has to reply within 
16s with a 408. The race then becomes possibly if it takes 16 for the 
response to propagate - if thats a problem, then the noninvite 
transaction duration itself is too small for that network.

1.2 is not solved. I see this is quite separate from the other problems. 
TCP does solve it tho.

1.3 is solved. You should get a response now before the transaction 
completes.

1.4 is solved. 408 is still useful. Its always sent by the uas before 
the upstream transactions terminate.

1.5 is sort of solved, but not completely. If an element has failed, 
you'll not get a 408 at all, so the proxy still waits too long to get a 
response from the failed element. Need to think about that some more.

1.6 is still a problem; thats fundamentally unsolvable without some kind 
of negotiation or mandatory minimum.


This is, to some degree, a vairation on the Timeleft approach in your 
futures draft.

-Jonathan R.





Robert Sparks wrote:

> All -
> 
> At the request of the chairs, I split the non-invite
> draft into three parts :
> 
> - the description of the problems
> - the things we seem to have consensus to fix now
> - the things we might do in the future but need to
>   argue over more.
> 
> 
> The resulting drafts are available in the repository:
> http://www.ietf.org/internet-drafts/draft-sparks-sip-nit-problems-00.txt
> http://www.ietf.org/internet-drafts/draft-sparks-sip-nit-actions-00.txt
> http://www.ietf.org/internet-drafts/draft-sparks-sip-nit-future-00.txt
> 
> 
> My recommendation is to LC the problems and actions drafts
> as soon as it is feasible to do so and send them to the IESG.
> 
> RjS
> 
> ps - if you prefer reading xml2rfc's html output, you
> can get that at:
> http://www.nostrum.com/~rjsparks/draft-sparks-sip-nit-problems-00.html
> http://www.nostrum.com/~rjsparks/draft-sparks-sip-nit-actions-00.html
> http://www.nostrum.com/~rjsparks/draft-sparks-sip-nit-future-00.html
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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: "Mary Barnes" <mary.barnes@nortelnetworks.com>
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Subject: [Sip] RE: sip-history-info-02: counter starter
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That's a good point;  I honestly can't remember if there was a specific
reason why we had a RECOMMENDED rather than a MUST.  But,  I don't think it
would necessarily cause interoperability problems, it just would make it a
bit more difficult for the end applications to handle as they would likely
have to normalize the indices back to 1 (I can't see the value in an
application in wanting to use or know that different proxies might start at
different indices and that certainly wasn't the intent).  So, unless someone
has a compelling reason why we should keep it as RECOMMENDED, I'll make the
change to MUST in the next rev.

Mary.

-----Original Message-----
From: Eric Burger [mailto:eburger@snowshore.com] 
Sent: Sunday, February 29, 2004 7:31 AM
To: Barnes, Mary [NGC:B622:EXCH]
Cc: SIP List (E-mail)
Subject: sip-history-info-02: counter starter


In section 4.3.3.1, the draft states "The index for this entry is
RECOMMENDED to start at 1."

On the one hand, why would the first proxy start at anything other than 1?

On the other hand, I could see doing it for debugging purposes: I set proxy
A to start at 1000 and proxy B to start at 100000, and then I get an extra
level of (non-interoperable, but yet still useful for debugging purposes)
history information.

I would either state some reasons why it is only recommended (or should), or
make it a must.  Personally, I like debugging stuff, so I would leave it as
is.

rjs: I would also plan on a torture test where a proxy receives a message
with a history-info index of, say, 32767 :-)


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charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>RE: sip-history-info-02: counter starter</TITLE>
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<BODY>

<P><FONT SIZE=3D2>That's a good point;&nbsp; I honestly can't remember =
if there was a specific reason why we had a RECOMMENDED rather than a =
MUST.&nbsp; But,&nbsp; I don't think it would necessarily cause =
interoperability problems, it just would make it a bit more difficult =
for the end applications to handle as they would likely have to =
normalize the indices back to 1 (I can't see the value in an =
application in wanting to use or know that different proxies might =
start at different indices and that certainly wasn't the intent).&nbsp; =
So, unless someone has a compelling reason why we should keep it as =
RECOMMENDED, I'll make the change to MUST in the next rev.</FONT></P>

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

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Eric Burger [<A =
HREF=3D"mailto:eburger@snowshore.com">mailto:eburger@snowshore.com</A>] =
</FONT>
<BR><FONT SIZE=3D2>Sent: Sunday, February 29, 2004 7:31 AM</FONT>
<BR><FONT SIZE=3D2>To: Barnes, Mary [NGC:B622:EXCH]</FONT>
<BR><FONT SIZE=3D2>Cc: SIP List (E-mail)</FONT>
<BR><FONT SIZE=3D2>Subject: sip-history-info-02: counter starter</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>In section 4.3.3.1, the draft states &quot;The index =
for this entry is RECOMMENDED to start at 1.&quot;</FONT>
</P>

<P><FONT SIZE=3D2>On the one hand, why would the first proxy start at =
anything other than 1?</FONT>
</P>

<P><FONT SIZE=3D2>On the other hand, I could see doing it for debugging =
purposes: I set proxy A to start at 1000 and proxy B to start at =
100000, and then I get an extra level of (non-interoperable, but yet =
still useful for debugging purposes) history information.</FONT></P>

<P><FONT SIZE=3D2>I would either state some reasons why it is only =
recommended (or should), or make it a must.&nbsp; Personally, I like =
debugging stuff, so I would leave it as is.</FONT></P>

<P><FONT SIZE=3D2>rjs: I would also plan on a torture test where a =
proxy receives a message with a history-info index of, say, 32767 =
:-)</FONT>
</P>

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</HTML>
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Subject: Re: [Sip] Re: INFO Documentation (was Re: Defining new requests)
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At least one possible use case is for location updates during emergency 
calls. I'll discuss this option during the relevant SIPPING segment. (At 
this point, I'm not arguing that this is how it should be done, just 
that it as a plausible option.)

Gonzalo Camarillo wrote:

> Keith,
> 
> OK, I re-read the thread and I have noticed that you were referencing my 
> last email. What I actually meant was that we should see whether we 
> deprecate INFO so that it only transports PSTN stuff, or, on the other 
> hand, we define INFO packages. We had some discussions about that some 
> time ago.
> 
> So, we need to discuss which requirements we need to fulfill, and that 
> should be done in SIPPING. The actual document updating INFO will be 
> done in SIP, as you suggest.


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