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Date: Fri, 01 May 2009 08:10:54 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Question on Require in responses
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Paul,

I am not disagreeing with you, but a consequence of this analysis is
that many option tags, or the way they are intended to be used,
guarantee very little when they appear in Supported. The only time an
option tag has real meaning in Supported is when it appears in a request
to mean that the UAS can safely use that capability when responding.
Examples of option tags used differently from this:
- norefersub - its use in one request means that the UA is capable of
accepting a subsequent REFER request without implicit subscription;
- join/replaces - use of either of these in one request indicates that
the UA would support the respective header field if received in another
request;
- from-change - use in one request indicates support for this feature in
subsequent requests on that dialog;
- eventlist - use in a SUBSCRIBE request indicates support for this
feature in received NOTIFY requests on dialogs created from that
SUBSCRIBE request;
- answermode - use in a REGISTER request indicates support for this
feature in a received INVITE request.

Even if there are no absolute guarantees, there is a general expectation
that things will work if used in the way described in the RFC where the
option tag is defined. For example, if receipt of a replaces option tag
does not mean that the Replaces header field is likely to be supported
if received in future request from outside the dialog, things like
attended call transfer would not work with any degree of reliability.

So I think the scope of an option tag is defined by the RFC where the
option tag is defined.

John

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 01 May 2009 00:23
> To: Hans Erik van Elburg
> Cc: SIPCORE
> Subject: Re: [sipcore] Question on Require in responses
>=20
>=20
>=20
> Hans Erik van Elburg wrote:
> >=20
> > Dean Willis wrote:
> >> The option tag is a means of 1) discovering whether or not=20
> the far-end=20
> >> supports a specific extension and of 2) declaring the=20
> criticality of=20
> >> understanding that extension relative to satisfying a=20
> request and 3)=20
> >> informing the UAC of what extensions were applied to the=20
> request in=20
> >> order to generate the response. This is purely a function of=20
> >> capability negotiation, not one of invocation.
> >>
> >> The extension itself (typically, an extension header=20
> field, although=20
> >> there are other types of extensions) is what causes the=20
> extension to=20
> >> be invoked.
> > Depends a bit how you define extension, but I would argue that if I=20
> > require an extension to be supported on a=20
> transaction/dialog by the UAS=20
> > and the UAS accepts the request that this means that for=20
> this call the=20
> > UAS behaves as it should according to the extensions specification.
>=20
> There are several issues in what you say above:
>=20
> "if I require an extension to be supported on a=20
> transaction/*dialog* by=20
> the UAS"
>=20
> supported for a transaction, and supported for a dialog are two very=20
> different things. We certainly don't have a way to say=20
> whether you want=20
> one or the other.
>=20
> These headers were derived from http, and there they really=20
> only apply=20
> for one transaction. Nominally that is all they mean for sip too.
>=20
> In practice, particular features really have a broader scope=20
> than that,=20
> but there is little/no formality about that.
>=20
> "if... and the UAS accepts the request that means that *for=20
> this call*=20
> the UAS behaves as it should according to the extensions=20
> specification"
>=20
> There you have assumed that the scope is, depending on your=20
> terminology,
> - the call
> - the session
> - the dialog
> - the dialog usage
>=20
> But in general you cannot assume that. The support may cease=20
> on the next=20
> transaction, explicitly or implicitly. It is far from clear that=20
> support, or requirements to support, continue if not restated=20
> in every=20
> transaction. Nor is there any guarantee that support, if indicated in=20
> the response to a transaction, will continue even momentarily=20
> before the=20
> next transaction.
>=20
> And of course there is no guarantee that features that are=20
> indicated as=20
> supported in response to an OPTIONS will continue to be=20
> supported when a=20
> call is made to the same address.
>=20
> 	Thanks,
> 	Paul
>=20
> > Saying that an extensions invokes itself is meaningless to me.
> >=20
> > A UAS may choose to only invoke the extension behaviour if=20
> it finds that=20
> > the extension is required. That comes close to saying that the=20
> > require:extension invokes the behaviour in the UAS, but it=20
> is not at all=20
> > the same.
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Question on Require in responses
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John,

I'm not trying to assert that the situation is a good one. But I have 
been trying to achieve a general understanding that you cannot, 
currently, infer too much from these headers. Do so and you will set 
yourself up for disappointment.

Perhaps we should work define the scope for these things, and where 
appropriate specify that it is more extensive than a single transaction. 
But I think it will be impossible to make any commitments that much of 
anything has dialog scope. I think the most we can hope for is: this 
remains true until a change is signaled.

	Thanks,
	Paul

Elwell, John wrote:
> Paul,
> 
> I am not disagreeing with you, but a consequence of this analysis is
> that many option tags, or the way they are intended to be used,
> guarantee very little when they appear in Supported. The only time an
> option tag has real meaning in Supported is when it appears in a request
> to mean that the UAS can safely use that capability when responding.
> Examples of option tags used differently from this:
> - norefersub - its use in one request means that the UA is capable of
> accepting a subsequent REFER request without implicit subscription;
> - join/replaces - use of either of these in one request indicates that
> the UA would support the respective header field if received in another
> request;
> - from-change - use in one request indicates support for this feature in
> subsequent requests on that dialog;
> - eventlist - use in a SUBSCRIBE request indicates support for this
> feature in received NOTIFY requests on dialogs created from that
> SUBSCRIBE request;
> - answermode - use in a REGISTER request indicates support for this
> feature in a received INVITE request.
> 
> Even if there are no absolute guarantees, there is a general expectation
> that things will work if used in the way described in the RFC where the
> option tag is defined. For example, if receipt of a replaces option tag
> does not mean that the Replaces header field is likely to be supported
> if received in future request from outside the dialog, things like
> attended call transfer would not work with any degree of reliability.
> 
> So I think the scope of an option tag is defined by the RFC where the
> option tag is defined.
> 
> John
> 
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org 
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
>> Sent: 01 May 2009 00:23
>> To: Hans Erik van Elburg
>> Cc: SIPCORE
>> Subject: Re: [sipcore] Question on Require in responses
>>
>>
>>
>> Hans Erik van Elburg wrote:
>>> Dean Willis wrote:
>>>> The option tag is a means of 1) discovering whether or not 
>> the far-end 
>>>> supports a specific extension and of 2) declaring the 
>> criticality of 
>>>> understanding that extension relative to satisfying a 
>> request and 3) 
>>>> informing the UAC of what extensions were applied to the 
>> request in 
>>>> order to generate the response. This is purely a function of 
>>>> capability negotiation, not one of invocation.
>>>>
>>>> The extension itself (typically, an extension header 
>> field, although 
>>>> there are other types of extensions) is what causes the 
>> extension to 
>>>> be invoked.
>>> Depends a bit how you define extension, but I would argue that if I 
>>> require an extension to be supported on a 
>> transaction/dialog by the UAS 
>>> and the UAS accepts the request that this means that for 
>> this call the 
>>> UAS behaves as it should according to the extensions specification.
>> There are several issues in what you say above:
>>
>> "if I require an extension to be supported on a 
>> transaction/*dialog* by 
>> the UAS"
>>
>> supported for a transaction, and supported for a dialog are two very 
>> different things. We certainly don't have a way to say 
>> whether you want 
>> one or the other.
>>
>> These headers were derived from http, and there they really 
>> only apply 
>> for one transaction. Nominally that is all they mean for sip too.
>>
>> In practice, particular features really have a broader scope 
>> than that, 
>> but there is little/no formality about that.
>>
>> "if... and the UAS accepts the request that means that *for 
>> this call* 
>> the UAS behaves as it should according to the extensions 
>> specification"
>>
>> There you have assumed that the scope is, depending on your 
>> terminology,
>> - the call
>> - the session
>> - the dialog
>> - the dialog usage
>>
>> But in general you cannot assume that. The support may cease 
>> on the next 
>> transaction, explicitly or implicitly. It is far from clear that 
>> support, or requirements to support, continue if not restated 
>> in every 
>> transaction. Nor is there any guarantee that support, if indicated in 
>> the response to a transaction, will continue even momentarily 
>> before the 
>> next transaction.
>>
>> And of course there is no guarantee that features that are 
>> indicated as 
>> supported in response to an OPTIONS will continue to be 
>> supported when a 
>> call is made to the same address.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Saying that an extensions invokes itself is meaningless to me.
>>>
>>> A UAS may choose to only invoke the extension behaviour if 
>> it finds that 
>>> the extension is required. That comes close to saying that the 
>>> require:extension invokes the behaviour in the UAS, but it 
>> is not at all 
>>> the same.
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
> 

From eburger@standardstrack.com  Fri May  1 06:13:21 2009
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] info-03 draft and "session negotiation exchange"
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Section 3 is where "negotiation" occurs.  I put negotiation in quotes  
because it is an advertising model, not an offer/answer model.

Would changing negotiation language to advertising language reduce  
confusion, or is it semantic dressing?

On Apr 30, 2009, at 2:30 AM, Attila Sipos wrote:

> Hi,
>
> Sorry for questions late in the day but I'm not clear on the
> negotiation methods of the info package draft.
>
> The draft mentions "session negotiation exchange" several times.
>
> Can someone tell me where this is defined?
>
> Section 3.1 (UA Behavior) shows INVITE/OK/ACK as an exchange.
>
> I see from section "5.2.  INFO Headers" of the "INFO tables" e-mail
> that PRACK and UPDATE are allowed to carry the Recv-Info header. So
> what is the "exchange" if say, INVITE and PRACK are involved?
>
> From section 3.1, it seems that the initial "offer" is in the INVITE
> and then "confirmed" in the ACK. So if PRACK is received, can one
> "confirm" in a PRACK?
>
> Also, if Recv-Info is allowed in UPDATE, how does that get  
> "confirmed"?
> Or is it that UPDATE just lists the packages already negotiated?
>
> Or is it that package renegotiation is only ever done in
> INVITE/re-INVITE?
> If so, I would like to see some text explicitly stating this.
> If not, what renegotiation combinations are there?
>
> Then there is this...
>   The limitation on requiring the negotiation to occur within the
>   context of a session negotiation exchange means that if the
>   initiating UA issues a re-INVITE (after the above ACK) with the
>   following.
>
>   INVITE ...
>   ...
>   Recv-Info: P, R, T
>   ...
>
>   The target UA MUST NOT send any package P INFO methods until the
>   target UA sees P in the final ACK from the initiating UA.
>
>
> So, let's say P was being sent by the target.   Then when the
> re-INVITE arrives, P has to be stopped (or queued), until P is
> seen again in the ACK.
>
> So what happens if there is a re-INVITE/INFO(package P) crossover?
> I think we need to say that if the initiating UA receives the INFO
> for package P, it should either:
> 1. still process it as normal
> or
> 2. issue a 4xx response (something like 491 Request Pending)
>   so that the package P event can be re-attempted once the ACK
>   has been received.
>
> Regards,
>
> Attila
>
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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References: <0D5F89FAC29E2C41B98A6A762007F5D001D4A39A@GBNTHT12009MSX.gb002.siemens.net> <49F71501.2020300@cisco.com> <49F742EE.30504@softarmor.com> <49F75022.4010905@cisco.com> <6B78E793-0C48-427E-8895-337D5F592957@softarmor.com> <CA9998CD4A020D418654FCDEF4E707DF0CA9565E@esealmw113.eemea.ericsson.se> <7C450C3A-33F9-450D-B424-4DCD86D5D99A@softarmor.com> <49F9BE92.1050601@gmail.com> <49FA32E9.3070509@cisco.com> <0D5F89FAC29E2C41B98A6A762007F5D001D8D5F2@GBNTHT12009MSX.gb002.siemens.net>
Date: Fri, 1 May 2009 16:46:15 +0200
Message-ID: <9ae56b1e0905010746g233b991bg35d9e66d21c1804c@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Question on Require in responses
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Yes.

/Hans Erik van Elburg


On Fri, May 1, 2009 at 9:10 AM, Elwell, John <
john.elwell@siemens-enterprise.com> wrote:

> Paul,
>
> I am not disagreeing with you, but a consequence of this analysis is
> that many option tags, or the way they are intended to be used,
> guarantee very little when they appear in Supported. The only time an
> option tag has real meaning in Supported is when it appears in a request
> to mean that the UAS can safely use that capability when responding.
> Examples of option tags used differently from this:
> - norefersub - its use in one request means that the UA is capable of
> accepting a subsequent REFER request without implicit subscription;
> - join/replaces - use of either of these in one request indicates that
> the UA would support the respective header field if received in another
> request;
> - from-change - use in one request indicates support for this feature in
> subsequent requests on that dialog;
> - eventlist - use in a SUBSCRIBE request indicates support for this
> feature in received NOTIFY requests on dialogs created from that
> SUBSCRIBE request;
> - answermode - use in a REGISTER request indicates support for this
> feature in a received INVITE request.
>
> Even if there are no absolute guarantees, there is a general expectation
> that things will work if used in the way described in the RFC where the
> option tag is defined. For example, if receipt of a replaces option tag
> does not mean that the Replaces header field is likely to be supported
> if received in future request from outside the dialog, things like
> attended call transfer would not work with any degree of reliability.
>
> So I think the scope of an option tag is defined by the RFC where the
> option tag is defined.
>
> John
>
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
> > Sent: 01 May 2009 00:23
> > To: Hans Erik van Elburg
> > Cc: SIPCORE
> > Subject: Re: [sipcore] Question on Require in responses
> >
> >
> >
> > Hans Erik van Elburg wrote:
> > >
> > > Dean Willis wrote:
> > >> The option tag is a means of 1) discovering whether or not
> > the far-end
> > >> supports a specific extension and of 2) declaring the
> > criticality of
> > >> understanding that extension relative to satisfying a
> > request and 3)
> > >> informing the UAC of what extensions were applied to the
> > request in
> > >> order to generate the response. This is purely a function of
> > >> capability negotiation, not one of invocation.
> > >>
> > >> The extension itself (typically, an extension header
> > field, although
> > >> there are other types of extensions) is what causes the
> > extension to
> > >> be invoked.
> > > Depends a bit how you define extension, but I would argue that if I
> > > require an extension to be supported on a
> > transaction/dialog by the UAS
> > > and the UAS accepts the request that this means that for
> > this call the
> > > UAS behaves as it should according to the extensions specification.
> >
> > There are several issues in what you say above:
> >
> > "if I require an extension to be supported on a
> > transaction/*dialog* by
> > the UAS"
> >
> > supported for a transaction, and supported for a dialog are two very
> > different things. We certainly don't have a way to say
> > whether you want
> > one or the other.
> >
> > These headers were derived from http, and there they really
> > only apply
> > for one transaction. Nominally that is all they mean for sip too.
> >
> > In practice, particular features really have a broader scope
> > than that,
> > but there is little/no formality about that.
> >
> > "if... and the UAS accepts the request that means that *for
> > this call*
> > the UAS behaves as it should according to the extensions
> > specification"
> >
> > There you have assumed that the scope is, depending on your
> > terminology,
> > - the call
> > - the session
> > - the dialog
> > - the dialog usage
> >
> > But in general you cannot assume that. The support may cease
> > on the next
> > transaction, explicitly or implicitly. It is far from clear that
> > support, or requirements to support, continue if not restated
> > in every
> > transaction. Nor is there any guarantee that support, if indicated in
> > the response to a transaction, will continue even momentarily
> > before the
> > next transaction.
> >
> > And of course there is no guarantee that features that are
> > indicated as
> > supported in response to an OPTIONS will continue to be
> > supported when a
> > call is made to the same address.
> >
> >       Thanks,
> >       Paul
> >
> > > Saying that an extensions invokes itself is meaningless to me.
> > >
> > > A UAS may choose to only invoke the extension behaviour if
> > it finds that
> > > the extension is required. That comes close to saying that the
> > > require:extension invokes the behaviour in the UAS, but it
> > is not at all
> > > the same.
> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
>

--00163646db76ec40560468dadf8d
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Yes.<br><br clear=3D"all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Fri, May 1, 2009 at 9:10 AM, Elwell, =
John <span dir=3D"ltr">&lt;<a href=3D"mailto:john.elwell@siemens-enterprise=
.com">john.elwell@siemens-enterprise.com</a>&gt;</span> wrote:<br><blockquo=
te class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204)=
; margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Paul,<br>
<br>
I am not disagreeing with you, but a consequence of this analysis is<br>
that many option tags, or the way they are intended to be used,<br>
guarantee very little when they appear in Supported. The only time an<br>
option tag has real meaning in Supported is when it appears in a request<br=
>
to mean that the UAS can safely use that capability when responding.<br>
Examples of option tags used differently from this:<br>
- norefersub - its use in one request means that the UA is capable of<br>
accepting a subsequent REFER request without implicit subscription;<br>
- join/replaces - use of either of these in one request indicates that<br>
the UA would support the respective header field if received in another<br>
request;<br>
- from-change - use in one request indicates support for this feature in<br=
>
subsequent requests on that dialog;<br>
- eventlist - use in a SUBSCRIBE request indicates support for this<br>
feature in received NOTIFY requests on dialogs created from that<br>
SUBSCRIBE request;<br>
- answermode - use in a REGISTER request indicates support for this<br>
feature in a received INVITE request.<br>
<br>
Even if there are no absolute guarantees, there is a general expectation<br=
>
that things will work if used in the way described in the RFC where the<br>
option tag is defined. For example, if receipt of a replaces option tag<br>
does not mean that the Replaces header field is likely to be supported<br>
if received in future request from outside the dialog, things like<br>
attended call transfer would not work with any degree of reliability.<br>
<br>
So I think the scope of an option tag is defined by the RFC where the<br>
option tag is defined.<br>
<font color=3D"#888888"><br>
John<br>
</font><div class=3D"im"><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf=
.org</a><br>
&gt; [mailto:<a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ie=
tf.org</a>] On Behalf Of Paul Kyzivat<br>
</div><div class=3D"im">&gt; Sent: 01 May 2009 00:23<br>
&gt; To: Hans Erik van Elburg<br>
&gt; Cc: SIPCORE<br>
&gt; Subject: Re: [sipcore] Question on Require in responses<br>
&gt;<br>
&gt;<br>
&gt;<br>
</div><div><div></div><div class=3D"h5">&gt; Hans Erik van Elburg wrote:<br=
>
&gt; &gt;<br>
&gt; &gt; Dean Willis wrote:<br>
&gt; &gt;&gt; The option tag is a means of 1) discovering whether or not<br=
>
&gt; the far-end<br>
&gt; &gt;&gt; supports a specific extension and of 2) declaring the<br>
&gt; criticality of<br>
&gt; &gt;&gt; understanding that extension relative to satisfying a<br>
&gt; request and 3)<br>
&gt; &gt;&gt; informing the UAC of what extensions were applied to the<br>
&gt; request in<br>
&gt; &gt;&gt; order to generate the response. This is purely a function of<=
br>
&gt; &gt;&gt; capability negotiation, not one of invocation.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; The extension itself (typically, an extension header<br>
&gt; field, although<br>
&gt; &gt;&gt; there are other types of extensions) is what causes the<br>
&gt; extension to<br>
&gt; &gt;&gt; be invoked.<br>
&gt; &gt; Depends a bit how you define extension, but I would argue that if=
 I<br>
&gt; &gt; require an extension to be supported on a<br>
&gt; transaction/dialog by the UAS<br>
&gt; &gt; and the UAS accepts the request that this means that for<br>
&gt; this call the<br>
&gt; &gt; UAS behaves as it should according to the extensions specificatio=
n.<br>
&gt;<br>
&gt; There are several issues in what you say above:<br>
&gt;<br>
&gt; &quot;if I require an extension to be supported on a<br>
&gt; transaction/*dialog* by<br>
&gt; the UAS&quot;<br>
&gt;<br>
&gt; supported for a transaction, and supported for a dialog are two very<b=
r>
&gt; different things. We certainly don&#39;t have a way to say<br>
&gt; whether you want<br>
&gt; one or the other.<br>
&gt;<br>
&gt; These headers were derived from http, and there they really<br>
&gt; only apply<br>
&gt; for one transaction. Nominally that is all they mean for sip too.<br>
&gt;<br>
&gt; In practice, particular features really have a broader scope<br>
&gt; than that,<br>
&gt; but there is little/no formality about that.<br>
&gt;<br>
&gt; &quot;if... and the UAS accepts the request that means that *for<br>
&gt; this call*<br>
&gt; the UAS behaves as it should according to the extensions<br>
&gt; specification&quot;<br>
&gt;<br>
&gt; There you have assumed that the scope is, depending on your<br>
&gt; terminology,<br>
&gt; - the call<br>
&gt; - the session<br>
&gt; - the dialog<br>
&gt; - the dialog usage<br>
&gt;<br>
&gt; But in general you cannot assume that. The support may cease<br>
&gt; on the next<br>
&gt; transaction, explicitly or implicitly. It is far from clear that<br>
&gt; support, or requirements to support, continue if not restated<br>
&gt; in every<br>
&gt; transaction. Nor is there any guarantee that support, if indicated in<=
br>
&gt; the response to a transaction, will continue even momentarily<br>
&gt; before the<br>
&gt; next transaction.<br>
&gt;<br>
&gt; And of course there is no guarantee that features that are<br>
&gt; indicated as<br>
&gt; supported in response to an OPTIONS will continue to be<br>
&gt; supported when a<br>
&gt; call is made to the same address.<br>
&gt;<br>
&gt; =A0 =A0 =A0 Thanks,<br>
&gt; =A0 =A0 =A0 Paul<br>
&gt;<br>
&gt; &gt; Saying that an extensions invokes itself is meaningless to me.<br=
>
&gt; &gt;<br>
&gt; &gt; A UAS may choose to only invoke the extension behaviour if<br>
&gt; it finds that<br>
&gt; &gt; the extension is required. That comes close to saying that the<br=
>
&gt; &gt; require:extension invokes the behaviour in the UAS, but it<br>
&gt; is not at all<br>
&gt; &gt; the same.<br>
&gt;<br>
</div></div><div><div></div><div class=3D"h5">&gt; ________________________=
_______________________<br>
&gt; sipcore mailing list<br>
&gt; <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_b=
lank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
&gt;<br>
</div></div></blockquote></div><br>

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References: <49F178CE.8000809@ericsson.com> <0ECC036E-2B21-49D5-AC42-1FEC8EC447BE@softarmor.com> <49F5D64B.20008@nostrum.com> <EE242275-D5A4-4C14-83A3-B945B8B0D233@softarmor.com> <193E3797-2C8D-44EB-A869-DDDE39B303D9@nostrum.com> <06F3B3D8-BEA3-4B5A-990D-B9E8DCAB1FF1@softarmor.com> <49F9718E.4070707@gmail.com> <9A564A10-3F22-471A-832B-28117412AD93@softarmor.com> <49F9F343.6090901@gmail.com> <49FA50CD.1080406@softarmor.com>
Date: Fri, 1 May 2009 17:08:21 +0200
Message-ID: <9ae56b1e0905010808s50a236dcge61b9f8250dfec9@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Dean Willis <dean.willis@softarmor.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Long reply to: Target URI and History Info bis
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It is not clear what you are disagreeing with. Because I agree that "the
goal is to make the information that was encoded into the requestURI (and
maybe as modified by intermediates, that use case is open) available to the
UAS.".

But this is a little imprecise as it doen't distinguish between the
Request-URI as it was crafted initially by the UAC and those that came about
due to request URI rewrites that happened because of routing and those that
happened due to the communication being redirected/forwarded to another user
then the one originally addressed.

If UA A sends an INVITE to URI of user B, but this call is forwarded on the
way to URI of user C. Then to C the Request URI value that was used by the
forwarding entity (URI of user C) is more relevant URI to decide how it will
render the call to the user then the URI of user B.

/Hans Erik van Elburg

On Fri, May 1, 2009 at 3:30 AM, Dean Willis <dean.willis@softarmor.com>wrote:

> Hans Erik van Elburg wrote:
> >
> >
> > Dean Willis wrote:
> >>
> >> On Apr 30, 2009, at 4:38 AM, Hans Erik van Elburg wrote:
> >>
> >>> Overloading in general is not a good design strategy, why do you
> >>> think that trying to mold everything (diversion/history info) into
> >>> Request URI is a good idea??
> >>
> >> The reason that we're having this conversation is that we have
> >> proposals, History-Info being one, for 1) putting our parameters in to
> >> the original request-URI, and 2) recovering that original request-URI
> >> after it has been contact-routed (and otherwise mangled) so that the
> >> parameters can be used at the UAS.
> >>
> >> So the presumption exists that the original request URI is useful to
> >> the UAS. SIP did, and now SIPCORE has, a chartered item for making
> >> that happen.
> >>
> >> I'm trying to back up and make sure that we know WHY we're trying to
> >> deliver the original request URI to the UAS, and then asking if some
> >> other approach that meets the same requirements might actually be
> >> easier to do.
> >>
> > That is fine of course, but the point is that we are not talking about
> > delivering parameters to an application, but about recovering a complete
> > URI that has gone lost on the way.
>
> but why exactly do we need to recover the URI? What is it that is
> special about the URI or is likely to get used in some useful sort of
> way downstream? If it isn't the "Request URI-ness" that is important,
> maybe we can find some more convenient way to deliver that information.
>
>
> >>> The problem here is that this information has nothing to do with
> >>> application invocation, but tells something about how the signalling
> >>> reached the callee (target). And this information is only relevant to
> >>> the callee if that decides that it is.
> >>
> >>
> >> That's a profound philosophical assertion. HTTP, for example, makes no
> >> consideration for "how the signaling reaches the target" in
> >> determining how to process a request. The fact that some think SIP
> >> should take this into consideration indicates to me that we have a
> >> layering problem in our design.
> >>
> > I don't think SIP and HTTP are comparable at all in this respect. In
> > HTTP you directly address the server that you like to get (post, delete,
> > update) information from. In SIP you adress the URI of a user, that will
> > lead you to the home proxy in most cases and which is then forwarded to
> > the UAS by overwriting the Request URI.
>
> Well, HTTP has proxies too, and SIP were originally intended to be
> comparable.
>
> '> No layering problem, just a flaw in SIP's design that causes usefull
> > information to be destroyed for no good reason. We're just trying to fix
> > that, no more no less.
> >> From one perspective, this makes the application invocation implicit;
> >> the UAS has to guess what to do based on clues sprinkled throughout
> >> the communication and across several layers of the ISO stack. It's
> >> much better to have this sort of determination made based on explicit
> >> indicators at a single application layer.
> >>
> > No. The goal of the target-uri discussion is to have all the relevant
> > target-uri information readily available in the SIP request received by
> > the UAS. That is all on the same OSI layer, layer 7.
> >
>
> I disgaree -- the goal is to make the information that was encoded into
> the requestURI (and maybe as modified by intermediates, that use case is
> open) available to the UAS.
>
>
> --
> dean
>

--0016367b6de6eba32f0468db2e49
Content-Type: text/html; charset=ISO-8859-1
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It is not clear what you are disagreeing with. Because I agree that &quot;t=
he goal is to make the information that was encoded into the requestURI (an=
d maybe as modified by intermediates, that use case is open) available to t=
he UAS.&quot;.<br>
<br>But this is a little imprecise as it doen&#39;t distinguish between the=
 Request-URI as it was crafted initially by the UAC and those that came abo=
ut due to request URI rewrites that happened because of routing and those t=
hat=A0 happened due to the communication being redirected/forwarded to anot=
her user then the one originally addressed.=A0 <br>
<br>If UA A sends an INVITE to URI of user B, but this call is forwarded on=
 the way to URI of user C. Then to C the Request URI value that was used by=
 the forwarding entity (URI of user C) is more relevant URI to decide how i=
t will render the call to the user then the URI of user B. <br>
<br clear=3D"all">/Hans Erik van Elburg<br>
<br><div class=3D"gmail_quote">On Fri, May 1, 2009 at 3:30 AM, Dean Willis =
<span dir=3D"ltr">&lt;<a href=3D"mailto:dean.willis@softarmor.com">dean.wil=
lis@softarmor.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0=
.8ex; padding-left: 1ex;">
<div class=3D"im">Hans Erik van Elburg wrote:<br>
&gt;<br>
&gt;<br>
&gt; Dean Willis wrote:<br>
&gt;&gt;<br>
&gt;&gt; On Apr 30, 2009, at 4:38 AM, Hans Erik van Elburg wrote:<br>
&gt;&gt;<br>
&gt;&gt;&gt; Overloading in general is not a good design strategy, why do y=
ou<br>
&gt;&gt;&gt; think that trying to mold everything (diversion/history info) =
into<br>
&gt;&gt;&gt; Request URI is a good idea??<br>
&gt;&gt;<br>
&gt;&gt; The reason that we&#39;re having this conversation is that we have=
<br>
&gt;&gt; proposals, History-Info being one, for 1) putting our parameters i=
n to<br>
&gt;&gt; the original request-URI, and 2) recovering that original request-=
URI<br>
&gt;&gt; after it has been contact-routed (and otherwise mangled) so that t=
he<br>
&gt;&gt; parameters can be used at the UAS.<br>
&gt;&gt;<br>
&gt;&gt; So the presumption exists that the original request URI is useful =
to<br>
&gt;&gt; the UAS. SIP did, and now SIPCORE has, a chartered item for making=
<br>
&gt;&gt; that happen.<br>
&gt;&gt;<br>
&gt;&gt; I&#39;m trying to back up and make sure that we know WHY we&#39;re=
 trying to<br>
&gt;&gt; deliver the original request URI to the UAS, and then asking if so=
me<br>
&gt;&gt; other approach that meets the same requirements might actually be<=
br>
&gt;&gt; easier to do.<br>
&gt;&gt;<br>
&gt; That is fine of course, but the point is that we are not talking about=
<br>
&gt; delivering parameters to an application, but about recovering a comple=
te<br>
&gt; URI that has gone lost on the way.<br>
<br>
</div>but why exactly do we need to recover the URI? What is it that is<br>
special about the URI or is likely to get used in some useful sort of<br>
way downstream? If it isn&#39;t the &quot;Request URI-ness&quot; that is im=
portant,<br>
maybe we can find some more convenient way to deliver that information.<br>
<div class=3D"im"><br>
<br>
&gt;&gt;&gt; The problem here is that this information has nothing to do wi=
th<br>
&gt;&gt;&gt; application invocation, but tells something about how the sign=
alling<br>
&gt;&gt;&gt; reached the callee (target). And this information is only rele=
vant to<br>
&gt;&gt;&gt; the callee if that decides that it is.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; That&#39;s a profound philosophical assertion. HTTP, for example, =
makes no<br>
&gt;&gt; consideration for &quot;how the signaling reaches the target&quot;=
 in<br>
&gt;&gt; determining how to process a request. The fact that some think SIP=
<br>
&gt;&gt; should take this into consideration indicates to me that we have a=
<br>
&gt;&gt; layering problem in our design.<br>
&gt;&gt;<br>
&gt; I don&#39;t think SIP and HTTP are comparable at all in this respect. =
In<br>
&gt; HTTP you directly address the server that you like to get (post, delet=
e,<br>
&gt; update) information from. In SIP you adress the URI of a user, that wi=
ll<br>
&gt; lead you to the home proxy in most cases and which is then forwarded t=
o<br>
&gt; the UAS by overwriting the Request URI.<br>
<br>
</div>Well, HTTP has proxies too, and SIP were originally intended to be<br=
>
comparable.<br>
<br>
&#39;&gt; No layering problem, just a flaw in SIP&#39;s design that causes =
usefull<br>
<div class=3D"im">&gt; information to be destroyed for no good reason. We&#=
39;re just trying to fix<br>
&gt; that, no more no less.<br>
&gt;&gt; From one perspective, this makes the application invocation implic=
it;<br>
&gt;&gt; the UAS has to guess what to do based on clues sprinkled throughou=
t<br>
&gt;&gt; the communication and across several layers of the ISO stack. It&#=
39;s<br>
&gt;&gt; much better to have this sort of determination made based on expli=
cit<br>
&gt;&gt; indicators at a single application layer.<br>
&gt;&gt;<br>
&gt; No. The goal of the target-uri discussion is to have all the relevant<=
br>
&gt; target-uri information readily available in the SIP request received b=
y<br>
&gt; the UAS. That is all on the same OSI layer, layer 7.<br>
&gt;<br>
<br>
</div>I disgaree -- the goal is to make the information that was encoded in=
to<br>
the requestURI (and maybe as modified by intermediates, that use case is<br=
>
open) available to the UAS.<br>
<br>
<br>
--<br>
<font color=3D"#888888">dean<br>
</font></blockquote></div><br>

--0016367b6de6eba32f0468db2e49--

From jmh@joelhalpern.com  Fri May  1 08:59:47 2009
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Date: Fri, 01 May 2009 12:01:06 -0400
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Subject: Re: [sipcore] Long reply to: Target URI and History Info bis
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Actually, the redirect case (when not done via an error to the 
originating UAC and retry from there) seems to get at the problem.

The original user specified a URI they were trying to reach, and some 
parameters.
Some of those parameters would still apply after the redirect, but some 
probably don't.
But it appears likely that no one knows which parameters are which.

And the redirector may have some parameter information to add.

And depending upon how everything was understood, and why it was being 
redirected, the destination of the redirect may or may not care about 
the original URI (may even care about the user part of the original URI) 
in full, in order to decide what to do (the voicemail examples.)

All of which leads to ???
 From where I sit, it leads back to H-I since in theory at least with 
History-Info you can reconstruct any of the information you want.  Maybe 
not easily.  But given that we have completely unspecified behaviors 
with regard to parameters and their preservation, and given that even 
the user part from the original request URI may be useful to the target, 
and any parameters that were added in the middle may be useful, ...


If we could be sure that all we wanted was original and final, then 
maybe there would be something we could do, but I am quite sure we can 
make more complicated realistic cases where that will not suffice.

Joel

Hans Erik van Elburg wrote:
> It is not clear what you are disagreeing with. Because I agree that "the 
> goal is to make the information that was encoded into the requestURI 
> (and maybe as modified by intermediates, that use case is open) 
> available to the UAS.".
> 
> But this is a little imprecise as it doen't distinguish between the 
> Request-URI as it was crafted initially by the UAC and those that came 
> about due to request URI rewrites that happened because of routing and 
> those that  happened due to the communication being redirected/forwarded 
> to another user then the one originally addressed. 
> 
> If UA A sends an INVITE to URI of user B, but this call is forwarded on 
> the way to URI of user C. Then to C the Request URI value that was used 
> by the forwarding entity (URI of user C) is more relevant URI to decide 
> how it will render the call to the user then the URI of user B.
> 
> /Hans Erik van Elburg
> 
> On Fri, May 1, 2009 at 3:30 AM, Dean Willis <dean.willis@softarmor.com 
> <mailto:dean.willis@softarmor.com>> wrote:
> 
>     Hans Erik van Elburg wrote:
>      >
>      >
>      > Dean Willis wrote:
>      >>
>      >> On Apr 30, 2009, at 4:38 AM, Hans Erik van Elburg wrote:
>      >>
>      >>> Overloading in general is not a good design strategy, why do you
>      >>> think that trying to mold everything (diversion/history info) into
>      >>> Request URI is a good idea??
>      >>
>      >> The reason that we're having this conversation is that we have
>      >> proposals, History-Info being one, for 1) putting our parameters
>     in to
>      >> the original request-URI, and 2) recovering that original
>     request-URI
>      >> after it has been contact-routed (and otherwise mangled) so that the
>      >> parameters can be used at the UAS.
>      >>
>      >> So the presumption exists that the original request URI is useful to
>      >> the UAS. SIP did, and now SIPCORE has, a chartered item for making
>      >> that happen.
>      >>
>      >> I'm trying to back up and make sure that we know WHY we're trying to
>      >> deliver the original request URI to the UAS, and then asking if some
>      >> other approach that meets the same requirements might actually be
>      >> easier to do.
>      >>
>      > That is fine of course, but the point is that we are not talking
>     about
>      > delivering parameters to an application, but about recovering a
>     complete
>      > URI that has gone lost on the way.
> 
>     but why exactly do we need to recover the URI? What is it that is
>     special about the URI or is likely to get used in some useful sort of
>     way downstream? If it isn't the "Request URI-ness" that is important,
>     maybe we can find some more convenient way to deliver that information.
> 
> 
>      >>> The problem here is that this information has nothing to do with
>      >>> application invocation, but tells something about how the
>     signalling
>      >>> reached the callee (target). And this information is only
>     relevant to
>      >>> the callee if that decides that it is.
>      >>
>      >>
>      >> That's a profound philosophical assertion. HTTP, for example,
>     makes no
>      >> consideration for "how the signaling reaches the target" in
>      >> determining how to process a request. The fact that some think SIP
>      >> should take this into consideration indicates to me that we have a
>      >> layering problem in our design.
>      >>
>      > I don't think SIP and HTTP are comparable at all in this respect. In
>      > HTTP you directly address the server that you like to get (post,
>     delete,
>      > update) information from. In SIP you adress the URI of a user,
>     that will
>      > lead you to the home proxy in most cases and which is then
>     forwarded to
>      > the UAS by overwriting the Request URI.
> 
>     Well, HTTP has proxies too, and SIP were originally intended to be
>     comparable.
> 
>     '> No layering problem, just a flaw in SIP's design that causes usefull
>      > information to be destroyed for no good reason. We're just trying
>     to fix
>      > that, no more no less.
>      >> From one perspective, this makes the application invocation
>     implicit;
>      >> the UAS has to guess what to do based on clues sprinkled throughout
>      >> the communication and across several layers of the ISO stack. It's
>      >> much better to have this sort of determination made based on
>     explicit
>      >> indicators at a single application layer.
>      >>
>      > No. The goal of the target-uri discussion is to have all the relevant
>      > target-uri information readily available in the SIP request
>     received by
>      > the UAS. That is all on the same OSI layer, layer 7.
>      >
> 
>     I disgaree -- the goal is to make the information that was encoded into
>     the requestURI (and maybe as modified by intermediates, that use case is
>     open) available to the UAS.
> 
> 
>     --
>     dean
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

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From: "Attila Sipos" <attila.sipos@vegastream.com>
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Subject: Re: [sipcore] info-03 draft and "session negotiation exchange"
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>>Section 3 is where "negotiation" occurs.  I put negotiation in quotes=20
>>because it is an advertising model, not an offer/answer model.
That sentence alone, for me, makes it much clearer.
I think the confusion comes from:
1. the word negotiation
2. the statement:
      the target UA MUST NOT send any package P INFO methods until the
      target UA sees P in the final ACK from the initiating UA.

Right now, I'd like the following to be considered:
1. call it a "capability exchange" not a "negotiation" or "renegotiation".
2. have a statement at the end of the first paragraph in section 3 - someth=
ing like this:
   Note well: this is an advertising model - it is NOT a negotiation or off=
er/answer model.
3. define what set of messages constitutes an exchange (this I'm still not =
clear about - especially where PRACK is involved?  can a PRACK response be =
considered in the same way as the ACK? )
4. also can we remove the statement
      The target UA MUST NOT send any package P INFO methods until the
      target UA sees P in the final ACK from the initiating UA.
   There seems contradiction between saying it's an advertising model but m=
aking the target UA wait for the ACK.
   Making the target UA wait implies that it needs to see what the initiato=
r says again, which seems more like negotiation behavior.
    Why not just allow event P anytime after the initiator has declared it =
in the Recv-Info of the re-INVITE/INVITE?
5.  If we must keep 4, then let's define what should happen if P occurs bef=
ore the ACK (which could happen
    if the package P INFO is sent at the same time as the re-INVITE)

Regards,
Attila

=20
=20
=20

________________________________

From: Eric Burger [mailto:eburger@standardstrack.com]
Sent: Fri 01/05/2009 14:12
To: Attila Sipos
Cc: SIPCORE
Subject: Re: [sipcore] info-03 draft and "session negotiation exchange"



Section 3 is where "negotiation" occurs.  I put negotiation in quotes=20
because it is an advertising model, not an offer/answer model.

Would changing negotiation language to advertising language reduce=20
confusion, or is it semantic dressing?

On Apr 30, 2009, at 2:30 AM, Attila Sipos wrote:

> Hi,
>
> Sorry for questions late in the day but I'm not clear on the
> negotiation methods of the info package draft.
>
> The draft mentions "session negotiation exchange" several times.
>
> Can someone tell me where this is defined?
>
> Section 3.1 (UA Behavior) shows INVITE/OK/ACK as an exchange.
>
> I see from section "5.2.  INFO Headers" of the "INFO tables" e-mail
> that PRACK and UPDATE are allowed to carry the Recv-Info header. So
> what is the "exchange" if say, INVITE and PRACK are involved?
>
> From section 3.1, it seems that the initial "offer" is in the INVITE
> and then "confirmed" in the ACK. So if PRACK is received, can one
> "confirm" in a PRACK?
>
> Also, if Recv-Info is allowed in UPDATE, how does that get=20
> "confirmed"?
> Or is it that UPDATE just lists the packages already negotiated?
>
> Or is it that package renegotiation is only ever done in
> INVITE/re-INVITE?
> If so, I would like to see some text explicitly stating this.
> If not, what renegotiation combinations are there?
>
> Then there is this...
>   The limitation on requiring the negotiation to occur within the
>   context of a session negotiation exchange means that if the
>   initiating UA issues a re-INVITE (after the above ACK) with the
>   following.
>
>   INVITE ...
>   ...
>   Recv-Info: P, R, T
>   ...
>
>   The target UA MUST NOT send any package P INFO methods until the
>   target UA sees P in the final ACK from the initiating UA.
>
>
> So, let's say P was being sent by the target.   Then when the
> re-INVITE arrives, P has to be stopped (or queued), until P is
> seen again in the ACK.
>
> So what happens if there is a re-INVITE/INFO(package P) crossover?
> I think we need to say that if the initiating UA receives the INFO
> for package P, it should either:
> 1. still process it as normal
> or
> 2. issue a 4xx response (something like 491 Request Pending)
>   so that the package P event can be re-attempted once the ACK
>   has been received.
>
> Regards,
>
> Attila
>
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore




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<HTML dir=3Dltr><HEAD><TITLE>Re: [sipcore] info-03 draft and "session negot=
iation exchange"</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dunicode">
<META content=3D"MSHTML 6.00.6000.16825" name=3DGENERATOR></HEAD>
<BODY>
<DIV id=3DidOWAReplyText24741 dir=3Dltr>
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 size=3D2>&gt;&gt;Section =
3 is where "negotiation" occurs.&nbsp; I put negotiation in quotes <BR>&gt;=
&gt;because it is an advertising model, not an offer/answer model.</FONT></=
DIV>
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 size=3D2>That sentence al=
one, for me, makes it much clearer.</FONT></DIV>
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 size=3D2>I think the conf=
usion comes from:<BR>1. the word negotiation<BR>2. the statement:<BR>&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp; the target UA MUST NOT send any package P INFO met=
hods until the<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; target UA sees P in the fi=
nal ACK from the initiating UA.</FONT></DIV><FONT face=3DArial color=3D#000=
000 size=3D2>
<DIV dir=3Dltr><BR>Right now, I'd like the following to be considered:<BR>1=
. call it a "capability exchange" not a "negotiation" or "renegotiation".<B=
R>2. have a statement at the end of the first paragraph in section 3 - some=
thing like this:<BR>&nbsp;&nbsp; Note well: this is an advertising model - =
it is NOT a negotiation or offer/answer model.<BR>3. define what set of mes=
sages constitutes an exchange (this I'm still not clear about - especially =
where PRACK is involved?&nbsp; can a PRACK response be considered in the sa=
me way as the ACK?&nbsp;)<BR>4. also can we remove the statement<BR>&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; The target UA MUST NOT send any package P INFO meth=
ods until the<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; target UA sees P in the fin=
al ACK from the initiating UA.<BR>&nbsp;&nbsp; There seems contradiction be=
tween saying it's an advertising model but making the target UA wait for th=
e ACK.<BR>&nbsp;&nbsp; Making the target UA wait implies that it needs to s=
ee what the initiator says again, which seems more like negotiation behavio=
r.<BR>&nbsp;&nbsp;&nbsp; Why not just allow event P anytime after the initi=
ator has declared it in the Recv-Info of the re-INVITE/INVITE?<BR>5.&nbsp; =
If we must keep 4, then let's define what should happen if P occurs before =
the ACK (which could happen<BR>&nbsp;&nbsp;&nbsp; if the package P INFO is =
sent at the same time as the re-INVITE)</DIV>
<DIV dir=3Dltr><BR>Regards,<BR>Attila<BR></DIV>
<DIV dir=3Dltr>&nbsp;</DIV></FONT>
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></DIV>
<DIV dir=3Dltr><BR>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Eric Burger [mailto:eburger@stand=
ardstrack.com]<BR><B>Sent:</B> Fri 01/05/2009 14:12<BR><B>To:</B> Attila Si=
pos<BR><B>Cc:</B> SIPCORE<BR><B>Subject:</B> Re: [sipcore] info-03 draft an=
d "session negotiation exchange"<BR></FONT><BR></DIV>
<DIV>
<P><FONT size=3D2>Section 3 is where "negotiation" occurs.&nbsp; I put nego=
tiation in quotes&nbsp;<BR>because it is an advertising model, not an offer=
/answer model.<BR><BR>Would changing negotiation language to advertising la=
nguage reduce&nbsp;<BR>confusion, or is it semantic dressing?<BR><BR>On Apr=
 30, 2009, at 2:30 AM, Attila Sipos wrote:<BR><BR>&gt; Hi,<BR>&gt;<BR>&gt; =
Sorry for questions late in the day but I'm not clear on the<BR>&gt; negoti=
ation methods of the info package draft.<BR>&gt;<BR>&gt; The draft mentions=
 "session negotiation exchange" several times.<BR>&gt;<BR>&gt; Can someone =
tell me where this is defined?<BR>&gt;<BR>&gt; Section 3.1 (UA Behavior) sh=
ows INVITE/OK/ACK as an exchange.<BR>&gt;<BR>&gt; I see from section "5.2.&=
nbsp; INFO Headers" of the "INFO tables" e-mail<BR>&gt; that PRACK and UPDA=
TE are allowed to carry the Recv-Info header. So<BR>&gt; what is the "excha=
nge" if say, INVITE and PRACK are involved?<BR>&gt;<BR>&gt; From section 3.=
1, it seems that the initial "offer" is in the INVITE<BR>&gt; and then "con=
firmed" in the ACK. So if PRACK is received, can one<BR>&gt; "confirm" in a=
 PRACK?<BR>&gt;<BR>&gt; Also, if Recv-Info is allowed in UPDATE, how does t=
hat get&nbsp;<BR>&gt; "confirmed"?<BR>&gt; Or is it that UPDATE just lists =
the packages already negotiated?<BR>&gt;<BR>&gt; Or is it that package rene=
gotiation is only ever done in<BR>&gt; INVITE/re-INVITE?<BR>&gt; If so, I w=
ould like to see some text explicitly stating this.<BR>&gt; If not, what re=
negotiation combinations are there?<BR>&gt;<BR>&gt; Then there is this...<B=
R>&gt;&nbsp;&nbsp; The limitation on requiring the negotiation to occur wit=
hin the<BR>&gt;&nbsp;&nbsp; context of a session negotiation exchange means=
 that if the<BR>&gt;&nbsp;&nbsp; initiating UA issues a re-INVITE (after th=
e above ACK) with the<BR>&gt;&nbsp;&nbsp; following.<BR>&gt;<BR>&gt;&nbsp;&=
nbsp; INVITE ...<BR>&gt;&nbsp;&nbsp; ...<BR>&gt;&nbsp;&nbsp; Recv-Info: P, =
R, T<BR>&gt;&nbsp;&nbsp; ...<BR>&gt;<BR>&gt;&nbsp;&nbsp; The target UA MUST=
 NOT send any package P INFO methods until the<BR>&gt;&nbsp;&nbsp; target U=
A sees P in the final ACK from the initiating UA.<BR>&gt;<BR>&gt;<BR>&gt; S=
o, let's say P was being sent by the target.&nbsp;&nbsp; Then when the<BR>&=
gt; re-INVITE arrives, P has to be stopped (or queued), until P is<BR>&gt; =
seen again in the ACK.<BR>&gt;<BR>&gt; So what happens if there is a re-INV=
ITE/INFO(package P) crossover?<BR>&gt; I think we need to say that if the i=
nitiating UA receives the INFO<BR>&gt; for package P, it should either:<BR>=
&gt; 1. still process it as normal<BR>&gt; or<BR>&gt; 2. issue a 4xx respon=
se (something like 491 Request Pending)<BR>&gt;&nbsp;&nbsp; so that the pac=
kage P event can be re-attempted once the ACK<BR>&gt;&nbsp;&nbsp; has been =
received.<BR>&gt;<BR>&gt; Regards,<BR>&gt;<BR>&gt; Attila<BR>&gt;<BR>&gt;<B=
R>&gt;<BR>&gt;<BR>&gt; _______________________________________________<BR>&=
gt; sipcore mailing list<BR>&gt; sipcore@ietf.org<BR>&gt; <A href=3D"https:=
//www.ietf.org/mailman/listinfo/sipcore">https://www.ietf.org/mailman/listi=
nfo/sipcore</A><BR><BR></FONT></P></DIV><br><br>
<P align=3Dleft><FONT style=3D"BACKGROUND-COLOR: #ffffff">.</FONT></P>
</body></HTML>=

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Joel M. Halpern wrote:

[snip]

> And depending upon how everything was understood, and why it was being 
> redirected, the destination of the redirect may or may not care about 
> the original URI (may even care about the user part of the original URI) 
> in full, in order to decide what to do (the voicemail examples.)

I suspect that the voicemail example is a subtext to many of the 
arguments on this subject.

And that troubles me. The arguments I think I hear for and about this are:
- Alice calls Bob
- Bob forwards to Carol
- its forwarded again to Voicemail
- the desire is for it to be Bob's voicemail, not Carol's.
- And so *the* voicemail server needs to figure out whether
   to use Bob's voicemailbox or Carol's.

The trouble with this is that it presumes that Bob and Carol share a 
common VM server, or at least VM servers that collude with one another. 
While this is frequently the case, IMO it is a terrible assumption upon 
which to base a mechanism. For instance, Bob may have been called at 
work, which has its own VM system, and Carol may be Bob's wife with a VM 
system provided by her cellphone SP. In that case, even if Carol's VM 
server is able to discern that the call was intended for Bob, it won't 
have access to Bob's voicemailbox.

Solving this problem is really a matter of arranging when forwarding to 
Carol that the call will fail rather than go to Carol's VM. Then Bob's 
proxy will have the opportunity to forward the call to Bob's VM server. 
And in that case there won't be any complex issues about what mailbox it is.

So, what *other* problems are there where knowing the original R-URI is 
important?

	Thanks,
	Paul

From dean.willis@softarmor.com  Fri May  1 15:39:37 2009
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Subject: Re: [sipcore] Long reply to: Target URI and History Info bis
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On May 1, 2009, at 1:37 PM, Paul Kyzivat wrote:

>
> I suspect that the voicemail example is a subtext to many of the  
> arguments on this subject.
>
> And that troubles me. The arguments I think I hear for and about  
> this are:
> - Alice calls Bob
> - Bob forwards to Carol
> - its forwarded again to Voicemail
> - the desire is for it to be Bob's voicemail, not Carol's.
> - And so *the* voicemail server needs to figure out whether
>  to use Bob's voicemailbox or Carol's.
>
> The trouble with this is that it presumes that Bob and Carol share a  
> common VM server, or at least VM servers that collude with one  
> another. While this is frequently the case, IMO it is a terrible  
> assumption upon which to base a mechanism. For instance, Bob may  
> have been called at work, which has its own VM system, and Carol may  
> be Bob's wife with a VM system provided by her cellphone SP. In that  
> case, even if Carol's VM server is able to discern that the call was  
> intended for Bob, it won't have access to Bob's voicemailbox.

That's just broken. The easiest solution is for Carol's node to return  
a "no answer" to Bob's forwarding server, which then needs to route  
the call to Bob's VM.

The question is "How does Carol's node know whether to send a no- 
answer instead of sending the call to Carol's VM?"

The only answer that works for me is that Bob's forwarding server  
needs to tag the request in some way that gives Carol's node this  
information. I suspect that caller-preferences (sip.automata)=FALSE)   
gives us this capability, and that the solution here is proper  
configuration of Bob's forwarding server (and support for caller- 
preferences in Carol's node, which Bob's server can test with Require:).

Others might argue that an extension to Diversion is required that  
would allow detection of this state. I'd argue that this is bogus;  
Carol's node should not have to guess what is wanted based on what has  
already happened (an implicit preference). Instead, it should be told  
explicitly what outcome is desired (an explicit preference).

Another model is to add yet another extension that says "If you can't  
answer, please route this call to this alternate URI". Bob's  
forwarding server would then add this extension, attaching the URI for  
Bob's VM. Carol's node would then dispatch the request appropriately.  
This approach just seems a little fudgy to me, with lots of  
undesirable security properties.

--
Dean




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Subject: Re: [sipcore] Long reply to: Target URI and History Info bis
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Yes, I did have in mind that callerprefs can cause the right thing to 
happen if inserted properly and honored. (Big ifs, but alternatives that 
do the right thing are also big ifs.)

I would like to know if people have this case in mind when worrying 
about Diversion/H-I/Target URI issues.

	Thanks,
	Paul

P.S. Dean - you cut off my text that said pretty much what you added.

Dean Willis wrote:
> 
> On May 1, 2009, at 1:37 PM, Paul Kyzivat wrote:
> 
>>
>> I suspect that the voicemail example is a subtext to many of the 
>> arguments on this subject.
>>
>> And that troubles me. The arguments I think I hear for and about this 
>> are:
>> - Alice calls Bob
>> - Bob forwards to Carol
>> - its forwarded again to Voicemail
>> - the desire is for it to be Bob's voicemail, not Carol's.
>> - And so *the* voicemail server needs to figure out whether
>>  to use Bob's voicemailbox or Carol's.
>>
>> The trouble with this is that it presumes that Bob and Carol share a 
>> common VM server, or at least VM servers that collude with one 
>> another. While this is frequently the case, IMO it is a terrible 
>> assumption upon which to base a mechanism. For instance, Bob may have 
>> been called at work, which has its own VM system, and Carol may be 
>> Bob's wife with a VM system provided by her cellphone SP. In that 
>> case, even if Carol's VM server is able to discern that the call was 
>> intended for Bob, it won't have access to Bob's voicemailbox.
> 
> That's just broken. The easiest solution is for Carol's node to return a 
> "no answer" to Bob's forwarding server, which then needs to route the 
> call to Bob's VM.
> 
> The question is "How does Carol's node know whether to send a no-answer 
> instead of sending the call to Carol's VM?"
> 
> The only answer that works for me is that Bob's forwarding server needs 
> to tag the request in some way that gives Carol's node this information. 
> I suspect that caller-preferences (sip.automata)=FALSE)  gives us this 
> capability, and that the solution here is proper configuration of Bob's 
> forwarding server (and support for caller-preferences in Carol's node, 
> which Bob's server can test with Require:).
> 
> Others might argue that an extension to Diversion is required that would 
> allow detection of this state. I'd argue that this is bogus; Carol's 
> node should not have to guess what is wanted based on what has already 
> happened (an implicit preference). Instead, it should be told explicitly 
> what outcome is desired (an explicit preference).
> 
> Another model is to add yet another extension that says "If you can't 
> answer, please route this call to this alternate URI". Bob's forwarding 
> server would then add this extension, attaching the URI for Bob's VM. 
> Carol's node would then dispatch the request appropriately. This 
> approach just seems a little fudgy to me, with lots of undesirable 
> security properties.
> 
> -- 
> Dean
> 
> 
> 
> 

From christer.holmberg@ericsson.com  Sun May  3 11:00:08 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Dean Willis" <dean.willis@softarmor.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
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Hi,

I think the why-is-this-needed questions should have been asked when we
decided to start working on this in the first place. And, again, the
different drafts that have been produced on this topic DO describe
different use-cases.

Regarding the solution, again, we had a long discussion about it, argued
about different alternatives, and finally decided to go for a H-I based
solution.

I am sure there are other ways to do this than using H-I, but everybody
had his/her chance to participate in the beauty contest on which
solution to choose, so at this point I don't think we should revise our
decission - unless it can be shown that the H-I based solution doesn't
work.=20

Regards,

Christer

=20

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
Behalf Of Dean Willis
Sent: Friday, May 01, 2009 4:31 AM
To: Hans Erik van Elburg
Cc: SIPCORE
Subject: Re: [sipcore] Long reply to: Target URI and History Info bis

Hans Erik van Elburg wrote:
>=20
>=20
> Dean Willis wrote:
>>
>> On Apr 30, 2009, at 4:38 AM, Hans Erik van Elburg wrote:
>>
>>> Overloading in general is not a good design strategy, why do you=20
>>> think that trying to mold everything (diversion/history info) into=20
>>> Request URI is a good idea??
>>
>> The reason that we're having this conversation is that we have=20
>> proposals, History-Info being one, for 1) putting our parameters in=20
>> to the original request-URI, and 2) recovering that original=20
>> request-URI after it has been contact-routed (and otherwise mangled)=20
>> so that the parameters can be used at the UAS.
>>
>> So the presumption exists that the original request URI is useful to=20
>> the UAS. SIP did, and now SIPCORE has, a chartered item for making=20
>> that happen.
>>
>> I'm trying to back up and make sure that we know WHY we're trying to=20
>> deliver the original request URI to the UAS, and then asking if some=20
>> other approach that meets the same requirements might actually be=20
>> easier to do.
>>
> That is fine of course, but the point is that we are not talking about

> delivering parameters to an application, but about recovering a=20
> complete URI that has gone lost on the way.

but why exactly do we need to recover the URI? What is it that is
special about the URI or is likely to get used in some useful sort of
way downstream? If it isn't the "Request URI-ness" that is important,
maybe we can find some more convenient way to deliver that information.


>>> The problem here is that this information has nothing to do with=20
>>> application invocation, but tells something about how the signalling

>>> reached the callee (target). And this information is only relevant=20
>>> to the callee if that decides that it is.
>>
>>
>> That's a profound philosophical assertion. HTTP, for example, makes=20
>> no consideration for "how the signaling reaches the target" in=20
>> determining how to process a request. The fact that some think SIP=20
>> should take this into consideration indicates to me that we have a=20
>> layering problem in our design.
>>
> I don't think SIP and HTTP are comparable at all in this respect. In=20
> HTTP you directly address the server that you like to get (post,=20
> delete,
> update) information from. In SIP you adress the URI of a user, that=20
> will lead you to the home proxy in most cases and which is then=20
> forwarded to the UAS by overwriting the Request URI.

Well, HTTP has proxies too, and SIP were originally intended to be
comparable.

'> No layering problem, just a flaw in SIP's design that causes usefull
> information to be destroyed for no good reason. We're just trying to=20
> fix that, no more no less.
>> From one perspective, this makes the application invocation implicit;

>> the UAS has to guess what to do based on clues sprinkled throughout=20
>> the communication and across several layers of the ISO stack. It's=20
>> much better to have this sort of determination made based on explicit

>> indicators at a single application layer.
>>
> No. The goal of the target-uri discussion is to have all the relevant=20
> target-uri information readily available in the SIP request received=20
> by the UAS. That is all on the same OSI layer, layer 7.
>=20

I disgaree -- the goal is to make the information that was encoded into
the requestURI (and maybe as modified by intermediates, that use case is
open) available to the UAS.


--
dean
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From christer.holmberg@ericsson.com  Sun May  3 11:24:38 2009
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Thread-Topic: [sipcore] Question on Require in responses
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Dean Willis" <dean.willis@softarmor.com>, "Hans Erik van Elburg" <ietf.hanserik@gmail.com>
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Hi,=20

>>The option tag is a means of 1) discovering whether or not the far-=20
>>end supports a specific extension and of 2) declaring the criticality=20
>>of understanding that extension relative to satisfying a request and=20
>>3) informing the UAC of what extensions were applied to the request=20
>>in order to generate the response. This is purely a function of=20
>>capability negotiation, not one of invocation.
>>
>>The extension itself (typically, an extension header field, although=20
>>there are other types of extensions) is what causes the extension to=20
>>be invoked.
>Depends a bit how you define extension, but I would argue that if I=20
>require an extension to be supported on a transaction/dialog by the=20
>UAS and the UAS accepts the request that this means that for this call=20
>the UAS behaves as it should according to the extensions=20
>specification.
>
>Saying that an extensions invokes itself is meaningless to me.
>
>A UAS may choose to only invoke the extension behaviour if it finds=20
>that the extension is required. That comes close to saying that the=20
>require:extension invokes the behaviour in the UAS, but it is not at=20
>all the same.
>
>
>The Require doesn't control whether the UAS invokes the extension. The
UAS may invoke the extension on its own recognizance.  Most likely, it
chooses to do so based on 1) the presence of a Supported=20
>indicator for the extension in a request as evidence that the UAC
supports the extension or the presence of an extension header in the
request that also indicates support in the UAC and 2) support for=20
>that extension in the UAS. Other extensions may not require support in
the UAC, so may be enacted by this UAS purely on local policy.

100rel does not carry any extension headers in the request, nor does
RFC3262 say that the UAC must insert Supported in addition to Require.

But, RFC3262 DOES say:

"The UAS MUST send any non-100 provisional response reliably if the
initial request contained a Require header field with the option tag
100rel."

So, in my opinion Require:100rel means:

1. The UAC supports 100rel (no Suppored needed)
2. The UAS MUST use the extension (if sending non-100 provisional
responses).

I have nothing against saying that the definition of each option-tag
should specify how it is used, and what Require really means as far as
the UAS is concerned, but I don't think the generic definition of
Supported and Require that you are proposing fits all cases.

Regards,

Christer


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On May 3, 2009, at 1:00 PM, Christer Holmberg wrote:

>
> Hi,
>
> I think the why-is-this-needed questions should have been asked when  
> we
> decided to start working on this in the first place. And, again, the
> different drafts that have been produced on this topic DO describe
> different use-cases.

I asked it then. I  asked them as part of the process that lead to us  
chartering a deliverable of solving the problem. The reason I ask it  
again is that we seem to have forgotten the goal of the exercise, and  
be caught up in a solution without remembering the problem we were  
trying to solve.

SIP work has a history of doing this; it is a large part of why we  
have a fragmented and difficult to manage family of protocol  
specifications.

It bears asking again before we make things even worse.

Do we understand the implications of always delivering  multiple H-I  
items, and do we understand how a UAS can figure out which H-I item or  
items represents the "correct" parameter set?

Because if we do NOT understand these things, then we have failed to  
meet our actual requirements. I'm not saying H-I doesn't meet the  
requirements; I'm asking the WG to 1) understand the requirements, and  
2) verify that our proposed solution REALLY meets them.

Otherwise, we run the risk of producing yet another pointless  
mechanism that does not solve a problem that we failed to understand.


--
Dean

From christer.holmberg@ericsson.com  Mon May  4 01:05:33 2009
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=20
Hi,

I've submitted the -00 version of draft-ietf-sipcore-keep.

It can also be found at:

http://users.piuha.net/cholmber/drafts/draft-ietf-sipcore-keep-00.txt

I have not done any changes based on Shaun's comments. Any changes will
be in the next revision.

Regards,

Christer




> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> Sent: 1. toukokuuta 2009 0:05
> To: SIPCORE;=20
> draft-niemi-sipping-event-throttle@tools.ietf.org;=20
> draft-holmberg-sipcore-keep@tools.ietf.org
> Subject: [sipcore] Document Adoption
>=20
> [as chair]
>=20
> This message announces the formal adoption of two additional=20
> documents as SIPCORE working group items to satisfy two of=20
> our chartered milestones.
>=20
> Two weeks ago, the document=20
> draft-niemi-sipping-event-throttle, which was accepted by the=20
> RAI community at the San Franciso SIPPING meeting, was=20
> proposed by the chairs as a candidate to satisfy the=20
> milestone "SIP Events throttling mechanism"[1]. There was one=20
> affirmation of this decision. There was one objection, which=20
> was withdrawn after the question under consideration was=20
> clarified[2]. Given the "previously adopted" status of this=20
> document, and the lack of any remaining objections, the=20
> chairs now consider this a working group document.
>=20
> We also had eight positive responses and no objections to=20
> adopting draft-holmberg-sipcore-keep as a working group item=20
> [3] to satisfy the milestone "Mechanism for indicating=20
> support for keep-alives".
>=20
> Authors: please submit a draft-ietf-sipcore version of your=20
> document as soon as practical. Also, please keep in mind that=20
> change control now resides with the working group -- you are=20
> advised to make no substantive changes without discussion on=20
> the working group mailing list.
>=20
> Thanks!
>=20
> /a
>=20
> --
> [1] http://www.ietf.org/mail-archive/web/sipcore/current/msg00005.html
> [2] http://www.ietf.org/mail-archive/web/sipcore/current/msg00019.html
> [3] http://www.ietf.org/mail-archive/web/sipcore/current/msg00021.html
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

From christer.holmberg@ericsson.com  Mon May  4 03:09:13 2009
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Hi Shaun,=20

>> When a UAS that supports the method defined in this specification
>> recieves an initial SIP request which contains a "keep" parameter in
>> the topmost Via header, and the UAS wants to allow the sender of the
>> initial SIP request to send keep-alives towards the UAS during the
>> duration of the call, the UAS proxy MUST add a "yes" value to the
>> "keep" parameter.  In addition the UAS MAY include a Flow-Timer
>> header field if the associated initial SIP response.
>
>Which response is this text referring to when this is for a call?
>Does this include the 100 trying? If in the 100 trying, does the keep
need to be in all responses to the INVITE? WHat about the Flow-Timer?
>
>The reason I'm asking is that 100 trying is often generated at a very
different place than the other responses. It may or may not be feasible
to provide the relevant information in that response.
>I'm trying to understand the intent...

The keep-alives are always sent between two entities communicating
directly with each other (we don't expect SIP proxies for forward STUN
requests etc).

Also, the parameter is inserted in the Via of the request, so it will be
copied into whatever response is generated based on the request.

Regards,

Christer




-----Original Message-----
From: sip-bounces@ietf.org on behalf of Christer Holmberg
Sent: Thu 4/16/2009 4:31 AM
To: sipcore@ietf.org; sip@ietf.org
Cc: Adam Roach; Gonzalo Camarillo; Mary Barnes
Subject: [Sip] Draft new version: draft-holmberg-sipcore-keep-00
(previouslyknown as draft-holmberg-sip-keep)
=20

Hi,

I've submitted a draft-holmberg-sipcore version of the keep draft.

The draft is identical to the previous version (for which Mary sent out
the ask-for-support e-mail), but I have added one more example.

The draft can also be found at:

http://users.piuha.net/cholmber/drafts/draft-holmberg-sipcore-keep-00.tx
t


Regards,

Christer

Ps. I applogize for sending this e-mail to both SIP and SIPCORE, but
since everyone may not have subscribed to SIPCORE yet...






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Hi,

4244 in general is kinda messy, and one of the purposes of the proposed
4244bis work is to try to make it more clear.

But, as far as the target delivery work is concerned, in my opinion the
"goal" is clear: to be able to provide the original target to the UAS
(again, the draft explains some cases where it can be useful to be able
to do so).

Regards,

Christer


> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: 4. toukokuuta 2009 8:55
> To: Christer Holmberg
> Cc: Hans Erik van Elburg; SIPCORE
> Subject: Re: [sipcore] Long reply to: Target URI and History Info bis
>=20
>=20
> On May 3, 2009, at 1:00 PM, Christer Holmberg wrote:
>=20
> >
> > Hi,
> >
> > I think the why-is-this-needed questions should have been=20
> asked when=20
> > we decided to start working on this in the first place. And, again,=20
> > the different drafts that have been produced on this topic=20
> DO describe=20
> > different use-cases.
>=20
> I asked it then. I  asked them as part of the process that=20
> lead to us chartering a deliverable of solving the problem.=20
> The reason I ask it again is that we seem to have forgotten=20
> the goal of the exercise, and be caught up in a solution=20
> without remembering the problem we were trying to solve.
>=20
> SIP work has a history of doing this; it is a large part of=20
> why we have a fragmented and difficult to manage family of=20
> protocol specifications.
>=20
> It bears asking again before we make things even worse.
>=20
> Do we understand the implications of always delivering =20
> multiple H-I items, and do we understand how a UAS can figure=20
> out which H-I item or items represents the "correct" parameter set?
>=20
> Because if we do NOT understand these things, then we have=20
> failed to meet our actual requirements. I'm not saying H-I=20
> doesn't meet the requirements; I'm asking the WG to 1)=20
> understand the requirements, and
> 2) verify that our proposed solution REALLY meets them.
>=20
> Otherwise, we run the risk of producing yet another pointless=20
> mechanism that does not solve a problem that we failed to understand.
>=20
>=20
> --
> Dean
>=20

From SBharrat@sonusnet.com  Mon May  4 05:47:10 2009
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Thread-Topic: [Sip] Draft new version: draft-holmberg-sipcore-keep-00 (previouslyknown as draft-holmberg-sip-keep)
Thread-Index: Acm+bcAszQkyqV++Ts+HPG29TfGFgALlQSopAIXeZxAAJri/0A==
From: "Bharrat, Shaun" <SBharrat@sonusnet.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, <sipcore@ietf.org>
Subject: Re: [sipcore] [Sip] Draft new version: draft-holmberg-sipcore-keep-00 (previouslyknown as draft-holmberg-sip-keep)
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Christer:

> Also, the parameter is inserted in the Via of the request, so=20
> it will be copied into whatever response is generated based=20
> on the request.

I understand that the parameter itself will be copied back. What I
am wondering is whether it needs to be set to "yes" in every=20
response or can this work if it is set to "yes" in the final response
but not necessarily in the 100 or provisionals.

Support of this functionality is not necessarily global (within
a box). There are architectures where it will not be possible to know
whether support is enabled for that peer at the time or place that=20
the 100 is sent. If no one else envisions a problem along those lines,=20
then I guess my concern is moot. =20

Cheers,
Shaun


> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: Monday, May 04, 2009 6:09 AM
> To: Bharrat, Shaun; sipcore@ietf.org
> Subject: RE: [Sip] Draft new version:=20
> draft-holmberg-sipcore-keep-00 (previouslyknown as=20
> draft-holmberg-sip-keep)
>=20
>=20
> Hi Shaun,=20
>=20
> >> When a UAS that supports the method defined in this specification=20
> >> recieves an initial SIP request which contains a "keep"=20
> parameter in=20
> >> the topmost Via header, and the UAS wants to allow the=20
> sender of the=20
> >> initial SIP request to send keep-alives towards the UAS during the=20
> >> duration of the call, the UAS proxy MUST add a "yes" value to the=20
> >> "keep" parameter.  In addition the UAS MAY include a Flow-Timer=20
> >> header field if the associated initial SIP response.
> >
> >Which response is this text referring to when this is for a call?
> >Does this include the 100 trying? If in the 100 trying, does the keep
> need to be in all responses to the INVITE? WHat about the Flow-Timer?
> >
> >The reason I'm asking is that 100 trying is often generated at a very
> different place than the other responses. It may or may not=20
> be feasible to provide the relevant information in that response.
> >I'm trying to understand the intent...
>=20
> The keep-alives are always sent between two entities=20
> communicating directly with each other (we don't expect SIP=20
> proxies for forward STUN requests etc).
>=20
> Also, the parameter is inserted in the Via of the request, so=20
> it will be copied into whatever response is generated based=20
> on the request.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: sip-bounces@ietf.org on behalf of Christer Holmberg
> Sent: Thu 4/16/2009 4:31 AM
> To: sipcore@ietf.org; sip@ietf.org
> Cc: Adam Roach; Gonzalo Camarillo; Mary Barnes
> Subject: [Sip] Draft new version:=20
> draft-holmberg-sipcore-keep-00 (previouslyknown as=20
> draft-holmberg-sip-keep)
> =20
>=20
> Hi,
>=20
> I've submitted a draft-holmberg-sipcore version of the keep draft.
>=20
> The draft is identical to the previous version (for which=20
> Mary sent out the ask-for-support e-mail), but I have added=20
> one more example.
>=20
> The draft can also be found at:
>=20
> http://users.piuha.net/cholmber/drafts/draft-holmberg-sipcore-
> keep-00.tx
> t
>=20
>=20
> Regards,
>=20
> Christer
>=20
> Ps. I applogize for sending this e-mail to both SIP and=20
> SIPCORE, but since everyone may not have subscribed to SIPCORE yet...
>=20
>=20
>=20
>=20
>=20
>=20

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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Question on Require in responses
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Christer,

I haven't gone back to 3262 to verify that the words are consistent with 
what you say, but I trust you and my memory on that.

IMO it is not a good thing that there is such variation of meaning of 
Supported and Require. It would be much better if it was consistent 
across all options, in the way that Dean has been saying. But I guess 
that horse is out of the barn. Maybe we can try to enforce a consistent 
meaning across all *new* uses, but the old ones will have to remain as 
they are, which makes it more difficult to create a generic mechanism in 
an implementation.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi, 
> 
>>> The option tag is a means of 1) discovering whether or not the far- 
>>> end supports a specific extension and of 2) declaring the criticality 
>>> of understanding that extension relative to satisfying a request and 
>>> 3) informing the UAC of what extensions were applied to the request 
>>> in order to generate the response. This is purely a function of 
>>> capability negotiation, not one of invocation.
>>>
>>> The extension itself (typically, an extension header field, although 
>>> there are other types of extensions) is what causes the extension to 
>>> be invoked.
>> Depends a bit how you define extension, but I would argue that if I 
>> require an extension to be supported on a transaction/dialog by the 
>> UAS and the UAS accepts the request that this means that for this call 
>> the UAS behaves as it should according to the extensions 
>> specification.
>>
>> Saying that an extensions invokes itself is meaningless to me.
>>
>> A UAS may choose to only invoke the extension behaviour if it finds 
>> that the extension is required. That comes close to saying that the 
>> require:extension invokes the behaviour in the UAS, but it is not at 
>> all the same.
>>
>>
>> The Require doesn't control whether the UAS invokes the extension. The
> UAS may invoke the extension on its own recognizance.  Most likely, it
> chooses to do so based on 1) the presence of a Supported 
>> indicator for the extension in a request as evidence that the UAC
> supports the extension or the presence of an extension header in the
> request that also indicates support in the UAC and 2) support for 
>> that extension in the UAS. Other extensions may not require support in
> the UAC, so may be enacted by this UAS purely on local policy.
> 
> 100rel does not carry any extension headers in the request, nor does
> RFC3262 say that the UAC must insert Supported in addition to Require.
> 
> But, RFC3262 DOES say:
> 
> "The UAS MUST send any non-100 provisional response reliably if the
> initial request contained a Require header field with the option tag
> 100rel."
> 
> So, in my opinion Require:100rel means:
> 
> 1. The UAC supports 100rel (no Suppored needed)
> 2. The UAS MUST use the extension (if sending non-100 provisional
> responses).
> 
> I have nothing against saying that the definition of each option-tag
> should specify how it is used, and what Require really means as far as
> the UAS is concerned, but I don't think the generic definition of
> Supported and Require that you are proposing fits all cases.
> 
> Regards,
> 
> Christer
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From hannu.hietalahti@nokia.com  Mon May  4 06:21:19 2009
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From: <hannu.hietalahti@nokia.com>
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Date: Mon, 4 May 2009 15:21:46 +0200
Thread-Topic: Summary of issues around registering feature tags
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Subject: Re: [sipcore] Summary of issues around registering feature tags
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Thanks Cullen,

this solves some of the action points from the previous 3GPP - IETF telco (=
the good news)

Unfortunately it leaves more work to do (the bad news)

It seems best to close the AP's in 3GPP - IETF meeting later today, as this=
 task will need to be carried out by someone who is willing to contribute a=
nd to drive the work in IETF.

BR,
Hannu Hietalahti
3GPP TSG CT Chairman
tel: +358 40 5021724
=20

>-----Original Message-----
>From: ext Cullen Jennings [mailto:fluffy@cisco.com]=20
>Sent: 29 April, 2009 20:13
>To: sipcore@ietf.org
>Cc: Robert Sparks; Gonzalo Camarillo; Hietalahti Hannu=20
>(Nokia-CIC/Oulu); Adam Roach; Mary Barnes
>Subject: Summary of issues around registering feature tags
>
>
>3GPP has requested that IANA register two new feature tags
>(g.3gpp.icsi-ref and g.3gpp.iari-ref) in the global tree. Per
>RFC 2506 (Section 3.1.2), the policy for registrations in the
>global tree is expert review.
>
>This is the description of the feature tags to be registered:
>
>"Each value of the Application Reference media feature-tag=20
>indicates the
>software applications supported by the agent. The values for this tag
>equal the IMS communication Service Identifier (ICSI) and IMS
>Application Reference Identifier (IARI) values supported by
>the agent. The Application Reference media feature tag is
>defined to fulfill the requirements for forking to an appropriate UE =20
>when
>multiple UEs are registered and dispatch to an appropriate application
>within the UE based upon the IMS communication Service Identifier
>(ICSI) and IMS Application Reference Identifier (IARI) values as
>stated in 3GPP TS 23.228.  Multiple tag-values can be included in the
>Application Reference media feature-tag."
>
>Section 12.1 of RFC 3840 states that feature tags that are=20
>applicable to
>SIP and whose meaning is only defined within that usage are to be
>registered in the SIP tree. Since these feature tags are to be=20
>used only
>within the context of SIP, they would need to be registered in the SIP
>tree and not in the global tree. The policy for registrations=20
>in the SIP
>tree is IETF Consensus. That is, registrations need to be documented in
>an RFC approved by the IESG.
>
>Regardless of the tree these feature tags would need to be registered
>in, the expert reviewer indicated that this tags will not=20
>contain simple
>features but rather more complex services or feature sets. Therefore,
>the expert reviewer did not find it appropriate to register=20
>them because
>Section 2.1 of RFC 2506 states that "media feature tags represent
>individual and simple characteristics" and that "each media feature tag
>identifies a single characteristic".
>
>Additionally, Section 6 of
>draft-ietf-sipping-service-identification-03.txt describes the=20
>perils of
>performing declarative service identifications. The way these feature
>tags are described make them look as if they were going to be used to
>perform such declarative service identifications.
>
>Consequently, 3GPP is encouraged to submit an Internet Draft describing
>the use of these feature tags so that they can be registered in the SIP
>tree. Such Internet Draft needs to address the issues discussed in
>draft-ietf-sipping-service-identification-03.txt clarifying why these
>tags will not be used to perform declarative service identifications.
>
>Cullen <RAI AD>
>
>Thank you to the people that helped prepare this summary.
>
>=

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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 Core Working Group of the IETF.


	Title           : Indication of support for keep-alive
	Author(s)       : C. Holmberg
	Filename        : draft-ietf-sipcore-keep-00.txt
	Pages           : 11
	Date            : 2009-05-04

This specification defines a new SIP Via header parameter, "keep"
which SIP entities can use to indicate support of the NAT keep-alive
techniques defined in SIP Outbound, in cases where the Outbound
procedures are not supported or cannot be applied.

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

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From christer.holmberg@ericsson.com  Mon May  4 13:10:04 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Hi,

I don't think there is an issue with Supported. Supported:foo means "I
support foo" - nothing more, nothing less.

When it comes to Require:foo, it of course means "I require you to
support foo".

BUT, whether Require:foo also means "I require you to always use foo",
"I require you to use foo in specific situations", or "I require you to
support foo, but you don't have to use it" I think depends very much on
the extension, and choosing one of the above meanings for all future
extensions could be very restrictive.

In addition, there also seems to be different opinions on whehter
Require:foo also indicates that the sender supports it or not, or only
requires the receiver to support it. In most cases I think it can be
assumed that the sender also supports the extension, because the sender
would somehow be involved in the procedures associated with the
extension. But, there COULD of course be an extension which is only
affects the receiver, and does not require any kind of support by the
sender.

And, I think we have more or less the same issues/questions with
Proxy-Require as we have for Require.

Regards,

Christer


=20

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Monday, May 04, 2009 3:53 PM
To: Christer Holmberg
Cc: Dean Willis; Hans Erik van Elburg; SIPCORE
Subject: Re: [sipcore] Question on Require in responses

Christer,

I haven't gone back to 3262 to verify that the words are consistent with
what you say, but I trust you and my memory on that.

IMO it is not a good thing that there is such variation of meaning of
Supported and Require. It would be much better if it was consistent
across all options, in the way that Dean has been saying. But I guess
that horse is out of the barn. Maybe we can try to enforce a consistent
meaning across all *new* uses, but the old ones will have to remain as
they are, which makes it more difficult to create a generic mechanism in
an implementation.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi,
>=20
>>> The option tag is a means of 1) discovering whether or not the far-=20
>>> end supports a specific extension and of 2) declaring the=20
>>> criticality of understanding that extension relative to satisfying a

>>> request and
>>> 3) informing the UAC of what extensions were applied to the request=20
>>> in order to generate the response. This is purely a function of=20
>>> capability negotiation, not one of invocation.
>>>
>>> The extension itself (typically, an extension header field, although

>>> there are other types of extensions) is what causes the extension to

>>> be invoked.
>> Depends a bit how you define extension, but I would argue that if I=20
>> require an extension to be supported on a transaction/dialog by the=20
>> UAS and the UAS accepts the request that this means that for this=20
>> call the UAS behaves as it should according to the extensions=20
>> specification.
>>
>> Saying that an extensions invokes itself is meaningless to me.
>>
>> A UAS may choose to only invoke the extension behaviour if it finds=20
>> that the extension is required. That comes close to saying that the=20
>> require:extension invokes the behaviour in the UAS, but it is not at=20
>> all the same.
>>
>>
>> The Require doesn't control whether the UAS invokes the extension.=20
>> The
> UAS may invoke the extension on its own recognizance.  Most likely, it

> chooses to do so based on 1) the presence of a Supported
>> indicator for the extension in a request as evidence that the UAC
> supports the extension or the presence of an extension header in the=20
> request that also indicates support in the UAC and 2) support for
>> that extension in the UAS. Other extensions may not require support=20
>> in
> the UAC, so may be enacted by this UAS purely on local policy.
>=20
> 100rel does not carry any extension headers in the request, nor does
> RFC3262 say that the UAC must insert Supported in addition to Require.
>=20
> But, RFC3262 DOES say:
>=20
> "The UAS MUST send any non-100 provisional response reliably if the=20
> initial request contained a Require header field with the option tag=20
> 100rel."
>=20
> So, in my opinion Require:100rel means:
>=20
> 1. The UAC supports 100rel (no Suppored needed) 2. The UAS MUST use=20
> the extension (if sending non-100 provisional responses).
>=20
> I have nothing against saying that the definition of each option-tag=20
> should specify how it is used, and what Require really means as far as

> the UAS is concerned, but I don't think the generic definition of=20
> Supported and Require that you are proposing fits all cases.
>=20
> Regards,
>=20
> Christer
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

From christer.holmberg@ericsson.com  Mon May  4 13:17:34 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Bharrat, Shaun" <SBharrat@sonusnet.com>, <sipcore@ietf.org>
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Hi,=20

>>Also, the parameter is inserted in the Via of the request, so it will
be copied into whatever response is generated based on the request.
>
>I understand that the parameter itself will be copied back. What I am
wondering is whether it needs to be set to "yes" in every response or
can this work if it is set to "yes" in the final response but=20
>not necessarily in the 100 or provisionals.
>
>Support of this functionality is not necessarily global (within a box).
There are architectures where it will not be possible to know whether
support is enabled for that peer at the time or place that=20
>the 100 is sent. If no one else envisions a problem along those lines,
then I guess my concern is moot. =20

Since 100 responses are hop-to-hop, they are always generated by the
UAS, so I guess the UAS could choose not to insert a "yes" value in the
received Via when creating the response. We don't use 100 for other
types of negotiations either, so I see no harm in not inserting "yes" in
the 100 response, but still insert it in the forwarded request so it
will be present in all other responses, generated by the remote
endpoint.

Regards,

Christer




> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Monday, May 04, 2009 6:09 AM
> To: Bharrat, Shaun; sipcore@ietf.org
> Subject: RE: [Sip] Draft new version:=20
> draft-holmberg-sipcore-keep-00 (previouslyknown as
> draft-holmberg-sip-keep)
>=20
>=20
> Hi Shaun,
>=20
> >> When a UAS that supports the method defined in this specification=20
> >> recieves an initial SIP request which contains a "keep"
> parameter in
> >> the topmost Via header, and the UAS wants to allow the
> sender of the
> >> initial SIP request to send keep-alives towards the UAS during the=20
> >> duration of the call, the UAS proxy MUST add a "yes" value to the=20
> >> "keep" parameter.  In addition the UAS MAY include a Flow-Timer=20
> >> header field if the associated initial SIP response.
> >
> >Which response is this text referring to when this is for a call?
> >Does this include the 100 trying? If in the 100 trying, does the keep
> need to be in all responses to the INVITE? WHat about the Flow-Timer?
> >
> >The reason I'm asking is that 100 trying is often generated at a very
> different place than the other responses. It may or may not be=20
> feasible to provide the relevant information in that response.
> >I'm trying to understand the intent...
>=20
> The keep-alives are always sent between two entities communicating=20
> directly with each other (we don't expect SIP proxies for forward STUN

> requests etc).
>=20
> Also, the parameter is inserted in the Via of the request, so it will=20
> be copied into whatever response is generated based on the request.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: sip-bounces@ietf.org on behalf of Christer Holmberg
> Sent: Thu 4/16/2009 4:31 AM
> To: sipcore@ietf.org; sip@ietf.org
> Cc: Adam Roach; Gonzalo Camarillo; Mary Barnes
> Subject: [Sip] Draft new version:=20
> draft-holmberg-sipcore-keep-00 (previouslyknown as
> draft-holmberg-sip-keep)
> =20
>=20
> Hi,
>=20
> I've submitted a draft-holmberg-sipcore version of the keep draft.
>=20
> The draft is identical to the previous version (for which Mary sent=20
> out the ask-for-support e-mail), but I have added one more example.
>=20
> The draft can also be found at:
>=20
> http://users.piuha.net/cholmber/drafts/draft-holmberg-sipcore-
> keep-00.tx
> t
>=20
>=20
> Regards,
>=20
> Christer
>=20
> Ps. I applogize for sending this e-mail to both SIP and SIPCORE, but=20
> since everyone may not have subscribed to SIPCORE yet...
>=20
>=20
>=20
>=20
>=20
>=20

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Subject: Re: [sipcore] Question on Require in responses
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On Fri, 2009-05-01 at 08:10 +0100, Elwell, John wrote:
> Even if there are no absolute guarantees, there is a general expectation
> that things will work if used in the way described in the RFC where the
> option tag is defined. For example, if receipt of a replaces option tag
> does not mean that the Replaces header field is likely to be supported
> if received in future request from outside the dialog, things like
> attended call transfer would not work with any degree of reliability.
> 
> So I think the scope of an option tag is defined by the RFC where the
> option tag is defined.

This is an exceedingly good description of the state of practice.  But
the RFCs aren't written particularly well with this concept in mind.
Consider "eventlist".  You describe its scope quite cleanly:

        - eventlist - use in [Supported in] a SUBSCRIBE request
        indicates support for this feature in received NOTIFY requests
        on dialogs created from that SUBSCRIBE request;

But the text of the RFC doesn't seem to *say* that.  As far as I can
tell (without reading all of it), (normatively) the RFC says that it
creates the "eventlist" tag, and then says (explicitly non-normatively):

      Any client that supports the event list extension will include an
      option tag of "eventlist" in a "Supported" header field of every
      SUBSCRIBE message for a subscription for which it is willing to
      process a list.  If the subscription is made to a URI that
      represents a list, the RLS will include "eventlist" in a "Require"
      header field of the response to the SUBSCRIBE, and in all NOTIFY
      messages within that subscription.

The first sentence is close to what you have written, but as you note,
that fact needs to be stated normatively.  The second sentence talks
about "Require" in a response, which is far as I can tell, has no
defined meaning.

We've not been at all good about defining the scope of meaning of option
tags, and beyond that, we haven't been careful to regularize and
standardize the scopes, so that each usage doesn't need custom code to
be handled.

Dale



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On Mon, 2009-05-04 at 22:11 +0200, Christer Holmberg wrote:
> In addition, there also seems to be different opinions on whehter
> Require:foo also indicates that the sender supports it or not, or only
> requires the receiver to support it. In most cases I think it can be
> assumed that the sender also supports the extension, because the sender
> would somehow be involved in the procedures associated with the
> extension. But, there COULD of course be an extension which is only
> affects the receiver, and does not require any kind of support by the
> sender.

I think it would be best, and also safe, to *not* have "Require: foo"
imply "Supported: foo".  Of course, in most cases, the first without the
second would be useless. as the response to the request would include
information that the UAC could not process usefully.  But there is no
reason that an extension need be symmetric between the UAC and UAS sides
of a transaction, or that in an ongoing dialog, that using an extension
for requests going in one direction requires using the extension for
requests going in the other direction.

Dale



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--NextPart

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


	Title           : Example call flows using Session Initiation Protocol (SIP) security mechanisms
	Author(s)       : C. Jennings, et al.
	Filename        : draft-ietf-sipcore-sec-flows-00.txt
	Pages           : 59
	Date            : 2009-05-04

This document shows example call flows demonstrating the use of
Transport Layer Security (TLS), and Secure/Multipurpose Internet Mail
Extensions (S/MIME) in Session Initiation Protocol (SIP).  It also
provides information that helps implementers build interoperable SIP
software.  To help facilitate interoperability testing, it includes
certificates used in the example call flows and processes to create
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From dean.willis@softarmor.com  Mon May  4 18:56:57 2009
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On May 4, 2009, at 3:11 PM, Christer Holmberg wrote:

>
> Hi,
>
> I don't think there is an issue with Supported. Supported:foo means "I
> support foo" - nothing more, nothing less.

I agree 100%.

>
>
> When it comes to Require:foo, it of course means "I require you to
> support foo".

Right so far.

>
>
> BUT, whether Require:foo also means "I require you to always use foo",
> "I require you to use foo in specific situations", or "I require you  
> to
> support foo, but you don't have to use it" I think depends very much  
> on
> the extension, and choosing one of the above meanings for all future
> extensions could be very restrictive.

How about the authorization question: Christer requires the UAS to  
support and use the Foo: extension, but Christer is not authorized to  
use the Foo: extension at this UAS. UAS would be willing to process  
the request for Christer without using Foo:, but SIP provides no  
mechanism for UAS to tell Christer not to use Foo:.

Consequently, we need that "very restrictive" model, or we need new  
feature-authorization-negotiation capabilities.


> In addition, there also seems to be different opinions on whehter
> Require:foo also indicates that the sender supports it or not, or only
> requires the receiver to support it. In most cases I think it can be
> assumed that the sender also supports the extension, because the  
> sender
> would somehow be involved in the procedures associated with the
> extension. But, there COULD of course be an extension which is only
> affects the receiver, and does not require any kind of support by the
> sender.

Bingo. That's why it should also be in the sender's uipported:, and  
any extension that triggers on the REQUIRE to indicate support in the  
UAC is just badly written.

And yes, we have erroneous RFCs. Just because we wrote it that way  
doesn't make it right -- either something has to change in RFC 3261,  
or something has to change in the RFCs that misuse Require.


> And, I think we have more or less the same issues/questions with
> Proxy-Require as we have for Require.

Perhaps not quite so much, as we have written so few proxy-extensions,  
but probably true.

--
Dean

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On May 4, 2009, at 5:02 PM, Dale Worley wrote:
>
> The first sentence is close to what you have written, but as you note,
> that fact needs to be stated normatively.  The second sentence talks
> about "Require" in a response, which is far as I can tell, has no
> defined meaning.



"Require" in a response has a defined meaning: It means that the  
referenced extension was applied in formulating the response.  Of  
course, we have erroneous RFCs (including some I edited, dang it!)  
that don't do this right.

To quote RFC 3261  8.2.4 Applying Extensions:
> Any extensions applied to a non-421 response MUST be listed in a  
> Require header field included in the response.


Note that the following sentence, also from that section, is under  
debate -- I think it is not justifiable and should allow informational- 
track RFCs.
> Of course, the server MUST NOT apply extensions not listed in the  
> Supported header field in the request. As a result of this, the  
> Require header field in a response will only ever contain option  
> tags defined in standards- track RFCs.


--
Dean


From dean.willis@softarmor.com  Mon May  4 19:06:42 2009
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Subject: Re: [sipcore] Question on Require in responses
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On May 4, 2009, at 8:57 PM, Dean Willis wrote:

>
> On May 4, 2009, at 3:11 PM, Christer Holmberg wrote:
>
>>
>> Hi,
>>
>> I don't think there is an issue with Supported. Supported:foo means  
>> "I
>> support foo" - nothing more, nothing less.
>
> I agree 100%.
>
>>
>>
>> When it comes to Require:foo, it of course means "I require you to
>> support foo".
>
> Right so far.
>
>>
>>
>> BUT, whether Require:foo also means "I require you to always use  
>> foo",
>> "I require you to use foo in specific situations", or "I require  
>> you to
>> support foo, but you don't have to use it" I think depends very  
>> much on
>> the extension, and choosing one of the above meanings for all future
>> extensions could be very restrictive.
>
> How about the authorization question: Christer requires the UAS to  
> support and use the Foo: extension, but Christer is not authorized  
> to use the Foo: extension at this UAS. UAS would be willing to  
> process the request for Christer without using Foo:, but SIP  
> provides no mechanism for UAS to tell Christer not to use Foo:.
>
> Consequently, we need that "very restrictive" model, or we need new  
> feature-authorization-negotiation capabilities.
>
>
>> In addition, there also seems to be different opinions on whehter
>> Require:foo also indicates that the sender supports it or not, or  
>> only
>> requires the receiver to support it. In most cases I think it can be
>> assumed that the sender also supports the extension, because the  
>> sender
>> would somehow be involved in the procedures associated with the
>> extension. But, there COULD of course be an extension which is only
>> affects the receiver, and does not require any kind of support by the
>> sender.
>
> Bingo. That's why it should also be in the sender's uipported:, and  
> any extension that triggers on the REQUIRE to indicate support in  
> the UAC is just badly written.
>

Oops - it should be in the senders "Supported". If I could just type,  
I would only look half as stupid. Okay, maybe 3/4.

> And yes, we have erroneous RFCs. Just because we wrote it that way  
> doesn't make it right -- either something has to change in RFC 3261,  
> or something has to change in the RFCs that misuse Require.
>
>
>> And, I think we have more or less the same issues/questions with
>> Proxy-Require as we have for Require.
>
> Perhaps not quite so much, as we have written so few proxy- 
> extensions, but probably true.
>
> --
> Dean
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


From pkyzivat@cisco.com  Mon May  4 19:37:30 2009
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Question on Require in responses
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Dean Willis wrote:
> 
> On May 4, 2009, at 3:11 PM, Christer Holmberg wrote:
> 
>>
>> Hi,
>>
>> I don't think there is an issue with Supported. Supported:foo means "I
>> support foo" - nothing more, nothing less.
> 
> I agree 100%.

What about those cases where a UAC and UAS must both "support" it for it 
to be used, but where the roles are asymmetric?

There are lots of those. For instance, eventlist. Do you only advertise 
support if you can *receive* an eventlist in a NOTIFY?

What do you say if you are capable of sending eventlist in a NOTIFY, but 
not of receiving it. (Perhaps because you never subscribe to anything.)

	Thanks,
	Paul

>> When it comes to Require:foo, it of course means "I require you to
>> support foo".
> 
> Right so far.
> 
>>
>>
>> BUT, whether Require:foo also means "I require you to always use foo",
>> "I require you to use foo in specific situations", or "I require you to
>> support foo, but you don't have to use it" I think depends very much on
>> the extension, and choosing one of the above meanings for all future
>> extensions could be very restrictive.
> 
> How about the authorization question: Christer requires the UAS to 
> support and use the Foo: extension, but Christer is not authorized to 
> use the Foo: extension at this UAS. UAS would be willing to process the 
> request for Christer without using Foo:, but SIP provides no mechanism 
> for UAS to tell Christer not to use Foo:.
> 
> Consequently, we need that "very restrictive" model, or we need new 
> feature-authorization-negotiation capabilities.
> 
> 
>> In addition, there also seems to be different opinions on whehter
>> Require:foo also indicates that the sender supports it or not, or only
>> requires the receiver to support it. In most cases I think it can be
>> assumed that the sender also supports the extension, because the sender
>> would somehow be involved in the procedures associated with the
>> extension. But, there COULD of course be an extension which is only
>> affects the receiver, and does not require any kind of support by the
>> sender.
> 
> Bingo. That's why it should also be in the sender's uipported:, and any 
> extension that triggers on the REQUIRE to indicate support in the UAC is 
> just badly written.
> 
> And yes, we have erroneous RFCs. Just because we wrote it that way 
> doesn't make it right -- either something has to change in RFC 3261, or 
> something has to change in the RFCs that misuse Require.
> 
> 
>> And, I think we have more or less the same issues/questions with
>> Proxy-Require as we have for Require.
> 
> Perhaps not quite so much, as we have written so few proxy-extensions, 
> but probably true.
> 
> -- 
> Dean
> 

From christer.holmberg@ericsson.com  Mon May  4 22:38:41 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Dale Worley" <dworley@nortel.com>
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Subject: Re: [sipcore] Question on Require in responses
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Hi,=20

>>In addition, there also seems to be different opinions on whehter=20
>>Require:foo also indicates that the sender supports it or=20
>>not, or only requires the receiver to support it. In most cases I
think=20
>>it can be assumed that the sender also supports the extension, because
the=20
>>sender would somehow be involved in the procedures associated with the

>>extension. But, there COULD of course be an extension which is only=20
>>affects the receiver, and does not require any kind of support by the=20
>>sender.
>=20
>I think it would be best, and also safe, to *not* have "Require: foo"
>imply "Supported: foo".  Of course, in most cases, the first=20
>without the second would be useless. as the response to the=20
>request would include information that the UAC could not=20
>process usefully.  But there is no reason that an extension=20
>need be symmetric between the UAC and UAS sides of a=20
>transaction, or that in an ongoing dialog, that using an=20
>extension for requests going in one direction requires using=20
>the extension for requests going in the other direction.

Currently I wonder if there is a single extension which requires the UAC
to send Supported:foo in addition to Require:foo to indicate that the
UAC also supports the extension. The reason is probably that most (all?)
extensions do require some kind of support from both the UAC and UAS.

Regards,

Christer


From christer.holmberg@ericsson.com  Mon May  4 23:09:10 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Dean Willis" <dean.willis@softarmor.com>
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Hi,=20

>>>In addition, there also seems to be different opinions on whehter=20
>>>Require:foo also indicates that the sender supports it or not, or=20
>>>only requires the receiver to support it. In most cases I think it=20
>>>can be assumed that the sender also supports the extension, because=20
>>>the sender would somehow be involved in the procedures associated=20
>>>with the extension. But, there COULD of course be an extension which=20
>>>is only affects the receiver, and does not require any kind of=20
>>>support by the sender.
>>
>>Bingo. That's why it should also be in the sender's uipported:, and=20
>>any extension that triggers on the REQUIRE to indicate support in the=20
>>UAC is just badly written.
>>=20
>Oops - it should be in the senders "Supported". If I could=20
>just type, I would only look half as stupid. Okay, maybe 3/4.

Doesn't that mean that most extensions are badly written?
=20
>>And yes, we have erroneous RFCs. Just because we wrote it that way=20
>>doesn't make it right -- either something has to change in RFC 3261,=20
>>or something has to change in the RFCs that misuse Require.

Some extensions are already widely deployed, so I don't think we can do
very much about those.=20

Regards,

Christer


From ya-ching.tan@nsn.com  Tue May  5 04:26:39 2009
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From: "Tan, Ya Ching (NSN - DE/Munich)" <ya-ching.tan@nsn.com>
To: "ext Christer Holmberg" <christer.holmberg@ericsson.com>
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=20
Some comments :

1. Please update all references of [I-D.ietf-behave-rfc3489bis] to RFC
5389.

2. It is stated under Section 6 (Server behavior) that "The UAS MUST
also process double-CRLF keep-alives", but there is no mention of the
requirement to support the CRLF keep alive technic under Section 5
(Client behavior).

3. The draft repeatedly uses the term 'initial SIP request'.  What is
the definition of this as it is not a known technical term in SIP [RFC
3261].  The sentence in Section 5 "If the UAC wishes to apply keep-alive
for future calls, it MUST insert a "keep" Via parameter in the initial
SIP request of those calls".  This seems to imply that 'initial SIP
request' means the initial INVITE.  'Initial INVITE' is what RFC 3261
calls the dialog-initiating INVITE.  Should the "keep" parameter in Via
only be used in initial REGISTER and initial INVITE ?

4. What is the definition of 'initial SIP response' used throughout the
draft ?  I take it that the "keep=3Dyes" parameter is only meaningful =
and
useful if the response is a 2xx.  Is that right ?

5. The Via headers in the REGISTER/INVITE sent by the Proxy P1 are in
the wrong order.  "Via:P1" should be the topmost Via header.

6. Section 5 : "... and the UAC wants to send keep-alives towards the
next hop, the entity MUST include a "keep" Via parameter in the SIP
request".  It should be "... the entity MUST include a "keep" parameter
in the Via header it inserts into the SIP request".

7. Section 5 : "If the Via header header in the initial SIP responses
contains a "keep" parameter with a "yes" value..." should read "If the
top Via header in the 2xx SIP response contains a "keep" parameter with
a "yes" value...".

8. Third paragraph of Section 5 : "If the initial SIP response contains
a Flow-Timer header...".  In the SIP outbound draft, the Flow-Timer
header is only meant to be present in 'a successful registration
response' (i.e. in a REGISTER response).

9. Typos in Section 9.1 and 9.2 : "recommented" -> "recommended".


Regards,
Ya-Ching

From christer.holmberg@ericsson.com  Tue May  5 05:02:16 2009
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Hi,=20

>1. Please update all references of [I-D.ietf-behave-rfc3489bis] to RFC
5389.

Will do.


>2. It is stated under Section 6 (Server behavior) that "The=20
>UAS MUST also process double-CRLF keep-alives", but there is=20
>no mention of the requirement to support the CRLF keep alive=20
>technic under Section 5 (Client behavior).

I'll look into that.


>3. The draft repeatedly uses the term 'initial SIP request'. =20
>What is the definition of this as it is not a known technical=20
>term in SIP [RFC 3261].  The sentence in Section 5 "If the=20
>UAC wishes to apply keep-alive for future calls, it MUST=20
>insert a "keep" Via parameter in the initial SIP request of=20
>those calls".  This seems to imply that 'initial SIP request'=20
>means the initial INVITE.  'Initial INVITE' is what RFC 3261=20
>calls the dialog-initiating INVITE.  Should the "keep"=20
>parameter in Via only be used in initial REGISTER and initial INVITE ?

I think that is the current assumption, yes, but I am not sure whether
we need to make that restriction.

Regarding the usage of "initial SIP request", I guess it should say
"initial SIP dialog request", or something like that. I think similar
wording is used in some places of the document.


>4. What is the definition of 'initial SIP response' used=20
>throughout the draft ?  I take it that the "keep=3Dyes"=20
>parameter is only meaningful and useful if the response is a=20
>2xx.  Is that right ?

I guess it should be "response to the initial SIP request", or something
like that.

In addition to 2xx, the keep=3Dyes can also be useful in provisional
responses.

=20
>5. The Via headers in the REGISTER/INVITE sent by the Proxy=20
>P1 are in the wrong order.  "Via:P1" should be the topmost Via header.

True. I noted that already earlier, but forgot to fix it.


>6. Section 5 : "... and the UAC wants to send keep-alives=20
>towards the next hop, the entity MUST include a "keep" Via=20
>parameter in the SIP request".  It should be "... the entity=20
>MUST include a "keep" parameter in the Via header it inserts=20
>into the SIP request".

Correct.


>7. Section 5 : "If the Via header header in the initial SIP=20
>responses contains a "keep" parameter with a "yes" value..."=20
>should read "If the top Via header in the 2xx SIP response=20
>contains a "keep" parameter with a "yes" value...".

Correct.


>8. Third paragraph of Section 5 : "If the initial SIP=20
>response contains a Flow-Timer header...".  In the SIP=20
>outbound draft, the Flow-Timer header is only meant to be=20
>present in 'a successful registration response' (i.e. in a=20
>REGISTER response).

I was thinking about the same. But, does outbound actaully forbid
Flow-Timer to be used elsewhere than in REGISTER responses? I know that
the procedures only talk about registration responses, but that is not
the same thing.

I guess what is missing from outbound is the normal table which shows
for which methods and message types the header can be used.


>9. Typos in Section 9.1 and 9.2 : "recommented" -> "recommended".

I'll fix it.


Thank you very much for your comments!

Regards,

Christer


From christer.holmberg@ericsson.com  Tue May  5 05:10:25 2009
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Hi,

One issue which was raised by Ya-Ching, when giving comments on the
keep-draft, was whether it is allowed to use the Flow-Timer header in
other messages than REGISTER responses?

The keep draft also uses it in INVITEs when keep-alive is negotiated.

The procedures in Outbound naturally only mention the Flow-Timer header
in REGISTER responses, but it is nowever specifically indicated that the
header is applicable for REGISTER responses only.

What seems to be missing from Outbound is the table we normally use to
define for which methods and message types a header is applicable.

Example:

Header field          where   proxy ACK BYE CAN INV OPT REG
------------------------------------------------------------------------
--------------------

Flow-Timer             18x                -      -       -      o     -
-
Flow-Timer             2xx                -      -       -      o     -
o =20


Regards,

Christer

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<P><FONT SIZE=3D2 FACE=3D"Arial">Hi,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">One issue which was raised by Ya-Ching, =
when giving comments on the keep-draft, was whether it is allowed to use =
the Flow-Timer header in other messages than REGISTER =
responses?</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">The keep draft also uses it in INVITEs =
when keep-alive is negotiated.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">The procedures in Outbound naturally =
only mention the Flow-Timer header in REGISTER responses, but it is =
nowever specifically indicated that the header is applicable for =
REGISTER responses only.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">What seems to be missing from Outbound =
is the table we normally use to define for which methods and message =
types a header is applicable.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Example:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Header =
field&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
where&nbsp;&nbsp; proxy ACK BYE CAN INV OPT REG</FONT>

<BR><FONT SIZE=3D2 =
FACE=3D"Arial">----------------------------------------------------------=
----------------------------------</FONT>
</P>

<P><FONT SIZE=3D2 =
FACE=3D"Arial">Flow-Timer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp; =
18x&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
o&nbsp;&nbsp;&nbsp;&nbsp; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; -</FONT>

<BR><FONT SIZE=3D2 =
FACE=3D"Arial">Flow-Timer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp; =
2xx&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
o&nbsp;&nbsp;&nbsp;&nbsp; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; o&nbsp; =
</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">Regards,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Christer</FONT>
</P>

</BODY>
</HTML>
------_=_NextPart_001_01C9CD7A.A95C5AA9--

From john.elwell@siemens-enterprise.com  Tue May  5 06:52:09 2009
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Date: Tue, 05 May 2009 14:53:13 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: 05 May 2009 13:04
> To: Tan, Ya Ching (NSN - DE/Munich)
> Cc: SIPCORE
> Subject: Re: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>=20
>=20
> >3. The draft repeatedly uses the term 'initial SIP request'. =20
> >What is the definition of this as it is not a known technical=20
> >term in SIP [RFC 3261].  The sentence in Section 5 "If the=20
> >UAC wishes to apply keep-alive for future calls, it MUST=20
> >insert a "keep" Via parameter in the initial SIP request of=20
> >those calls".  This seems to imply that 'initial SIP request'=20
> >means the initial INVITE.  'Initial INVITE' is what RFC 3261=20
> >calls the dialog-initiating INVITE.  Should the "keep"=20
> >parameter in Via only be used in initial REGISTER and=20
> initial INVITE ?
>=20
> I think that is the current assumption, yes, but I am not sure whether
> we need to make that restriction.
>=20
> Regarding the usage of "initial SIP request", I guess it should say
> "initial SIP dialog request", or something like that. I think similar
> wording is used in some places of the document.
[JRE] So the sender of an "initial SIP dialog request" would invoke the
capability between itself and the next downstream SIP entity. Presumably
a SIP proxy that does not record-route would  never invoke this
capability. But what about the next SIP entity downstream, if that
doesn't record-route? Presumably it too should refrain from engaging in
this capability. Mid-dialog requests would by-pass any
non-record-routing proxies, so there would be no opportunity at the time
of the "initial SIP dialog request" to create, and hence keep-alive, the
connections needed for mid-dialog requests. I couldn't find any
discussion of this in the latest draft.

John


From john.elwell@siemens-enterprise.com  Tue May  5 07:07:43 2009
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Date: Tue, 05 May 2009 15:08:26 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
In-reply-to: <9E895F3D-EE68-49E9-B262-034CFE035C15@softarmor.com>
To: Dean Willis <dean.willis@softarmor.com>, Christer Holmberg <christer.holmberg@ericsson.com>
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Dean,

I tend to agree with the use case that yourself and Paul articulated so
well. Why did I agree to the H-I solution a few months back? Well,
perhaps the use case was not properly articulated at the time, and it
was difficult to see the wood for the trees. I seem to recall some
freephone use case that some people wanted. I was quite happy with the
Rosenberg loose route solution as a means of providing the last target
AoR URI on the last hop, but when JDR stopped pursuing that approach, I
accepted that H-I could do that too. However, when we start talking
about H-I providing reliable information about earlier targets and
expecting special behaviour to take place at the UAS based on this
information (e.g., selecting a different mailbox) I get VERY nervous.

John


> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Dean Willis
> Sent: 04 May 2009 06:55
> To: Christer Holmberg
> Cc: SIPCORE
> Subject: Re: [sipcore] Long reply to: Target URI and History Info bis
>=20
>=20
> On May 3, 2009, at 1:00 PM, Christer Holmberg wrote:
>=20
> >
> > Hi,
> >
> > I think the why-is-this-needed questions should have been=20
> asked when =20
> > we
> > decided to start working on this in the first place. And, again, the
> > different drafts that have been produced on this topic DO describe
> > different use-cases.
>=20
> I asked it then. I  asked them as part of the process that=20
> lead to us =20
> chartering a deliverable of solving the problem. The reason I ask it =20
> again is that we seem to have forgotten the goal of the=20
> exercise, and =20
> be caught up in a solution without remembering the problem we were =20
> trying to solve.
>=20
> SIP work has a history of doing this; it is a large part of why we =20
> have a fragmented and difficult to manage family of protocol =20
> specifications.
>=20
> It bears asking again before we make things even worse.
>=20
> Do we understand the implications of always delivering  multiple H-I =20
> items, and do we understand how a UAS can figure out which=20
> H-I item or =20
> items represents the "correct" parameter set?
>=20
> Because if we do NOT understand these things, then we have failed to =20
> meet our actual requirements. I'm not saying H-I doesn't meet the =20
> requirements; I'm asking the WG to 1) understand the=20
> requirements, and =20
> 2) verify that our proposed solution REALLY meets them.
>=20
> Otherwise, we run the risk of producing yet another pointless =20
> mechanism that does not solve a problem that we failed to understand.
>=20
>=20
> --
> Dean
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

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Date: Tue, 05 May 2009 16:59:20 +0200
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Long reply to:  Target URI and History Info bis
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The following observation then must be comforting: There are no 
requirements to deliver "earlier targets". The target-uri work only 
requires the current target's uri information to be delivered.

If the consequences of using H-I has some unnerving side-effects, 
well... its a little late to realise that.
I am somewhat puzzled as to what proposal is being brought forward here.

/Hans Erik

Elwell, John wrote:
> Dean,
>
> I tend to agree with the use case that yourself and Paul articulated so
> well. Why did I agree to the H-I solution a few months back? Well,
> perhaps the use case was not properly articulated at the time, and it
> was difficult to see the wood for the trees. I seem to recall some
> freephone use case that some people wanted. I was quite happy with the
> Rosenberg loose route solution as a means of providing the last target
> AoR URI on the last hop, but when JDR stopped pursuing that approach, I
> accepted that H-I could do that too. However, when we start talking
> about H-I providing reliable information about earlier targets and
> expecting special behaviour to take place at the UAS based on this
> information (e.g., selecting a different mailbox) I get VERY nervous.
>
> John
>
>
>   
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org 
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Dean Willis
>> Sent: 04 May 2009 06:55
>> To: Christer Holmberg
>> Cc: SIPCORE
>> Subject: Re: [sipcore] Long reply to: Target URI and History Info bis
>>
>>
>> On May 3, 2009, at 1:00 PM, Christer Holmberg wrote:
>>
>>     
>>> Hi,
>>>
>>> I think the why-is-this-needed questions should have been 
>>>       
>> asked when  
>>     
>>> we
>>> decided to start working on this in the first place. And, again, the
>>> different drafts that have been produced on this topic DO describe
>>> different use-cases.
>>>       
>> I asked it then. I  asked them as part of the process that 
>> lead to us  
>> chartering a deliverable of solving the problem. The reason I ask it  
>> again is that we seem to have forgotten the goal of the 
>> exercise, and  
>> be caught up in a solution without remembering the problem we were  
>> trying to solve.
>>
>> SIP work has a history of doing this; it is a large part of why we  
>> have a fragmented and difficult to manage family of protocol  
>> specifications.
>>
>> It bears asking again before we make things even worse.
>>
>> Do we understand the implications of always delivering  multiple H-I  
>> items, and do we understand how a UAS can figure out which 
>> H-I item or  
>> items represents the "correct" parameter set?
>>
>> Because if we do NOT understand these things, then we have failed to  
>> meet our actual requirements. I'm not saying H-I doesn't meet the  
>> requirements; I'm asking the WG to 1) understand the 
>> requirements, and  
>> 2) verify that our proposed solution REALLY meets them.
>>
>> Otherwise, we run the risk of producing yet another pointless  
>> mechanism that does not solve a problem that we failed to understand.
>>
>>
>> --
>> Dean
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>>     
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>   

From dean.willis@softarmor.com  Tue May  5 09:37:23 2009
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Subject: Re: [sipcore] Question on Require in responses
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On May 4, 2009, at 9:38 PM, Paul Kyzivat wrote:

>
>
> Dean Willis wrote:
>> On May 4, 2009, at 3:11 PM, Christer Holmberg wrote:
>>>
>>> Hi,
>>>
>>> I don't think there is an issue with Supported. Supported:foo  
>>> means "I
>>> support foo" - nothing more, nothing less.
>> I agree 100%.
>
> What about those cases where a UAC and UAS must both "support" it  
> for it to be used, but where the roles are asymmetric?

What about them? Sending a "Supported" in a request or response means  
the UAC supports the extension. Sending a "Require" in a request means  
the UAS must support the extension. Whether or not the extension  
actually gets used is dependent on policy, authorization, and the  
extension itself.

>
>
> There are lots of those. For instance, eventlist. Do you only  
> advertise support if you can *receive* an eventlist in a NOTIFY?

Yes.

>
>
> What do you say if you are capable of sending eventlist in a NOTIFY,  
> but not of receiving it. (Perhaps because you never subscribe to  
> anything.)

Nothing (at least from the current spec, AFAIK).

If we need asymmetric negotiation, we need to define two different  
option tags, for example "event-list-receive" and "eventlist-send".

--
Dean


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On May 5, 2009, at 1:10 AM, Christer Holmberg wrote:
>>
>
> Doesn't that mean that most extensions are badly written?

Most of our documents could stand improvement. Perfection keeps  
eluding me. Please let me know when you feel you've achieved it, as I  
enjoy comedy.


>
>
>>> And yes, we have erroneous RFCs. Just because we wrote it that way
>>> doesn't make it right -- either something has to change in RFC 3261,
>>> or something has to change in the RFCs that misuse Require.
>
> Some extensions are already widely deployed, so I don't think we can  
> do
> very much about those.


Other than fixing the specification, there's not much we can do.

Backward compatibility can be assured through Postel's maxim.  
Specification fixes should explain how this applies to the  
specification in question.

--
Dean



From dean.willis@softarmor.com  Tue May  5 09:45:03 2009
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Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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On May 5, 2009, at 7:11 AM, Christer Holmberg wrote:

>
> Hi,
>
> One issue which was raised by Ya-Ching, when giving comments on the  
> keep-draft, was whether it is allowed to use the Flow-Timer header  
> in other messages than REGISTER responses?
>
> The keep draft also uses it in INVITEs when keep-alive is negotiated.
>
> The procedures in Outbound naturally only mention the Flow-Timer  
> header in REGISTER responses, but it is nowever specifically  
> indicated that the header is applicable for REGISTER responses only.
>
> What seems to be missing from Outbound is the table we normally use  
> to define for which methods and message types a header is applicable.
>
> Example:
>
> Header field          where   proxy ACK BYE CAN INV OPT REG
> --------------------------------------------------------------------------------------------
>
> Flow-Timer             18x                -      -       -       
> o     -       -
> Flow-Timer             2xx                -      -       -       
> o     -       o
>
That's probably because that table is the source of more confusion,  
IMHO, than anything else in the entire SIP specification family. It  
direly needs a retrofit, although I'm not sure that such is really  
possible: we add another column to the table for each method-type  
extension, such as SUB, NOT, INF, etc. This raises the question: If a  
new request type is specified, is it up to that request-type to fix  
"the table" for every header field, or do we need to revise the RC for  
each extension so as to add the new column to their tables?

Consequently, I greatly prefer to specify the behavior in text, using  
more general terms such as "requests" and "responses", which specific  
caveats around where such an extension header is meaningful, where it  
is forbidden, and where it is ignorable.

--
Dean


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Subject: Re: [sipcore] Long reply to:  Target URI and History Info bis
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On May 5, 2009, at 9:59 AM, Hans Erik van Elburg wrote:

> The following observation then must be comforting: There are no  
> requirements to deliver "earlier targets". The target-uri work only  
> requires the current target's uri information to be delivered.
>
> If the consequences of using H-I has some unnerving side-effects,  
> well... its a little late to realise that.
> I am somewhat puzzled as to what proposal is being brought forward  
> here.

It's not too late. We haven't died yet. We haven't even published an  
RFC, and that's far less final as we often revise RFCs by publishing  
new ones.

--
Dean


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Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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In outbound, the following should be added:

      Header field          where   proxy ACK BYE CAN INV OPT REG
      ___________________________________________________________
      Flow-Timer             2xx      a    -   -   -   -   -   o

In Keep, the following:=20

      Header field          where   proxy ACK BYE CAN INV OPT REG
      ___________________________________________________________
      Flow-Timer             18x      a    -   -   -   o   -   -
      Flow-Timer             2xx      a    -   -   -   o   -   o

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Dean Willis
> Sent: Tuesday, May 05, 2009 09:46
> To: Christer Holmberg
> Cc: SIPCORE
> Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER=20
> responses?
>=20
>=20
> On May 5, 2009, at 7:11 AM, Christer Holmberg wrote:
>=20
> >
> > Hi,
> >
> > One issue which was raised by Ya-Ching, when giving comments on the=20
> > keep-draft, was whether it is allowed to use the Flow-Timer=20
> header in=20
> > other messages than REGISTER responses?
> >
> > The keep draft also uses it in INVITEs when keep-alive is=20
> negotiated.
> >
> > The procedures in Outbound naturally only mention the Flow-Timer=20
> > header in REGISTER responses, but it is nowever=20
> specifically indicated=20
> > that the header is applicable for REGISTER responses only.
> >
> > What seems to be missing from Outbound is the table we=20
> normally use to=20
> > define for which methods and message types a header is applicable.
> >
> > Example:
> >
> > Header field          where   proxy ACK BYE CAN INV OPT REG
> >=20
> ----------------------------------------------------------------------
> > ----------------------
> >
> > Flow-Timer             18x                -      -       -      =20
> > o     -       -
> > Flow-Timer             2xx                -      -       -      =20
> > o     -       o
> >
> That's probably because that table is the source of more=20
> confusion, IMHO, than anything else in the entire SIP=20
> specification family. It direly needs a retrofit, although=20
> I'm not sure that such is really
> possible: we add another column to the table for each=20
> method-type extension, such as SUB, NOT, INF, etc. This=20
> raises the question: If a new request type is specified, is=20
> it up to that request-type to fix "the table" for every=20
> header field, or do we need to revise the RC for each=20
> extension so as to add the new column to their tables?
>=20
> Consequently, I greatly prefer to specify the behavior in=20
> text, using more general terms such as "requests" and=20
> "responses", which specific caveats around where such an=20
> extension header is meaningful, where it is forbidden, and=20
> where it is ignorable.
>=20
> --
> Dean
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Question on Require in responses
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Dean Willis wrote:
> 
> On May 4, 2009, at 9:38 PM, Paul Kyzivat wrote:
> 
>>
>>
>> Dean Willis wrote:
>>> On May 4, 2009, at 3:11 PM, Christer Holmberg wrote:
>>>>
>>>> Hi,
>>>>
>>>> I don't think there is an issue with Supported. Supported:foo means "I
>>>> support foo" - nothing more, nothing less.
>>> I agree 100%.
>>
>> What about those cases where a UAC and UAS must both "support" it for 
>> it to be used, but where the roles are asymmetric?
> 
> What about them? Sending a "Supported" in a request or response means 
> the UAC supports the extension.

Which role does it mean it supports? Assuming the asymmetric roles can 
be ascribed to the UAC and UAS, then perhaps a Supported:foo in a 
request means that the sender supports the UAC role for foo, and a 
Supported:foo in a response means that the sender supports the UAS role 
for foo?

But what if I want to know if you support the UAC role for foo? How can 
I find that out? I could send you an OPTIONS request, but then your 
answer would be telling me if you support the UAS role, not the UAC role.

> Sending a "Require" in a request means 
> the UAS must support the extension.

Again, does that mean it must support the UAS role. What if I send a 
Require as part of a REFER in expectation that the option will be used 
in the referred request? In that case it will be the UAC role that it 
must use.

> Whether or not the extension 
> actually gets used is dependent on policy, authorization, and the 
> extension itself.

I agree that is the way it should have been.

>> There are lots of those. For instance, eventlist. Do you only 
>> advertise support if you can *receive* an eventlist in a NOTIFY?
> 
> Yes.
> 
>> What do you say if you are capable of sending eventlist in a NOTIFY, 
>> but not of receiving it. (Perhaps because you never subscribe to 
>> anything.)
> 
> Nothing (at least from the current spec, AFAIK).
> 
> If we need asymmetric negotiation, we need to define two different 
> option tags, for example "event-list-receive" and "eventlist-send".

I think many of the options are asymmetric, so perhaps many of them 
really need pairs of options.

	Thanks,
	Paul

From dean.willis@softarmor.com  Tue May  5 11:21:42 2009
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Subject: Re: [sipcore] Question on Require in responses
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On May 5, 2009, at 12:29 PM, Paul Kyzivat wrote:

>
>
> Dean Willis wrote:
>> On May 4, 2009, at 9:38 PM, Paul Kyzivat wrote:
>>>
>>>
>>> Dean Willis wrote:
>>>> On May 4, 2009, at 3:11 PM, Christer Holmberg wrote:
>>>>>
>>>>> Hi,
>>>>>
>>>>> I don't think there is an issue with Supported. Supported:foo  
>>>>> means "I
>>>>> support foo" - nothing more, nothing less.
>>>> I agree 100%.
>>>
>>> What about those cases where a UAC and UAS must both "support" it  
>>> for it to be used, but where the roles are asymmetric?
>> What about them? Sending a "Supported" in a request or response  
>> means the UAC supports the extension.
>
> Which role does it mean it supports? Assuming the asymmetric roles  
> can be ascribed to the UAC and UAS, then perhaps a Supported:foo in  
> a request means that the sender supports the UAC role for foo, and a  
> Supported:foo in a response means that the sender supports the UAS  
> role for foo?

Whichever role the assigning RFC defines for the option tag. By  
default, this is a symmetric negotiation, unless the RFC says otherwise.

> But what if I want to know if you support the UAC role for foo? How  
> can I find that out? I could send you an OPTIONS request, but then  
> your answer would be telling me if you support the UAS role, not the  
> UAC role.

That depends on the RFC. Most if not all currently only define one  
role for negotiation.


>
>
>> Sending a "Require" in a request means the UAS must support the  
>> extension.
>
> Again, does that mean it must support the UAS role. What if I send a  
> Require as part of a REFER in expectation that the option will be  
> used in the referred request? In that case it will be the UAC role  
> that it must use.

Perhaps I should have said "the node that was acting as a UAS in the  
handling of the receipt of the request, even if it becomes a UAC in  
some future transaction wherein the extension applies".

But sending a Require in the body of a REFER says very little or  
nothing about whether or not the extension will be applied in the  
transaction induced by the REFER; in general, if you want that to  
apply, you have to encode it as part of the Refer-To URI, because  
that's what gets applied to the induced request.

Require in a body never really "induces" behavior. For example, you  
might Require me to support a feature that I support don't allow you  
to use. I don't fail the request with a 420 -- instead, I just ignore  
your Require.

>
>
>> Whether or not the extension actually gets used is dependent on  
>> policy, authorization, and the extension itself.
>
> I agree that is the way it should have been.

No, really it's that way. See the above example on not using an  
extension  . . . It's just the the discussion in some of the RFCs is  
misleading or downright wrong.

>
>
>>> There are lots of those. For instance, eventlist. Do you only  
>>> advertise support if you can *receive* an eventlist in a NOTIFY?
>> Yes.
>>> What do you say if you are capable of sending eventlist in a  
>>> NOTIFY, but not of receiving it. (Perhaps because you never  
>>> subscribe to anything.)
>> Nothing (at least from the current spec, AFAIK).
>> If we need asymmetric negotiation, we need to define two different  
>> option tags, for example "event-list-receive" and "eventlist-send".
>
> I think many of the options are asymmetric, so perhaps many of them  
> really need pairs of options.

I agree with that conclusion.

--
Dean Willis



From john.elwell@siemens-enterprise.com  Tue May  5 12:32:10 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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What about SUBSCRIBE and REFER requests and responses?

John=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Francois Audet
> Sent: 05 May 2009 17:54
> To: Dean Willis; Christer Holmberg
> Cc: SIPCORE
> Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER=20
> responses?
>=20
> In outbound, the following should be added:
>=20
>       Header field          where   proxy ACK BYE CAN INV OPT REG
>       ___________________________________________________________
>       Flow-Timer             2xx      a    -   -   -   -   -   o
>=20
> In Keep, the following:=20
>=20
>       Header field          where   proxy ACK BYE CAN INV OPT REG
>       ___________________________________________________________
>       Flow-Timer             18x      a    -   -   -   o   -   -
>       Flow-Timer             2xx      a    -   -   -   o   -   o
>=20
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org=20
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Dean Willis
> > Sent: Tuesday, May 05, 2009 09:46
> > To: Christer Holmberg
> > Cc: SIPCORE
> > Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER=20
> > responses?
> >=20
> >=20
> > On May 5, 2009, at 7:11 AM, Christer Holmberg wrote:
> >=20
> > >
> > > Hi,
> > >
> > > One issue which was raised by Ya-Ching, when giving=20
> comments on the=20
> > > keep-draft, was whether it is allowed to use the Flow-Timer=20
> > header in=20
> > > other messages than REGISTER responses?
> > >
> > > The keep draft also uses it in INVITEs when keep-alive is=20
> > negotiated.
> > >
> > > The procedures in Outbound naturally only mention the Flow-Timer=20
> > > header in REGISTER responses, but it is nowever=20
> > specifically indicated=20
> > > that the header is applicable for REGISTER responses only.
> > >
> > > What seems to be missing from Outbound is the table we=20
> > normally use to=20
> > > define for which methods and message types a header is applicable.
> > >
> > > Example:
> > >
> > > Header field          where   proxy ACK BYE CAN INV OPT REG
> > >=20
> >=20
> ----------------------------------------------------------------------
> > > ----------------------
> > >
> > > Flow-Timer             18x                -      -       -      =20
> > > o     -       -
> > > Flow-Timer             2xx                -      -       -      =20
> > > o     -       o
> > >
> > That's probably because that table is the source of more=20
> > confusion, IMHO, than anything else in the entire SIP=20
> > specification family. It direly needs a retrofit, although=20
> > I'm not sure that such is really
> > possible: we add another column to the table for each=20
> > method-type extension, such as SUB, NOT, INF, etc. This=20
> > raises the question: If a new request type is specified, is=20
> > it up to that request-type to fix "the table" for every=20
> > header field, or do we need to revise the RC for each=20
> > extension so as to add the new column to their tables?
> >=20
> > Consequently, I greatly prefer to specify the behavior in=20
> > text, using more general terms such as "requests" and=20
> > "responses", which specific caveats around where such an=20
> > extension header is meaningful, where it is forbidden, and=20
> > where it is ignorable.
> >=20
> > --
> > Dean
> >=20
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
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To: Dean Willis <dean.willis@softarmor.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Long reply to:  Target URI and History Info bis
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Dean,

Again, considering that the requirements in the target-uri draft is what 
we want to solve, what IS your proposed solution?

BR,
/Hans Erik

Dean Willis wrote:
>
> On May 5, 2009, at 9:59 AM, Hans Erik van Elburg wrote:
>
>> The following observation then must be comforting: There are no 
>> requirements to deliver "earlier targets". The target-uri work only 
>> requires the current target's uri information to be delivered.
>>
>> If the consequences of using H-I has some unnerving side-effects, 
>> well... its a little late to realise that.
>> I am somewhat puzzled as to what proposal is being brought forward here.
>
> It's not too late. We haven't died yet. We haven't even published an 
> RFC, and that's far less final as we often revise RFCs by publishing 
> new ones.
>
> -- 
> Dean
>

From christer.holmberg@ericsson.com  Tue May  5 13:41:47 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "Francois Audet" <audet@nortel.com>, "Dean Willis" <dean.willis@softarmor.com>
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Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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Hi,=20

>What about SUBSCRIBE and REFER requests and responses?

I guess the mechanism could be useful also for SUB and REF dialogs.

Regards,

Christer


> -----Original Message-----
> From: sipcore-bounces@ietf.org
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Francois Audet
> Sent: 05 May 2009 17:54
> To: Dean Willis; Christer Holmberg
> Cc: SIPCORE
> Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER=20
> responses?
>=20
> In outbound, the following should be added:
>=20
>       Header field          where   proxy ACK BYE CAN INV OPT REG
>       ___________________________________________________________
>       Flow-Timer             2xx      a    -   -   -   -   -   o
>=20
> In Keep, the following:=20
>=20
>       Header field          where   proxy ACK BYE CAN INV OPT REG
>       ___________________________________________________________
>       Flow-Timer             18x      a    -   -   -   o   -   -
>       Flow-Timer             2xx      a    -   -   -   o   -   o
>=20
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Dean Willis
> > Sent: Tuesday, May 05, 2009 09:46
> > To: Christer Holmberg
> > Cc: SIPCORE
> > Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER=20
> > responses?
> >=20
> >=20
> > On May 5, 2009, at 7:11 AM, Christer Holmberg wrote:
> >=20
> > >
> > > Hi,
> > >
> > > One issue which was raised by Ya-Ching, when giving
> comments on the
> > > keep-draft, was whether it is allowed to use the Flow-Timer
> > header in
> > > other messages than REGISTER responses?
> > >
> > > The keep draft also uses it in INVITEs when keep-alive is
> > negotiated.
> > >
> > > The procedures in Outbound naturally only mention the Flow-Timer=20
> > > header in REGISTER responses, but it is nowever
> > specifically indicated
> > > that the header is applicable for REGISTER responses only.
> > >
> > > What seems to be missing from Outbound is the table we
> > normally use to
> > > define for which methods and message types a header is applicable.
> > >
> > > Example:
> > >
> > > Header field          where   proxy ACK BYE CAN INV OPT REG
> > >=20
> >=20
> ----------------------------------------------------------------------
> > > ----------------------
> > >
> > > Flow-Timer             18x                -      -       -      =20
> > > o     -       -
> > > Flow-Timer             2xx                -      -       -      =20
> > > o     -       o
> > >
> > That's probably because that table is the source of more confusion,=20
> > IMHO, than anything else in the entire SIP specification family. It=20
> > direly needs a retrofit, although I'm not sure that such is really
> > possible: we add another column to the table for each method-type=20
> > extension, such as SUB, NOT, INF, etc. This raises the question: If=20
> > a new request type is specified, is it up to that request-type to=20
> > fix "the table" for every header field, or do we need to revise the=20
> > RC for each extension so as to add the new column to their tables?
> >=20
> > Consequently, I greatly prefer to specify the behavior in text,=20
> > using more general terms such as "requests" and "responses", which=20
> > specific caveats around where such an extension header is=20
> > meaningful, where it is forbidden, and where it is ignorable.
> >=20
> > --
> > Dean
> >=20
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

From christer.holmberg@ericsson.com  Tue May  5 13:43:38 2009
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Hi,=20

>>One issue which was raised by Ya-Ching, when giving comments on the=20
>>keep-draft, was whether it is allowed to use the Flow-Timer header in=20
>>other messages than REGISTER responses?
>>
>>The keep draft also uses it in INVITEs when keep-alive is negotiated.
>>
>>The procedures in Outbound naturally only mention the Flow-Timer=20
>>header in REGISTER responses, but it is nowever specifically indicated

>>that the header is applicable for REGISTER responses only.
>>
>>What seems to be missing from Outbound is the table we normally use to

>>define for which methods and message types a header is applicable.
>>
>>Example:
>>
>>Header field          where   proxy ACK BYE CAN INV OPT REG
>>----------------------------------------------------------------------
>>----------------------
>>
>>Flow-Timer             18x                -      -       -      =20
>>o     -       -
>>Flow-Timer             2xx                -      -       -      =20
>>o     -       o
>>
>That's probably because that table is the source of more confusion,
IMHO, than anything else in the entire SIP specification family. It
direly needs a retrofit, although I'm not sure that such is really
>possible: we add another column to the table for each method-type
extension, such as SUB, NOT, INF, etc. This raises the question: If a
new request type is specified, is it up to that request-type to fix
>"the table" for every header field, or do we need to revise the RC for
each extension so as to add the new column to their tables?
>
>Consequently, I greatly prefer to specify the behavior in text, using
more general terms such as "requests" and "responses", which specific
caveats around where such an extension header is meaningful,=20
>where it is forbidden, and where it is ignorable.

I think the table is clear.

Maybe we could have a table on some website. Then it would be easy to
update it every time new methods and/or headers are added.

Regards,

Christer


From christer.holmberg@ericsson.com  Tue May  5 13:48:21 2009
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Thread-Topic: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "Tan, Ya Ching (NSN - DE/Munich)" <ya-ching.tan@nsn.com>
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Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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Hi John,

You raise a good point. The draft should probably say that a SIP entity
MUST record-route (unless the entity is an endpoint) if it is going to
send and/or receive keep-alives.

Regards,

Christer

-----Original Message-----
From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
Sent: Tuesday, May 05, 2009 4:53 PM
To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
Cc: SIPCORE
Subject: RE: [sipcore] Draft submission: draft-ietf-sipcore-keep-00
(wasRE:Document Adoption)

=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: 05 May 2009 13:04
> To: Tan, Ya Ching (NSN - DE/Munich)
> Cc: SIPCORE
> Subject: Re: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>=20
>=20
> >3. The draft repeatedly uses the term 'initial SIP request'. =20
> >What is the definition of this as it is not a known technical term in

> >SIP [RFC 3261].  The sentence in Section 5 "If the UAC wishes to=20
> >apply keep-alive for future calls, it MUST insert a "keep" Via=20
> >parameter in the initial SIP request of those calls".  This seems to=20
> >imply that 'initial SIP request'
> >means the initial INVITE.  'Initial INVITE' is what RFC 3261 calls=20
> >the dialog-initiating INVITE.  Should the "keep"
> >parameter in Via only be used in initial REGISTER and
> initial INVITE ?
>=20
> I think that is the current assumption, yes, but I am not sure whether
> we need to make that restriction.
>=20
> Regarding the usage of "initial SIP request", I guess it should say
> "initial SIP dialog request", or something like that. I think similar
> wording is used in some places of the document.
[JRE] So the sender of an "initial SIP dialog request" would invoke the
capability between itself and the next downstream SIP entity. Presumably
a SIP proxy that does not record-route would  never invoke this
capability. But what about the next SIP entity downstream, if that
doesn't record-route? Presumably it too should refrain from engaging in
this capability. Mid-dialog requests would by-pass any
non-record-routing proxies, so there would be no opportunity at the time
of the "initial SIP dialog request" to create, and hence keep-alive, the
connections needed for mid-dialog requests. I couldn't find any
discussion of this in the latest draft.

John


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Dean Willis" <dean.willis@softarmor.com>
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Hi,=20

>>Doesn't that mean that most extensions are badly written?
>
>Most of our documents could stand improvement.=20

I couldn't agree more, but in this case you are talking about a rule
which may NEVER have been used for our specifications.

>Perfection keeps eluding me. Please let me know when you feel you've
achieved it, as I enjoy comedy.

There is no such thing as perfection in standards.

>>>And yes, we have erroneous RFCs. Just because we wrote it that way=20
>>>doesn't make it right -- either something has to change in RFC 3261,=20
>>>or something has to change in the RFCs that misuse Require.
>>
>>Some extensions are already widely deployed, so I don't think we can=20
>>do very much about those.
>
>Other than fixing the specification, there's not much we can do.
>
>Backward compatibility can be assured through Postel's maxim. =20
>Specification fixes should explain how this applies to the
specification in question.

Yes.

Regards,

Christer




From spencer@wonderhamster.org  Tue May  5 15:03:20 2009
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From: "Spencer Dawkins" <spencer@wonderhamster.org>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, "Dean Willis" <dean.willis@softarmor.com>
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Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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I'm kind of in the middle here...

> I think the table is clear.

I think that conceptually it is, or should be, clear. Having it split over a 
large number of documents today makes it less clear :-(

> Maybe we could have a table on some website. Then it would be easy to
> update it every time new methods and/or headers are added.

I think Dean's concern wasn't about the technology ("need more columns"), 
but about the review and approval process - if there's one table, and I 
create a new method or header, is it my job to fill in the entire table for 
the new row/column, or is it someone else's job (keeping in mind that I may 
not be the smartest person about everything that's in the table now - I 
might just be the smartest person about the method or header that I'm 
defining. If three of us define two rows and a column, how do we ensure that 
there's enough review for the new version of the table?

Stuff like that, I think. Or Dean can correct me.

Thanks,

Spencer 



From dean.willis@softarmor.com  Tue May  5 17:01:38 2009
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Subject: Re: [sipcore] Long reply to:  Target URI and History Info bis
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On May 5, 2009, at 3:20 PM, Hans Erik van Elburg wrote:

> Dean,
>
> Again, considering that the requirements in the target-uri draft is  
> what we want to solve, what IS your proposed solution?
>>

I'm not proposing one at this time. What I AM proposing is that we  
thoroughly analyze the proposal to make sure that it a) meets the  
given requirements (as stated in the draft), and b)  that the given  
requirements address both the chartered milestone requirements as  
written and the requirements as intended when we adopted that milestone.

Let's see if I can formulate what I think the problem with draft- 
rosenberg-sip-target-uri-delivery-01 is:

I'm suspecting that somewhere in the interaction between service- 
invocation and retargeting, we're losing the parameters.

Example A:

Let's say alice@example.com is calling bob@example.org. Alice wishes  
to include some parameters for Bob's UA application that are  
understood by all the user agents at example.org. Bob, however, has  
forwarded his calls to carol@example.org.

Alice's parameters go in the original request-URI.

1) Bob's proxy retargets to Carol, pushing the original request-URI  
into History-Info with an istarget parameter, because Identity in the  
reverse direction would now be expected to show Carol.

2) The request reaches Carol's proxy, which adds yet another history  
info (without istarget) containing Carol's contact-uri.

3) The request reaches Carol's phone, which works back the stack to  
retrieve the first "istarget", which now contains Carol's AOR and no  
parameters. It accepts the call, but it has no parameters for the  
application, so the application fails.  Also, since Carol is the  
"target" she thinks the call is for her, not for Bob (although  
ostensibly the To: still shows bob@example.org)


One might argue that Carol's UA can backtrack and know that the "first  
target" was Bob, and could potentially retrieve the parameters for  
Bob.  How would it know to do this?

And if it did, consider:

  Example B:

alice@example.com is calling bob@example.org. Alice wishes to include  
some parameters for Bob's UA application that are understood by all  
the user agents at example.org. Bob, however, has forwarded his calls  
to carol@example.org, and Bob's proxy is engaging "service invocation  
as per 4.5 of  raft-rosenberg-sip-target-uri-delivery-01. Carol,  
however, has also forwarded her calls to david@example.org

Alice's parameters go in the original request-URI.

1) Bob's proxy retargets to Carol, pushing the original request-URI  
into History-Info with an istarget parameter, because Identity in the  
reverse direction would now be expected to show Carol. I'm not sure  
whether Bob's proxy pushes the new service-invocation parameters into  
the "original" R-URI or not; I think it likely that it adds a second H- 
I "istarget" that has the modifed parameters.

2) The request reaches Carol's proxy, which adds yet another history  
info (with istarget) containing David's  AOR.

3) The request reaches David's proxy, which contact-routes to David's  
phone and adds  yet another H-I (without istarget) for  
david@davidscontact.

4) The request reaches David's phone, which works back the stack to  
retrieve the first "istarget", which now contains David's AOR and no  
parameters. It accepts the call, but it has no parameters for the  
application, so the application fails.   So which H-I should David's  
UA backtrack to to collect the parameters? It isn't the first one (bob@example.org 
), since Bob's proxy added some parameters. It's not the last one (carol@example.org 
) either. So what does it do?

--
Dean Willis



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Subject: Re: [sipcore] Question on Require in responses
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On May 5, 2009, at 4:26 PM, Christer Holmberg wrote:

>
> Hi,
>
>>> Doesn't that mean that most extensions are badly written?
>>
>> Most of our documents could stand improvement.
>
> I couldn't agree more, but in this case you are talking about a rule
> which may NEVER have been used for our specifications.

Yet it is written that way in RFC 3261.

So, where is the error? To achieve consistency, SOMETHING has to change.

--
Dean


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Spencer Dawkins" <spencer@wonderhamster.org>, "Dean Willis" <dean.willis@softarmor.com>
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Hi,=20

>>I'm kind of in the middle here...
>=20
>>I think the table is clear.
>=20
>I think that conceptually it is, or should be, clear. Having=20
>it split over a large number of documents today makes it less=20
>clear :-(
>
>=20
>>Maybe we could have a table on some website. Then it would be easy to=20
>>update it every time new methods and/or headers are added.
>=20
>I think Dean's concern wasn't about the technology ("need=20
>more columns"), but about the review and approval process -=20
>if there's one table, and I create a new method or header, is=20
>it my job to fill in the entire table for the new row/column,=20
>or is it someone else's job (keeping in mind that I may not=20
>be the smartest person about everything that's in the table=20
>now - I might just be the smartest person about the method or=20
>header that I'm defining. If three of us define two rows and=20
>a column, how do we ensure that there's enough review for the=20
>new version of the table?

Like everything else related to your new method/header, whatever is to
be added to the table is discussed and agreed upon among the interested
parties.

Regards,

Christer


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Dean Willis" <dean.willis@softarmor.com>
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>>>>Doesn't that mean that most extensions are badly written?
>>>
>>>Most of our documents could stand improvement.
>>
>>I couldn't agree more, but in this case you are talking about a rule=20
>>which may NEVER have been used for our specifications.
>=20
>Yet it is written that way in RFC 3261.
>=20
>So, where is the error? To achieve consistency, SOMETHING has=20
>to change.

Yes, and in this case LOTS of stuff would have to be changed. There
would be essential correction drafts to a number of specs :)

Regards,

Christer


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Subject: Re: [sipcore] Long reply to:  Target URI and History Info bis
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Analyzing requirements is always a good thing to do.

Note that the parameters are only part of the problem. The loss of the 
URI is the actual problem, if the URI contains parameters then they are 
lost together with the URI. That is why I do not agree with rephrasing 
this to a URI parameter loss problem. The whole URI is lost. The 
parameters only do carry their intended meaning within the context of 
this URI.

The problem I have with example A & B is that they assume that 
parameters which are intended for B, are also relevant for C or D. This 
requirement is not expressed in the target-uri draft, nor do i think 
that it is a valid requirement.

The service invocation use case explicitly considers: "use of complex 
URIs to address services within the network rather than subscribers. " 
This shows that the scenario that you are testing the solution against 
is not the intended use case.

Note that the solution currently outlined in the target-uri draft is not 
sufficient, as it does not  solve the freephone use case. So yes it 
still needs some work.

Best Regards,
/Hans Erik

Dean Willis wrote:
>
> On May 5, 2009, at 3:20 PM, Hans Erik van Elburg wrote:
>
>> Dean,
>>
>> Again, considering that the requirements in the target-uri draft is 
>> what we want to solve, what IS your proposed solution?
>>>
>
> I'm not proposing one at this time. What I AM proposing is that we 
> thoroughly analyze the proposal to make sure that it a) meets the 
> given requirements (as stated in the draft), and b)  that the given 
> requirements address both the chartered milestone requirements as 
> written and the requirements as intended when we adopted that milestone.
>
> Let's see if I can formulate what I think the problem with 
> draft-rosenberg-sip-target-uri-delivery-01 is:
>
> I'm suspecting that somewhere in the interaction between 
> service-invocation and retargeting, we're losing the parameters.
>
> Example A:
>
> Let's say alice@example.com is calling bob@example.org. Alice wishes 
> to include some parameters for Bob's UA application that are 
> understood by all the user agents at example.org. Bob, however, has 
> forwarded his calls to carol@example.org.
>
> Alice's parameters go in the original request-URI.
>
> 1) Bob's proxy retargets to Carol, pushing the original request-URI 
> into History-Info with an istarget parameter, because Identity in the 
> reverse direction would now be expected to show Carol.
>
> 2) The request reaches Carol's proxy, which adds yet another history 
> info (without istarget) containing Carol's contact-uri.
>
> 3) The request reaches Carol's phone, which works back the stack to 
> retrieve the first "istarget", which now contains Carol's AOR and no 
> parameters. It accepts the call, but it has no parameters for the 
> application, so the application fails.  Also, since Carol is the 
> "target" she thinks the call is for her, not for Bob (although 
> ostensibly the To: still shows bob@example.org)
>
>
> One might argue that Carol's UA can backtrack and know that the "first 
> target" was Bob, and could potentially retrieve the parameters for 
> Bob.  How would it know to do this?
>
> And if it did, consider:
>
>  Example B:
>
> alice@example.com is calling bob@example.org. Alice wishes to include 
> some parameters for Bob's UA application that are understood by all 
> the user agents at example.org. Bob, however, has forwarded his calls 
> to carol@example.org, and Bob's proxy is engaging "service invocation 
> as per 4.5 of  raft-rosenberg-sip-target-uri-delivery-01. Carol, 
> however, has also forwarded her calls to david@example.org
>
> Alice's parameters go in the original request-URI.
>
> 1) Bob's proxy retargets to Carol, pushing the original request-URI 
> into History-Info with an istarget parameter, because Identity in the 
> reverse direction would now be expected to show Carol. I'm not sure 
> whether Bob's proxy pushes the new service-invocation parameters into 
> the "original" R-URI or not; I think it likely that it adds a second 
> H-I "istarget" that has the modifed parameters.
>
> 2) The request reaches Carol's proxy, which adds yet another history 
> info (with istarget) containing David's  AOR.
>
> 3) The request reaches David's proxy, which contact-routes to David's 
> phone and adds  yet another H-I (without istarget) for 
> david@davidscontact.
>
> 4) The request reaches David's phone, which works back the stack to 
> retrieve the first "istarget", which now contains David's AOR and no 
> parameters. It accepts the call, but it has no parameters for the 
> application, so the application fails.   So which H-I should David's 
> UA backtrack to to collect the parameters? It isn't the first one 
> (bob@example.org), since Bob's proxy added some parameters. It's not 
> the last one (carol@example.org) either. So what does it do?
>
> -- 
> Dean Willis
>
>

From john.elwell@siemens-enterprise.com  Tue May  5 23:16:51 2009
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Date: Wed, 06 May 2009 07:17:58 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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Christer,

That seems somewhat back-to-front. You don't record route (and hence
determine the path of the dialog) just because you want to do
keep-alives. You first decide whether you are to be on the path of a
dialog, and then you should decide whether to do keep-alives with your
neighbour. I guess where non-record-routing proxies are involved, this
can be achieved only on the first mid-dialog request. However, for an
INVITE-initiated dialog this might be the ACK request, but unfortunately
that doesn't have a response, so it is not suitable. In some cases it
would be an UPDATE or PRACK request during the early dialog. For a
SUBSCRIBE- or REFER-initiated dialog it would be the first NOTIFY
request.

In extending the keep-alive principle to dialogs, we need to be very
careful how we specify it.

John

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 05 May 2009 21:50
> To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> Cc: SIPCORE
> Subject: RE: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>=20
>=20
> Hi John,
>=20
> You raise a good point. The draft should probably say that a=20
> SIP entity
> MUST record-route (unless the entity is an endpoint) if it is going to
> send and/or receive keep-alives.
>=20
> Regards,
>=20
> Christer
>=20
> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
> Sent: Tuesday, May 05, 2009 4:53 PM
> To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
> Cc: SIPCORE
> Subject: RE: [sipcore] Draft submission: draft-ietf-sipcore-keep-00
> (wasRE:Document Adoption)
>=20
> =20
>=20
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> > Sent: 05 May 2009 13:04
> > To: Tan, Ya Ching (NSN - DE/Munich)
> > Cc: SIPCORE
> > Subject: Re: [sipcore] Draft submission:=20
> > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> >=20
> >=20
> > >3. The draft repeatedly uses the term 'initial SIP request'. =20
> > >What is the definition of this as it is not a known=20
> technical term in
>=20
> > >SIP [RFC 3261].  The sentence in Section 5 "If the UAC wishes to=20
> > >apply keep-alive for future calls, it MUST insert a "keep" Via=20
> > >parameter in the initial SIP request of those calls". =20
> This seems to=20
> > >imply that 'initial SIP request'
> > >means the initial INVITE.  'Initial INVITE' is what RFC 3261 calls=20
> > >the dialog-initiating INVITE.  Should the "keep"
> > >parameter in Via only be used in initial REGISTER and
> > initial INVITE ?
> >=20
> > I think that is the current assumption, yes, but I am not=20
> sure whether
> > we need to make that restriction.
> >=20
> > Regarding the usage of "initial SIP request", I guess it should say
> > "initial SIP dialog request", or something like that. I=20
> think similar
> > wording is used in some places of the document.
> [JRE] So the sender of an "initial SIP dialog request" would=20
> invoke the
> capability between itself and the next downstream SIP entity.=20
> Presumably
> a SIP proxy that does not record-route would  never invoke this
> capability. But what about the next SIP entity downstream, if that
> doesn't record-route? Presumably it too should refrain from=20
> engaging in
> this capability. Mid-dialog requests would by-pass any
> non-record-routing proxies, so there would be no opportunity=20
> at the time
> of the "initial SIP dialog request" to create, and hence=20
> keep-alive, the
> connections needed for mid-dialog requests. I couldn't find any
> discussion of this in the latest draft.
>=20
> John
>=20
>=20

From christer.holmberg@ericsson.com  Tue May  5 23:41:51 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "Tan, Ya Ching (NSN - DE/Munich)" <ya-ching.tan@nsn.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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Hi John,

I think there are cases where you may record route because you want to
do keep-alives.

But, we don't need to argue about that. I guess your point is that we
should allow the keep-alive negotiation also in mid-dialog request?

Regards,

Christer

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
> Sent: 6. toukokuuta 2009 9:18
> To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
> Cc: SIPCORE
> Subject: RE: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>=20
> Christer,
>=20
> That seems somewhat back-to-front. You don't record route=20
> (and hence determine the path of the dialog) just because you=20
> want to do keep-alives. You first decide whether you are to=20
> be on the path of a dialog, and then you should decide=20
> whether to do keep-alives with your neighbour. I guess where=20
> non-record-routing proxies are involved, this can be achieved=20
> only on the first mid-dialog request. However, for an=20
> INVITE-initiated dialog this might be the ACK request, but=20
> unfortunately that doesn't have a response, so it is not=20
> suitable. In some cases it would be an UPDATE or PRACK=20
> request during the early dialog. For a
> SUBSCRIBE- or REFER-initiated dialog it would be the first=20
> NOTIFY request.
>=20
> In extending the keep-alive principle to dialogs, we need to=20
> be very careful how we specify it.
>=20
> John
>=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: 05 May 2009 21:50
> > To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> > Cc: SIPCORE
> > Subject: RE: [sipcore] Draft submission:=20
> > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> >=20
> >=20
> > Hi John,
> >=20
> > You raise a good point. The draft should probably say that a SIP=20
> > entity MUST record-route (unless the entity is an endpoint)=20
> if it is=20
> > going to send and/or receive keep-alives.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> > Sent: Tuesday, May 05, 2009 4:53 PM
> > To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
> > Cc: SIPCORE
> > Subject: RE: [sipcore] Draft submission: draft-ietf-sipcore-keep-00=20
> > (wasRE:Document Adoption)
> >=20
> > =20
> >=20
> > > -----Original Message-----
> > > From: sipcore-bounces@ietf.org
> > > [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> > > Sent: 05 May 2009 13:04
> > > To: Tan, Ya Ching (NSN - DE/Munich)
> > > Cc: SIPCORE
> > > Subject: Re: [sipcore] Draft submission:=20
> > > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> > >=20
> > >=20
> > > >3. The draft repeatedly uses the term 'initial SIP request'. =20
> > > >What is the definition of this as it is not a known
> > technical term in
> >=20
> > > >SIP [RFC 3261].  The sentence in Section 5 "If the UAC wishes to=20
> > > >apply keep-alive for future calls, it MUST insert a "keep" Via=20
> > > >parameter in the initial SIP request of those calls".
> > This seems to
> > > >imply that 'initial SIP request'
> > > >means the initial INVITE.  'Initial INVITE' is what RFC=20
> 3261 calls=20
> > > >the dialog-initiating INVITE.  Should the "keep"
> > > >parameter in Via only be used in initial REGISTER and
> > > initial INVITE ?
> > >=20
> > > I think that is the current assumption, yes, but I am not
> > sure whether
> > > we need to make that restriction.
> > >=20
> > > Regarding the usage of "initial SIP request", I guess it=20
> should say=20
> > > "initial SIP dialog request", or something like that. I
> > think similar
> > > wording is used in some places of the document.
> > [JRE] So the sender of an "initial SIP dialog request" would invoke=20
> > the capability between itself and the next downstream SIP entity.
> > Presumably
> > a SIP proxy that does not record-route would  never invoke this=20
> > capability. But what about the next SIP entity downstream, if that=20
> > doesn't record-route? Presumably it too should refrain from=20
> engaging=20
> > in this capability. Mid-dialog requests would by-pass any=20
> > non-record-routing proxies, so there would be no opportunity at the=20
> > time of the "initial SIP dialog request" to create, and hence=20
> > keep-alive, the connections needed for mid-dialog requests.=20
> I couldn't=20
> > find any discussion of this in the latest draft.
> >=20
> > John
> >=20
> >=20
>=20

From john.elwell@siemens-enterprise.com  Wed May  6 00:26:38 2009
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Christer,

Using the mechanism on mid-dialog requests might be one of the
consequences. Another consequence might be that non-record-routing
proxies should not attempt to invoke keep-alive.  Perhaps we might
deliberately exclude the mechanism for use between two entities between
which there was one or more non-record-routing proxy. I don't really
know - I just know that there are some issues to do with
non-record-routing proxies that don't arise in the sip-outbound case
(because the UA and the outbound proxy always stay in on the path).

I would like to see some discussion about the applicability of
keep-alives to dialogs, before we jump in and propose a mechanism.

John=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 06 May 2009 07:43
> To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> Cc: SIPCORE
> Subject: RE: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>=20
>=20
> Hi John,
>=20
> I think there are cases where you may record route because you want to
> do keep-alives.
>=20
> But, we don't need to argue about that. I guess your point is that we
> should allow the keep-alive negotiation also in mid-dialog request?
>=20
> Regards,
>=20
> Christer
>=20
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
> > Sent: 6. toukokuuta 2009 9:18
> > To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
> > Cc: SIPCORE
> > Subject: RE: [sipcore] Draft submission:=20
> > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> >=20
> > Christer,
> >=20
> > That seems somewhat back-to-front. You don't record route=20
> > (and hence determine the path of the dialog) just because you=20
> > want to do keep-alives. You first decide whether you are to=20
> > be on the path of a dialog, and then you should decide=20
> > whether to do keep-alives with your neighbour. I guess where=20
> > non-record-routing proxies are involved, this can be achieved=20
> > only on the first mid-dialog request. However, for an=20
> > INVITE-initiated dialog this might be the ACK request, but=20
> > unfortunately that doesn't have a response, so it is not=20
> > suitable. In some cases it would be an UPDATE or PRACK=20
> > request during the early dialog. For a
> > SUBSCRIBE- or REFER-initiated dialog it would be the first=20
> > NOTIFY request.
> >=20
> > In extending the keep-alive principle to dialogs, we need to=20
> > be very careful how we specify it.
> >=20
> > John
> >=20
> > > -----Original Message-----
> > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > Sent: 05 May 2009 21:50
> > > To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> > > Cc: SIPCORE
> > > Subject: RE: [sipcore] Draft submission:=20
> > > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> > >=20
> > >=20
> > > Hi John,
> > >=20
> > > You raise a good point. The draft should probably say that a SIP=20
> > > entity MUST record-route (unless the entity is an endpoint)=20
> > if it is=20
> > > going to send and/or receive keep-alives.
> > >=20
> > > Regards,
> > >=20
> > > Christer
> > >=20
> > > -----Original Message-----
> > > From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> > > Sent: Tuesday, May 05, 2009 4:53 PM
> > > To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
> > > Cc: SIPCORE
> > > Subject: RE: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00=20
> > > (wasRE:Document Adoption)
> > >=20
> > > =20
> > >=20
> > > > -----Original Message-----
> > > > From: sipcore-bounces@ietf.org
> > > > [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> > > > Sent: 05 May 2009 13:04
> > > > To: Tan, Ya Ching (NSN - DE/Munich)
> > > > Cc: SIPCORE
> > > > Subject: Re: [sipcore] Draft submission:=20
> > > > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> > > >=20
> > > >=20
> > > > >3. The draft repeatedly uses the term 'initial SIP request'. =20
> > > > >What is the definition of this as it is not a known
> > > technical term in
> > >=20
> > > > >SIP [RFC 3261].  The sentence in Section 5 "If the UAC=20
> wishes to=20
> > > > >apply keep-alive for future calls, it MUST insert a "keep" Via=20
> > > > >parameter in the initial SIP request of those calls".
> > > This seems to
> > > > >imply that 'initial SIP request'
> > > > >means the initial INVITE.  'Initial INVITE' is what RFC=20
> > 3261 calls=20
> > > > >the dialog-initiating INVITE.  Should the "keep"
> > > > >parameter in Via only be used in initial REGISTER and
> > > > initial INVITE ?
> > > >=20
> > > > I think that is the current assumption, yes, but I am not
> > > sure whether
> > > > we need to make that restriction.
> > > >=20
> > > > Regarding the usage of "initial SIP request", I guess it=20
> > should say=20
> > > > "initial SIP dialog request", or something like that. I
> > > think similar
> > > > wording is used in some places of the document.
> > > [JRE] So the sender of an "initial SIP dialog request"=20
> would invoke=20
> > > the capability between itself and the next downstream SIP entity.
> > > Presumably
> > > a SIP proxy that does not record-route would  never invoke this=20
> > > capability. But what about the next SIP entity=20
> downstream, if that=20
> > > doesn't record-route? Presumably it too should refrain from=20
> > engaging=20
> > > in this capability. Mid-dialog requests would by-pass any=20
> > > non-record-routing proxies, so there would be no=20
> opportunity at the=20
> > > time of the "initial SIP dialog request" to create, and hence=20
> > > keep-alive, the connections needed for mid-dialog requests.=20
> > I couldn't=20
> > > find any discussion of this in the latest draft.
> > >=20
> > > John
> > >=20
> > >=20
> >=20
>=20

From christer.holmberg@ericsson.com  Wed May  6 02:13:40 2009
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Thread-Topic: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "Tan, Ya Ching (NSN - DE/Munich)" <ya-ching.tan@nsn.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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Hi,=20

>Using the mechanism on mid-dialog requests might be one of the
consequences.

Yes.

>Another consequence might be that non-record-routing proxies should not
attempt to invoke keep-alive.

Yes.

>Perhaps we might deliberately exclude the mechanism for use between two
entities between which there was one or more non-record-routing proxy.

I am not sure I understand. The keep-alives are always between two
neighbour entities. If there are other proxies (no matter if they are
record-routing or not) between the two entities, thoe two entities will
not send keep-alives between each other. When a SIP entity receives a
request, it only checks the TOPMOST Via if it contains "keep", so if
some other proxy has added a Via header (without "keep") no keep-alives
will be sent.

Regards,

Christer




>I don't really know - I just know that there are some issues to do with

>non-record-routing proxies that don't arise in the=20
>sip-outbound case (because the UA and the outbound proxy=20
>always stay in on the path).
>=20
> I would like to see some discussion about the applicability=20
> of keep-alives to dialogs, before we jump in and propose a mechanism.
>=20
> John=20
>=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: 06 May 2009 07:43
> > To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> > Cc: SIPCORE
> > Subject: RE: [sipcore] Draft submission:=20
> > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> >=20
> >=20
> > Hi John,
> >=20
> > I think there are cases where you may record route because=20
> you want to=20
> > do keep-alives.
> >=20
> > But, we don't need to argue about that. I guess your point=20
> is that we=20
> > should allow the keep-alive negotiation also in mid-dialog request?
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> > > -----Original Message-----
> > > From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> > > Sent: 6. toukokuuta 2009 9:18
> > > To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
> > > Cc: SIPCORE
> > > Subject: RE: [sipcore] Draft submission:=20
> > > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> > >=20
> > > Christer,
> > >=20
> > > That seems somewhat back-to-front. You don't record route=20
> (and hence=20
> > > determine the path of the dialog) just because you want to do=20
> > > keep-alives. You first decide whether you are to be on=20
> the path of a=20
> > > dialog, and then you should decide whether to do keep-alives with=20
> > > your neighbour. I guess where non-record-routing proxies are=20
> > > involved, this can be achieved only on the first=20
> mid-dialog request.=20
> > > However, for an INVITE-initiated dialog this might be the ACK=20
> > > request, but unfortunately that doesn't have a response, so it is=20
> > > not suitable. In some cases it would be an UPDATE or=20
> PRACK request=20
> > > during the early dialog. For a
> > > SUBSCRIBE- or REFER-initiated dialog it would be the first NOTIFY=20
> > > request.
> > >=20
> > > In extending the keep-alive principle to dialogs, we need=20
> to be very=20
> > > careful how we specify it.
> > >=20
> > > John
> > >=20
> > > > -----Original Message-----
> > > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > > Sent: 05 May 2009 21:50
> > > > To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> > > > Cc: SIPCORE
> > > > Subject: RE: [sipcore] Draft submission:=20
> > > > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> > > >=20
> > > >=20
> > > > Hi John,
> > > >=20
> > > > You raise a good point. The draft should probably say=20
> that a SIP=20
> > > > entity MUST record-route (unless the entity is an endpoint)
> > > if it is
> > > > going to send and/or receive keep-alives.
> > > >=20
> > > > Regards,
> > > >=20
> > > > Christer
> > > >=20
> > > > -----Original Message-----
> > > > From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> > > > Sent: Tuesday, May 05, 2009 4:53 PM
> > > > To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
> > > > Cc: SIPCORE
> > > > Subject: RE: [sipcore] Draft submission:=20
> > draft-ietf-sipcore-keep-00
> > > > (wasRE:Document Adoption)
> > > >=20
> > > > =20
> > > >=20
> > > > > -----Original Message-----
> > > > > From: sipcore-bounces@ietf.org
> > > > > [mailto:sipcore-bounces@ietf.org] On Behalf Of=20
> Christer Holmberg
> > > > > Sent: 05 May 2009 13:04
> > > > > To: Tan, Ya Ching (NSN - DE/Munich)
> > > > > Cc: SIPCORE
> > > > > Subject: Re: [sipcore] Draft submission:=20
> > > > > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> > > > >=20
> > > > >=20
> > > > > >3. The draft repeatedly uses the term 'initial SIP=20
> request'. =20
> > > > > >What is the definition of this as it is not a known
> > > > technical term in
> > > >=20
> > > > > >SIP [RFC 3261].  The sentence in Section 5 "If the UAC
> > wishes to
> > > > > >apply keep-alive for future calls, it MUST insert a=20
> "keep" Via=20
> > > > > >parameter in the initial SIP request of those calls".
> > > > This seems to
> > > > > >imply that 'initial SIP request'
> > > > > >means the initial INVITE.  'Initial INVITE' is what RFC
> > > 3261 calls
> > > > > >the dialog-initiating INVITE.  Should the "keep"
> > > > > >parameter in Via only be used in initial REGISTER and
> > > > > initial INVITE ?
> > > > >=20
> > > > > I think that is the current assumption, yes, but I am not
> > > > sure whether
> > > > > we need to make that restriction.
> > > > >=20
> > > > > Regarding the usage of "initial SIP request", I guess it
> > > should say
> > > > > "initial SIP dialog request", or something like that. I
> > > > think similar
> > > > > wording is used in some places of the document.
> > > > [JRE] So the sender of an "initial SIP dialog request"=20
> > would invoke
> > > > the capability between itself and the next downstream=20
> SIP entity.
> > > > Presumably
> > > > a SIP proxy that does not record-route would  never invoke this=20
> > > > capability. But what about the next SIP entity
> > downstream, if that
> > > > doesn't record-route? Presumably it too should refrain from
> > > engaging
> > > > in this capability. Mid-dialog requests would by-pass any=20
> > > > non-record-routing proxies, so there would be no
> > opportunity at the
> > > > time of the "initial SIP dialog request" to create, and hence=20
> > > > keep-alive, the connections needed for mid-dialog requests.
> > > I couldn't
> > > > find any discussion of this in the latest draft.
> > > >=20
> > > > John
> > > >=20
> > > >=20
> > >=20
> >=20
>=20

From john.elwell@siemens-enterprise.com  Wed May  6 03:04:28 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 06 May 2009 09:59
> To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> Cc: SIPCORE
> Subject: RE: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>=20
>=20
> Hi,=20
>=20
> >Using the mechanism on mid-dialog requests might be one of the
> consequences.
>=20
> Yes.
>=20
> >Another consequence might be that non-record-routing proxies=20
> should not
> attempt to invoke keep-alive.
>=20
> Yes.
>=20
> >Perhaps we might deliberately exclude the mechanism for use=20
> between two
> entities between which there was one or more non-record-routing proxy.
>=20
> I am not sure I understand. The keep-alives are always between two
> neighbour entities. If there are other proxies (no matter if they are
> record-routing or not) between the two entities, thoe two=20
> entities will
> not send keep-alives between each other. When a SIP entity receives a
> request, it only checks the TOPMOST Via if it contains "keep", so if
> some other proxy has added a Via header (without "keep") no=20
> keep-alives
> will be sent.
[JRE] Suppose an INVITE request is routed:
	UA1 -> Proxy1 - >Proxy2 -> Proxy3 -> UA2
Suppose also that Proxy1 and Proxy3 record-route, but Proxy2 does not.
There is no point in doing keep-alive between Proxy1 and Proxy2, or
between Proxy2 and Proxy3. When the first mid-dialog request is sent, it
will go directly from Proxy1 to Proxy3 or vice versa. My question was
whether we want to allow the invocation of keep-alive on that hop
(Proxy1 to Proxy3) or whether we want to exclude this? Bear in mind that
the first mid-dialog request (excluding ACK) could occur some
considerable time after the call is established (e.g., an UPDATE request
when session timer expires), so the benefit of keep-alive would last
only between that request and the final BYE request.

For SUBSCRIBE- and REFER-initiated dialogs it is a little simpler,
because there is always an immediate NOTIFY request to form the dialog.

John
>=20

From christer.holmberg@ericsson.com  Wed May  6 05:38:11 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "Tan, Ya Ching (NSN - DE/Munich)" <ya-ching.tan@nsn.com>
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Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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Hi,=20

>>>Perhaps we might deliberately exclude the mechanism for use
>>>between two entities between which there was one or more=20
>>>non-record-routing proxy.
>>=20
>>I am not sure I understand. The keep-alives are always between two=20
>>neighbour entities. If there are other proxies (no matter if they are=20
>>record-routing or not) between the two entities, thoe two entities=20
>>will not send keep-alives between each other. When a SIP entity=20
>>receives a request, it only checks the TOPMOST Via if it contains=20
>>"keep", so if some other proxy has added a Via header (without "keep")
no keep-alives will be sent.
>>
>[JRE] Suppose an INVITE request is routed:
>
>UA1 -> Proxy1 - >Proxy2 -> Proxy3 -> UA2=20
>
>Suppose also that Proxy1 and Proxy3 record-route, but Proxy2 does not.
>
>There is no point in doing keep-alive between Proxy1 and Proxy2, or
between Proxy2 and Proxy3.

Correct. So, in that case, if Proxy2 doesn't Record-Route, it should not
insert "yes" in the Via received from Proxy1, and it should not insert
"keep" in its Via sent towards Proxy3.


>When the first mid-dialog request is sent, it will go directly from
Proxy1 to Proxy3 or vice versa.=20

Correct.


>My question was whether we want to allow the invocation of keep-alive
on that hop (Proxy1 to Proxy3) or whether we want to exclude this?=20

Technically I see no harm in allowing sending keep-alives TO an entity
which didn't record-route, eventhough I am not sure it is useful.

However, I don't think it shall be possible to send keep-alives FROM an
entity which didn't record-route. Because, that entity would not know
when the registration, or the session, terminates, in order to stop
sending keep-alives.


>Bear in mind that the first mid-dialog request (excluding ACK)=20
>could occur some considerable time after the call is=20
>established (e.g., an UPDATE request when session timer=20
>expires), so the benefit of keep-alive would last only=20
>between that request and the final BYE request.

Yes.

But, I think that in many (most?) cases keep-alives will be sent between
entities which are going to be in the signalling path anyway.

For example, I think one of the most common use-case will be between a
UA and an edge proxy. So, the UA inserts "keep" in the Via of the
initial request, and the edge proxy will record-route and insert "yes".


>For SUBSCRIBE- and REFER-initiated dialogs it is a little simpler,
because there is always an immediate NOTIFY request to form the dialog.

Yes.

Regards,

Christer


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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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I have never had a problem with the idea of capturing this info in a 
table of this sort. It isn't perfect (some nuances still require text 
descriptions) but still useful.

What is a problem about the table is maintaining it as extensions are 
made. Some things add new rows, and other things add new columns. We 
don't have a mechanism for getting it all together.

A web site might be useful, but we would need some way to maintain it 
normatively. This is similar to the kinds of registries that IANA keeps 
for us, but it is more complex than anything else I have seen them do. 
We could explore if they could maintain such a structure. But that would 
not obviate the RFCs that introduce changes from enumerating the impact 
of those changes on all previously defined rows and columns.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi, 
> 
>>> One issue which was raised by Ya-Ching, when giving comments on the 
>>> keep-draft, was whether it is allowed to use the Flow-Timer header in 
>>> other messages than REGISTER responses?
>>>
>>> The keep draft also uses it in INVITEs when keep-alive is negotiated.
>>>
>>> The procedures in Outbound naturally only mention the Flow-Timer 
>>> header in REGISTER responses, but it is nowever specifically indicated
> 
>>> that the header is applicable for REGISTER responses only.
>>>
>>> What seems to be missing from Outbound is the table we normally use to
> 
>>> define for which methods and message types a header is applicable.
>>>
>>> Example:
>>>
>>> Header field          where   proxy ACK BYE CAN INV OPT REG
>>> ----------------------------------------------------------------------
>>> ----------------------
>>>
>>> Flow-Timer             18x                -      -       -       
>>> o     -       -
>>> Flow-Timer             2xx                -      -       -       
>>> o     -       o
>>>
>> That's probably because that table is the source of more confusion,
> IMHO, than anything else in the entire SIP specification family. It
> direly needs a retrofit, although I'm not sure that such is really
>> possible: we add another column to the table for each method-type
> extension, such as SUB, NOT, INF, etc. This raises the question: If a
> new request type is specified, is it up to that request-type to fix
>> "the table" for every header field, or do we need to revise the RC for
> each extension so as to add the new column to their tables?
>> Consequently, I greatly prefer to specify the behavior in text, using
> more general terms such as "requests" and "responses", which specific
> caveats around where such an extension header is meaningful, 
>> where it is forbidden, and where it is ignorable.
> 
> I think the table is clear.
> 
> Maybe we could have a table on some website. Then it would be easy to
> update it every time new methods and/or headers are added.
> 
> Regards,
> 
> Christer
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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Spencer Dawkins wrote:
> I'm kind of in the middle here...
> 
>> I think the table is clear.
> 
> I think that conceptually it is, or should be, clear. Having it split 
> over a large number of documents today makes it less clear :-(
> 
>> Maybe we could have a table on some website. Then it would be easy to
>> update it every time new methods and/or headers are added.
> 
> I think Dean's concern wasn't about the technology ("need more 
> columns"), but about the review and approval process - if there's one 
> table, and I create a new method or header, is it my job to fill in the 
> entire table for the new row/column, or is it someone else's job 
> (keeping in mind that I may not be the smartest person about everything 
> that's in the table now - I might just be the smartest person about the 
> method or header that I'm defining. If three of us define two rows and a 
> column, how do we ensure that there's enough review for the new version 
> of the table?
> 
> Stuff like that, I think. Or Dean can correct me.

Those are valid concerns. But they are valid whether there is a table or 
not. If we expect text in the document to replace the table, then the 
text has to address the same concerns.

There is no way out - the interaction of a new method with all headers 
and responses must be considered, and the interaction of a new header or 
response with all methods must be considered.

	Thanks,
	Paul

> Thanks,
> 
> Spencer
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00	(wasRE:Document Adoption)
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Christer Holmberg wrote:
> Hi John,
> 
> I think there are cases where you may record route because you want to
> do keep-alives.

Christer - can you explain that. I agree with John that it seems 
backwards. What would be a use case where the decision the do keep 
alives comes first and the need to r-r in order to be present for those 
keep alives is secondary?

	Thanks,
	Paul

> But, we don't need to argue about that. I guess your point is that we
> should allow the keep-alive negotiation also in mid-dialog request?
> 
> Regards,
> 
> Christer
> 
>> -----Original Message-----
>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com] 
>> Sent: 6. toukokuuta 2009 9:18
>> To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
>> Cc: SIPCORE
>> Subject: RE: [sipcore] Draft submission: 
>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>
>> Christer,
>>
>> That seems somewhat back-to-front. You don't record route 
>> (and hence determine the path of the dialog) just because you 
>> want to do keep-alives. You first decide whether you are to 
>> be on the path of a dialog, and then you should decide 
>> whether to do keep-alives with your neighbour. I guess where 
>> non-record-routing proxies are involved, this can be achieved 
>> only on the first mid-dialog request. However, for an 
>> INVITE-initiated dialog this might be the ACK request, but 
>> unfortunately that doesn't have a response, so it is not 
>> suitable. In some cases it would be an UPDATE or PRACK 
>> request during the early dialog. For a
>> SUBSCRIBE- or REFER-initiated dialog it would be the first 
>> NOTIFY request.
>>
>> In extending the keep-alive principle to dialogs, we need to 
>> be very careful how we specify it.
>>
>> John
>>
>>> -----Original Message-----
>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>> Sent: 05 May 2009 21:50
>>> To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
>>> Cc: SIPCORE
>>> Subject: RE: [sipcore] Draft submission: 
>>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>>
>>>
>>> Hi John,
>>>
>>> You raise a good point. The draft should probably say that a SIP 
>>> entity MUST record-route (unless the entity is an endpoint) 
>> if it is 
>>> going to send and/or receive keep-alives.
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>> -----Original Message-----
>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>> Sent: Tuesday, May 05, 2009 4:53 PM
>>> To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
>>> Cc: SIPCORE
>>> Subject: RE: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 
>>> (wasRE:Document Adoption)
>>>
>>>  
>>>
>>>> -----Original Message-----
>>>> From: sipcore-bounces@ietf.org
>>>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
>>>> Sent: 05 May 2009 13:04
>>>> To: Tan, Ya Ching (NSN - DE/Munich)
>>>> Cc: SIPCORE
>>>> Subject: Re: [sipcore] Draft submission: 
>>>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>>>
>>>>
>>>>> 3. The draft repeatedly uses the term 'initial SIP request'.  
>>>>> What is the definition of this as it is not a known
>>> technical term in
>>>
>>>>> SIP [RFC 3261].  The sentence in Section 5 "If the UAC wishes to 
>>>>> apply keep-alive for future calls, it MUST insert a "keep" Via 
>>>>> parameter in the initial SIP request of those calls".
>>> This seems to
>>>>> imply that 'initial SIP request'
>>>>> means the initial INVITE.  'Initial INVITE' is what RFC 
>> 3261 calls 
>>>>> the dialog-initiating INVITE.  Should the "keep"
>>>>> parameter in Via only be used in initial REGISTER and
>>>> initial INVITE ?
>>>>
>>>> I think that is the current assumption, yes, but I am not
>>> sure whether
>>>> we need to make that restriction.
>>>>
>>>> Regarding the usage of "initial SIP request", I guess it 
>> should say 
>>>> "initial SIP dialog request", or something like that. I
>>> think similar
>>>> wording is used in some places of the document.
>>> [JRE] So the sender of an "initial SIP dialog request" would invoke 
>>> the capability between itself and the next downstream SIP entity.
>>> Presumably
>>> a SIP proxy that does not record-route would  never invoke this 
>>> capability. But what about the next SIP entity downstream, if that 
>>> doesn't record-route? Presumably it too should refrain from 
>> engaging 
>>> in this capability. Mid-dialog requests would by-pass any 
>>> non-record-routing proxies, so there would be no opportunity at the 
>>> time of the "initial SIP dialog request" to create, and hence 
>>> keep-alive, the connections needed for mid-dialog requests. 
>> I couldn't 
>>> find any discussion of this in the latest draft.
>>>
>>> John
>>>
>>>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From spencer@wonderhamster.org  Wed May  6 07:55:50 2009
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To: "Paul Kyzivat" <pkyzivat@cisco.com>, "Christer Holmberg" <christer.holmberg@ericsson.com>
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Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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I'm probably not being clear here.

> What is a problem about the table is maintaining it as extensions are 
> made. Some things add new rows, and other things add new columns. We don't 
> have a mechanism for getting it all together.
>
> A web site might be useful, but we would need some way to maintain it 
> normatively. This is similar to the kinds of registries that IANA keeps 
> for us, but it is more complex than anything else I have seen them do. We 
> could explore if they could maintain such a structure. But that would not 
> obviate the RFCs that introduce changes from enumerating the impact of 
> those changes on all previously defined rows and columns.

What I was saying was that we know how to write, and review, multiple 
extensions in parallel using Internet Drafts, but we don't actually have 
infrastructure in place to do this using a single table.

We certainly COULD create this table as an IANA registry - but that's not 
what we have now.

I think Paul and I are agreeing that this would involve a gatekeeper who is 
more involved, earlier in the process, with more authority to say "no", than 
IANA usually is today.

This isn't a technical concern, it's a process concern.

Thanks,

Spencer 



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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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Spencer,

I agree with you, but I'm saying a bit more:

Whether we have the table or not, the need for the gatekeeping is there. 
Trying to keep the table straight just *exposes* the need more clearly.
When new message types, headers, or return codes are defined, the info 
equivalent to updating the table is needed, whether in the form of a 
table update or not.

	Thanks,
	Paul

Spencer Dawkins wrote:
> I'm probably not being clear here.
> 
>> What is a problem about the table is maintaining it as extensions are 
>> made. Some things add new rows, and other things add new columns. We 
>> don't have a mechanism for getting it all together.
>>
>> A web site might be useful, but we would need some way to maintain it 
>> normatively. This is similar to the kinds of registries that IANA 
>> keeps for us, but it is more complex than anything else I have seen 
>> them do. We could explore if they could maintain such a structure. But 
>> that would not obviate the RFCs that introduce changes from 
>> enumerating the impact of those changes on all previously defined rows 
>> and columns.
> 
> What I was saying was that we know how to write, and review, multiple 
> extensions in parallel using Internet Drafts, but we don't actually have 
> infrastructure in place to do this using a single table.
> 
> We certainly COULD create this table as an IANA registry - but that's 
> not what we have now.
> 
> I think Paul and I are agreeing that this would involve a gatekeeper who 
> is more involved, earlier in the process, with more authority to say 
> "no", than IANA usually is today.
> 
> This isn't a technical concern, it's a process concern.
> 
> Thanks,
> 
> Spencer
> 
> 

From christer.holmberg@ericsson.com  Wed May  6 11:27:03 2009
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Thread-Topic: [sipcore] Draft submission: draft-ietf-sipcore-keep-00	(wasRE:Document Adoption)
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00	(wasRE:Document Adoption)
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Hi,=20

>>I think there are cases where you may record route because you want to
do keep-alives.
>
>Christer - can you explain that. I agree with John that it seems
backwards. What would be a use case where the decision the do keep
alives comes first and the need to r-r in order to be present for those=20
>keep alives is secondary?

I am not sure I understand what is meant by "first and secondary". My
point is that an entity can choose to Record-Route because it wants to
either send or receive keep-alives (in the same way that it may choose
to record-record due to whatever other reasons). An edge proxy, if it
detects that the UA is behind a NAT, could Record-Route in order to
allow the UA to send keep-alives towards it.

But, of course, we can allow an entity to indicate that it is willing to
receive keep-alives (by inserting a "yes" value) even if it doesn't
record-route.

Regards,

Christer




> But, we don't need to argue about that. I guess your point is that we=20
> should allow the keep-alive negotiation also in mid-dialog request?
>=20
> Regards,
>=20
> Christer
>=20
>> -----Original Message-----
>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>> Sent: 6. toukokuuta 2009 9:18
>> To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
>> Cc: SIPCORE
>> Subject: RE: [sipcore] Draft submission:=20
>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>
>> Christer,
>>
>> That seems somewhat back-to-front. You don't record route (and hence=20
>> determine the path of the dialog) just because you want to do=20
>> keep-alives. You first decide whether you are to be on the path of a=20
>> dialog, and then you should decide whether to do keep-alives with=20
>> your neighbour. I guess where non-record-routing proxies are=20
>> involved, this can be achieved only on the first mid-dialog request.=20
>> However, for an INVITE-initiated dialog this might be the ACK=20
>> request, but unfortunately that doesn't have a response, so it is not

>> suitable. In some cases it would be an UPDATE or PRACK request during

>> the early dialog. For a
>> SUBSCRIBE- or REFER-initiated dialog it would be the first NOTIFY=20
>> request.
>>
>> In extending the keep-alive principle to dialogs, we need to be very=20
>> careful how we specify it.
>>
>> John
>>
>>> -----Original Message-----
>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>> Sent: 05 May 2009 21:50
>>> To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
>>> Cc: SIPCORE
>>> Subject: RE: [sipcore] Draft submission:=20
>>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>>
>>>
>>> Hi John,
>>>
>>> You raise a good point. The draft should probably say that a SIP=20
>>> entity MUST record-route (unless the entity is an endpoint)
>> if it is
>>> going to send and/or receive keep-alives.
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>> -----Original Message-----
>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>> Sent: Tuesday, May 05, 2009 4:53 PM
>>> To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
>>> Cc: SIPCORE
>>> Subject: RE: [sipcore] Draft submission: draft-ietf-sipcore-keep-00=20
>>> (wasRE:Document Adoption)
>>>
>>> =20
>>>
>>>> -----Original Message-----
>>>> From: sipcore-bounces@ietf.org
>>>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
>>>> Sent: 05 May 2009 13:04
>>>> To: Tan, Ya Ching (NSN - DE/Munich)
>>>> Cc: SIPCORE
>>>> Subject: Re: [sipcore] Draft submission:=20
>>>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>>>
>>>>
>>>>> 3. The draft repeatedly uses the term 'initial SIP request'. =20
>>>>> What is the definition of this as it is not a known
>>> technical term in
>>>
>>>>> SIP [RFC 3261].  The sentence in Section 5 "If the UAC wishes to=20
>>>>> apply keep-alive for future calls, it MUST insert a "keep" Via=20
>>>>> parameter in the initial SIP request of those calls".
>>> This seems to
>>>>> imply that 'initial SIP request'
>>>>> means the initial INVITE.  'Initial INVITE' is what RFC
>> 3261 calls
>>>>> the dialog-initiating INVITE.  Should the "keep"
>>>>> parameter in Via only be used in initial REGISTER and
>>>> initial INVITE ?
>>>>
>>>> I think that is the current assumption, yes, but I am not
>>> sure whether
>>>> we need to make that restriction.
>>>>
>>>> Regarding the usage of "initial SIP request", I guess it
>> should say
>>>> "initial SIP dialog request", or something like that. I
>>> think similar
>>>> wording is used in some places of the document.
>>> [JRE] So the sender of an "initial SIP dialog request" would invoke=20
>>> the capability between itself and the next downstream SIP entity.
>>> Presumably
>>> a SIP proxy that does not record-route would  never invoke this=20
>>> capability. But what about the next SIP entity downstream, if that=20
>>> doesn't record-route? Presumably it too should refrain from
>> engaging
>>> in this capability. Mid-dialog requests would by-pass any=20
>>> non-record-routing proxies, so there would be no opportunity at the=20
>>> time of the "initial SIP dialog request" to create, and hence=20
>>> keep-alive, the connections needed for mid-dialog requests.
>> I couldn't
>>> find any discussion of this in the latest draft.
>>>
>>> John
>>>
>>>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

From spencer@wonderhamster.org  Wed May  6 11:37:26 2009
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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We agree on the bit more as well. 

Thanks,

Spencer


> Spencer,
> 
> I agree with you, but I'm saying a bit more:
> 
> Whether we have the table or not, the need for the gatekeeping is there. 
> Trying to keep the table straight just *exposes* the need more clearly.
> When new message types, headers, or return codes are defined, the info 
> equivalent to updating the table is needed, whether in the form of a 
> table update or not.
> 
> Thanks,
> Paul


From dean.willis@softarmor.com  Wed May  6 11:43:45 2009
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Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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On May 6, 2009, at 8:31 AM, Paul Kyzivat wrote:

>
>
> Spencer Dawkins wrote:
>> I'm kind of in the middle here...
>>> I think the table is clear.
>> I think that conceptually it is, or should be, clear. Having it  
>> split over a large number of documents today makes it less clear :-(
>>> Maybe we could have a table on some website. Then it would be easy  
>>> to
>>> update it every time new methods and/or headers are added.
>> I think Dean's concern wasn't about the technology ("need more  
>> columns"), but about the review and approval process - if there's  
>> one table, and I create a new method or header, is it my job to  
>> fill in the entire table for the new row/column, or is it someone  
>> else's job (keeping in mind that I may not be the smartest person  
>> about everything that's in the table now - I might just be the  
>> smartest person about the method or header that I'm defining. If  
>> three of us define two rows and a column, how do we ensure that  
>> there's enough review for the new version of the table?
>> Stuff like that, I think. Or Dean can correct me.
>
> Those are valid concerns. But they are valid whether there is a  
> table or not. If we expect text in the document to replace the  
> table, then the text has to address the same concerns.

I agree.

>
>
> There is no way out - the interaction of a new method with all  
> headers and responses must be considered, and the interaction of a  
> new header or response with all methods must be considered.
>

Although strictly from a documentation procedure: the current format  
wouldn't allow enough columns to appear in an internet-draft format to  
actually allow a new RFC to properly updated the table.

Plus, there are a lot of "conditionals" encoded in the table as  
additional row comments. I'm pretty sure that this confuses a lot of  
people -- I know it confuses me. Of course, it would help if we didn't  
keep defining special cases of application-affecting-transport logic  
like weird responses codes that Change the Rules.

Perhaps some sort of formal language construct could express the logic  
in some sort of fsm-verifiable way?

--
Dean






From christer.holmberg@ericsson.com  Wed May  6 12:04:21 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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>>There is no way out - the interaction of a new method with all headers

>>and responses must be considered, and the interaction of a new header=20
>>or response with all methods must be considered.
>
>
>Although strictly from a documentation procedure: the current format
wouldn't allow enough columns to appear in an internet-draft format to
actually allow a new RFC to properly updated the table.
>
>Plus, there are a lot of "conditionals" encoded in the table as
additional row comments. I'm pretty sure that this confuses a lot of
people -- I know it confuses me. Of course, it would help if we didn't=20
>keep defining special cases of application-affecting-transport logic
like weird responses codes that Change the Rules.
>
>Perhaps some sort of formal language construct could express the logic
in some sort of fsm-verifiable way?

I guess we could start using the same mechanism as in 3GPP TS 24.229
Annex A. I am sure Keith would be more and happy to teach us :)

Regards,

Christer








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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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Christer,=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 06 May 2009 13:31
> To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> Cc: SIPCORE
> Subject: RE: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>=20
>=20
> Hi,=20
>=20
> >>>Perhaps we might deliberately exclude the mechanism for use
> >>>between two entities between which there was one or more=20
> >>>non-record-routing proxy.
> >>=20
> >>I am not sure I understand. The keep-alives are always between two=20
> >>neighbour entities. If there are other proxies (no matter=20
> if they are=20
> >>record-routing or not) between the two entities, thoe two entities=20
> >>will not send keep-alives between each other. When a SIP entity=20
> >>receives a request, it only checks the TOPMOST Via if it contains=20
> >>"keep", so if some other proxy has added a Via header=20
> (without "keep")
> no keep-alives will be sent.
> >>
> >[JRE] Suppose an INVITE request is routed:
> >
> >UA1 -> Proxy1 - >Proxy2 -> Proxy3 -> UA2=20
> >
> >Suppose also that Proxy1 and Proxy3 record-route, but Proxy2=20
> does not.
> >
> >There is no point in doing keep-alive between Proxy1 and Proxy2, or
> between Proxy2 and Proxy3.
>=20
> Correct. So, in that case, if Proxy2 doesn't Record-Route, it=20
> should not
> insert "yes" in the Via received from Proxy1, and it should not insert
> "keep" in its Via sent towards Proxy3.
>=20
>=20
> >When the first mid-dialog request is sent, it will go directly from
> Proxy1 to Proxy3 or vice versa.=20
>=20
> Correct.
>=20
>=20
> >My question was whether we want to allow the invocation of keep-alive
> on that hop (Proxy1 to Proxy3) or whether we want to exclude this?=20
>=20
> Technically I see no harm in allowing sending keep-alives TO an entity
> which didn't record-route, eventhough I am not sure it is useful.
[JRE] That is not the point. I am asking about keep-alives between
Proxy1 and Proxy3, both of which record-routed.

John



>=20
> However, I don't think it shall be possible to send=20
> keep-alives FROM an
> entity which didn't record-route. Because, that entity would not know
> when the registration, or the session, terminates, in order to stop
> sending keep-alives.
>=20
>=20
> >Bear in mind that the first mid-dialog request (excluding ACK)=20
> >could occur some considerable time after the call is=20
> >established (e.g., an UPDATE request when session timer=20
> >expires), so the benefit of keep-alive would last only=20
> >between that request and the final BYE request.
>=20
> Yes.
>=20
> But, I think that in many (most?) cases keep-alives will be=20
> sent between
> entities which are going to be in the signalling path anyway.
>=20
> For example, I think one of the most common use-case will be between a
> UA and an edge proxy. So, the UA inserts "keep" in the Via of the
> initial request, and the edge proxy will record-route and=20
> insert "yes".
>=20
>=20
> >For SUBSCRIBE- and REFER-initiated dialogs it is a little simpler,
> because there is always an immediate NOTIFY request to form=20
> the dialog.
>=20
> Yes.
>=20
> Regards,
>=20
> Christer
>=20
>=20

From christer.holmberg@ericsson.com  Wed May  6 13:58:06 2009
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References: <49FA1267.5040204@nostrum.com> <CA9998CD4A020D418654FCDEF4E707DF0CBA68B7@esealmw113.eemea.ericsson.se> <601FBEDC0E5A5B4E8AFC11D2CF149AB8DFF11D@DEMUEXC014.nsn-intra.net> <CA9998CD4A020D418654FCDEF4E707DF0CC974A7@esealmw113.eemea.ericsson.se> <0D5F89FAC29E2C41B98A6A762007F5D001D8DF9B@GBNTHT12009MSX.gb002.siemens.net> <CA9998CD4A020D418654FCDEF4E707DF0B16821B@esealmw113.eemea.ericsson.se> <0D5F89FAC29E2C41B98A6A762007F5D001DE15C3@GBNTHT12009MSX.gb002.siemens.net> <CA9998CD4A020D418654FCDEF4E707DF0CCC8B8F@esealmw113.eemea.ericsson.se> <0D5F89FAC29E2C41B98A6A762007F5D001DE15D2@GBNTHT12009MSX.gb002.siemens.net> <CA9998CD4A020D418654FCDEF4E707DF0CCC9245@esealmw113.eemea.ericsson.se> <0D5F89FAC29E2C41B98A6A762007F5D001DE1785@GBNTHT12009MSX.gb002.siemens.net> <CA9998CD4A020D418654FCDEF4E707DF0CCFC049@esealmw113.eemea.ericsson.se> <0D5F89FAC29E2C41B98A6A762007F5D001DE1AD4@GBNTHT12009MSX.gb002.siemens.net>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "Tan, Ya Ching (NSN - DE/Munich)" <ya-ching.tan@nsn.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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Hi,=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: 06 May 2009 13:31
> To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> Cc: SIPCORE
> Subject: RE: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>=20
>=20
> Hi,
>=20
> >>>Perhaps we might deliberately exclude the mechanism for use between

> >>>two entities between which there was one or more non-record-routing

> >>>proxy.
> >>=20
> >>I am not sure I understand. The keep-alives are always between two=20
> >>neighbour entities. If there are other proxies (no matter
> if they are
> >>record-routing or not) between the two entities, thoe two entities=20
> >>will not send keep-alives between each other. When a SIP entity=20
> >>receives a request, it only checks the TOPMOST Via if it contains=20
> >>"keep", so if some other proxy has added a Via header
> (without "keep")
> no keep-alives will be sent.
> >>
> >[JRE] Suppose an INVITE request is routed:
> >
> >UA1 -> Proxy1 - >Proxy2 -> Proxy3 -> UA2
> >
> >Suppose also that Proxy1 and Proxy3 record-route, but Proxy2
> does not.
> >
> >There is no point in doing keep-alive between Proxy1 and Proxy2, or
> between Proxy2 and Proxy3.
>=20
> Correct. So, in that case, if Proxy2 doesn't Record-Route, it should=20
> not insert "yes" in the Via received from Proxy1, and it should not=20
> insert "keep" in its Via sent towards Proxy3.
>=20
>=20
> >When the first mid-dialog request is sent, it will go directly from
> Proxy1 to Proxy3 or vice versa.=20
>=20
> Correct.
>=20
>=20
> >My question was whether we want to allow the invocation of keep-alive
> on that hop (Proxy1 to Proxy3) or whether we want to exclude this?=20
>=20
> Technically I see no harm in allowing sending keep-alives TO an entity

> which didn't record-route, eventhough I am not sure it is useful.
>[JRE] That is not the point. I am asking about keep-alives between
>Proxy1 and Proxy3, both of which record-routed.

In that case the top Via (Proxy2) will not contain the "keep" parameter.
So, Proxy3 would need to make sure that the last entity (Proxy1) which
inserted "keep" is also the last entity which record-routed. And, I am
not sure we want to start comparing Via- and Record-Route headers in
order to figure that out.

An alternative would of course be to define "keep" as a Record-Route
(and Path) parameter. Then it doesn't matter how many proxies (read: Via
headers) there are between Proxy1 and Proxy3, as long as the top
Record-Route contains the keep parameter.

However, UAs don't insert Record-Route, so a UAC that supports sending
of keep-alives would have to insert "keep" somewhere else.

Regards,

Christer





> However, I don't think it shall be possible to send keep-alives FROM=20
> an entity which didn't record-route. Because, that entity would not=20
> know when the registration, or the session, terminates, in order to=20
> stop sending keep-alives.
>=20
>=20
> >Bear in mind that the first mid-dialog request (excluding ACK) could=20
> >occur some considerable time after the call is established (e.g., an=20
> >UPDATE request when session timer expires), so the benefit of=20
> >keep-alive would last only between that request and the final BYE=20
> >request.
>=20
> Yes.
>=20
> But, I think that in many (most?) cases keep-alives will be sent=20
> between entities which are going to be in the signalling path anyway.
>=20
> For example, I think one of the most common use-case will be between a

> UA and an edge proxy. So, the UA inserts "keep" in the Via of the=20
> initial request, and the edge proxy will record-route and insert=20
> "yes".
>=20
>=20
> >For SUBSCRIBE- and REFER-initiated dialogs it is a little simpler,
> because there is always an immediate NOTIFY request to form the=20
> dialog.
>=20
> Yes.
>=20
> Regards,
>=20
> Christer
>=20
>=20

From john.elwell@siemens-enterprise.com  Wed May  6 23:14:40 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 06 May 2009 21:59
> To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> Cc: SIPCORE
> Subject: RE: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>=20
>=20
> Hi,=20
>=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: 06 May 2009 13:31
> > To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> > Cc: SIPCORE
> > Subject: RE: [sipcore] Draft submission:=20
> > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> >=20
> >=20
> > Hi,
> >=20
> > >>>Perhaps we might deliberately exclude the mechanism for=20
> use between
>=20
> > >>>two entities between which there was one or more=20
> non-record-routing
>=20
> > >>>proxy.
> > >>=20
> > >>I am not sure I understand. The keep-alives are always=20
> between two=20
> > >>neighbour entities. If there are other proxies (no matter
> > if they are
> > >>record-routing or not) between the two entities, thoe two=20
> entities=20
> > >>will not send keep-alives between each other. When a SIP entity=20
> > >>receives a request, it only checks the TOPMOST Via if it contains=20
> > >>"keep", so if some other proxy has added a Via header
> > (without "keep")
> > no keep-alives will be sent.
> > >>
> > >[JRE] Suppose an INVITE request is routed:
> > >
> > >UA1 -> Proxy1 - >Proxy2 -> Proxy3 -> UA2
> > >
> > >Suppose also that Proxy1 and Proxy3 record-route, but Proxy2
> > does not.
> > >
> > >There is no point in doing keep-alive between Proxy1 and Proxy2, or
> > between Proxy2 and Proxy3.
> >=20
> > Correct. So, in that case, if Proxy2 doesn't Record-Route,=20
> it should=20
> > not insert "yes" in the Via received from Proxy1, and it should not=20
> > insert "keep" in its Via sent towards Proxy3.
> >=20
> >=20
> > >When the first mid-dialog request is sent, it will go directly from
> > Proxy1 to Proxy3 or vice versa.=20
> >=20
> > Correct.
> >=20
> >=20
> > >My question was whether we want to allow the invocation of=20
> keep-alive
> > on that hop (Proxy1 to Proxy3) or whether we want to exclude this?=20
> >=20
> > Technically I see no harm in allowing sending keep-alives=20
> TO an entity
>=20
> > which didn't record-route, eventhough I am not sure it is useful.
> >[JRE] That is not the point. I am asking about keep-alives between
> >Proxy1 and Proxy3, both of which record-routed.
>=20
> In that case the top Via (Proxy2) will not contain the "keep"=20
> parameter.
> So, Proxy3 would need to make sure that the last entity (Proxy1) which
> inserted "keep" is also the last entity which record-routed. And, I am
> not sure we want to start comparing Via- and Record-Route headers in
> order to figure that out.
[JRE] No, I would not want to do that. My point is that at the time of
the dialog-forming request, there will be no direct transport connection
between proxy1 and proxy3, so keep-alive is not relevant. However, as a
result of the first mid-dialog request, there will be a direct transport
connection between Proxy1 and Proxy3. So the question is whether we want
to figure out a way of keeping that alive or not.

>=20
> An alternative would of course be to define "keep" as a Record-Route
> (and Path) parameter. Then it doesn't matter how many proxies=20
> (read: Via
> headers) there are between Proxy1 and Proxy3, as long as the top
> Record-Route contains the keep parameter.
>=20
> However, UAs don't insert Record-Route, so a UAC that supports sending
> of keep-alives would have to insert "keep" somewhere else.
[JRE] And also it would not apply to REGISTER keep-alives. This would be
a significant departure from what we are chartered to do -  I am not
sure we want to go there.

To summarise:
- The mechanism we currently have works for REGISTER-initiated
keep-alives.
- It also works for dialog-based keep-alives between entities that were
adjacent at dialog-initiation time and remain on the dialog path.
- It does not work for dialog-based keep-alives between entities that
were not adjacent at dialog-initiation time but become adjacent
subsequently (as a result of intermediate proxies not record-routing).

The last case does not arise if B2BUAs are used or all proxies
record-route. Does the WG want to address the last case or not?

John

From christer.holmberg@ericsson.com  Thu May  7 07:45:19 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "Tan, Ya Ching (NSN - DE/Munich)" <ya-ching.tan@nsn.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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Hi,=20

> > > Hi,
> > >=20
> > > >>>Perhaps we might deliberately exclude the mechanism for
> > use between
> >=20
> > > >>>two entities between which there was one or more
> > non-record-routing
> >=20
> > > >>>proxy.
> > > >>=20
> > > >>I am not sure I understand. The keep-alives are always
> > between two
> > > >>neighbour entities. If there are other proxies (no matter
> > > if they are
> > > >>record-routing or not) between the two entities, thoe two
> > entities
> > > >>will not send keep-alives between each other. When a SIP entity=20
> > > >>receives a request, it only checks the TOPMOST Via if=20
> it contains=20
> > > >>"keep", so if some other proxy has added a Via header
> > > (without "keep")
> > > no keep-alives will be sent.
> > > >>
> > > >[JRE] Suppose an INVITE request is routed:
> > > >
> > > >UA1 -> Proxy1 - >Proxy2 -> Proxy3 -> UA2
> > > >
> > > >Suppose also that Proxy1 and Proxy3 record-route, but Proxy2
> > > does not.
> > > >
> > > >There is no point in doing keep-alive between Proxy1 and=20
> Proxy2, or
> > > between Proxy2 and Proxy3.
> > >=20
> > > Correct. So, in that case, if Proxy2 doesn't Record-Route,
> > it should
> > > not insert "yes" in the Via received from Proxy1, and it=20
> should not=20
> > > insert "keep" in its Via sent towards Proxy3.
> > >=20
> > >=20
> > > >When the first mid-dialog request is sent, it will go=20
> directly from
> > > Proxy1 to Proxy3 or vice versa.=20
> > >=20
> > > Correct.
> > >=20
> > >=20
> > > >My question was whether we want to allow the invocation of
> > keep-alive
> > > on that hop (Proxy1 to Proxy3) or whether we want to=20
> exclude this?=20
> > >=20
> > > Technically I see no harm in allowing sending keep-alives
> > TO an entity
> >=20
>>>which didn't record-route, eventhough I am not sure it is useful.
>>>[JRE] That is not the point. I am asking about keep-alives between
>>>Proxy1 and Proxy3, both of which record-routed.
>>=20
>>In that case the top Via (Proxy2) will not contain the "keep"
parameter.
>>So, Proxy3 would need to make sure that the last entity (Proxy1) which

>>inserted "keep" is also the last entity which record-routed. And, I am

>>not sure we want to start comparing Via- and Record-Route headers in=20
>>order to figure that out.
>[JRE] No, I would not want to do that. My point is that at=20
>the time of the dialog-forming request, there will be no=20
>direct transport connection between proxy1 and proxy3, so=20
>keep-alive is not relevant. However, as a result of the first=20
>mid-dialog request, there will be a direct transport=20
>connection between Proxy1 and Proxy3. So the question is=20
>whether we want to figure out a way of keeping that alive or not.

I think there will be a direct transport connection between proxy1 and
proxy3 already at the time of the response to the dialog-forming
request. So, you can start sending keep-alives at that point. You don't
need to wait for the first mid-dialog request.


=20
>>An alternative would of course be to define "keep" as a Record-Route=20
>>and Path) parameter. Then it doesn't matter how many proxies
>>(read: Via headers) there are between Proxy1 and Proxy3, as long as
the top=20
>>Record-Route contains the keep parameter.
>>=20
>>However, UAs don't insert Record-Route, so a UAC that supports sending

>>of keep-alives would have to insert "keep" somewhere else.
>[JRE] And also it would not apply to REGISTER keep-alives.=20
>This would be a significant departure from what we are=20
>chartered to do -  I am not sure we want to go there.

For REGISTER you would use the Path header. Outbound also uses that.


Regards,

Christer


>To summarise:
>- The mechanism we currently have works for REGISTER-initiated
keep-alives.
>- It also works for dialog-based keep-alives between entities=20
>that were adjacent at dialog-initiation time and remain on=20
>the dialog path.
>- It does not work for dialog-based keep-alives between=20
>entities that were not adjacent at dialog-initiation time but=20
>become adjacent subsequently (as a result of intermediate=20
>proxies not record-routing).
>=20
>The last case does not arise if B2BUAs are used or all=20
>proxies record-route. Does the WG want to address the last=20
>case or not?
>=20
> John
>=20

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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00	(wasRE:Document Adoption)
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Christer Holmberg wrote:
> Hi, 
> 
>>> I think there are cases where you may record route because you want to
> do keep-alives.
>> Christer - can you explain that. I agree with John that it seems
> backwards. What would be a use case where the decision the do keep
> alives comes first and the need to r-r in order to be present for those 
>> keep alives is secondary?
> 
> I am not sure I understand what is meant by "first and secondary". My
> point is that an entity can choose to Record-Route because it wants to
> either send or receive keep-alives (in the same way that it may choose
> to record-record due to whatever other reasons).

I agree that it *could* do that.
But I fail to understand *why* it would do that?
Aren't the keepalives there to assure there is a working communication 
path between them? If it has no other reason to be on a path, then why 
would it want the keepalives? They don't have value in their own right.

> An edge proxy, if it
> detects that the UA is behind a NAT, could Record-Route in order to
> allow the UA to send keep-alives towards it.

Maybe its just semantics, but in that case the reason it is R-R'ing is 
not because it wants to do keepalives. It is R-R'ing in order to 
maintain a working path to the UA. Doing keepalives is something else 
might need to do to keep that working path up.

> But, of course, we can allow an entity to indicate that it is willing to
> receive keep-alives (by inserting a "yes" value) even if it doesn't
> record-route.

Does that mean that keepalives should be sent even though the target of 
the keepalives is of no use?

	Thanks,
	Paul

> Regards,
> 
> Christer
> 
> 
> 
> 
>> But, we don't need to argue about that. I guess your point is that we 
>> should allow the keep-alive negotiation also in mid-dialog request?
>>
>> Regards,
>>
>> Christer
>>
>>> -----Original Message-----
>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>> Sent: 6. toukokuuta 2009 9:18
>>> To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
>>> Cc: SIPCORE
>>> Subject: RE: [sipcore] Draft submission: 
>>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>>
>>> Christer,
>>>
>>> That seems somewhat back-to-front. You don't record route (and hence 
>>> determine the path of the dialog) just because you want to do 
>>> keep-alives. You first decide whether you are to be on the path of a 
>>> dialog, and then you should decide whether to do keep-alives with 
>>> your neighbour. I guess where non-record-routing proxies are 
>>> involved, this can be achieved only on the first mid-dialog request. 
>>> However, for an INVITE-initiated dialog this might be the ACK 
>>> request, but unfortunately that doesn't have a response, so it is not
> 
>>> suitable. In some cases it would be an UPDATE or PRACK request during
> 
>>> the early dialog. For a
>>> SUBSCRIBE- or REFER-initiated dialog it would be the first NOTIFY 
>>> request.
>>>
>>> In extending the keep-alive principle to dialogs, we need to be very 
>>> careful how we specify it.
>>>
>>> John
>>>
>>>> -----Original Message-----
>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>>> Sent: 05 May 2009 21:50
>>>> To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
>>>> Cc: SIPCORE
>>>> Subject: RE: [sipcore] Draft submission: 
>>>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>>>
>>>>
>>>> Hi John,
>>>>
>>>> You raise a good point. The draft should probably say that a SIP 
>>>> entity MUST record-route (unless the entity is an endpoint)
>>> if it is
>>>> going to send and/or receive keep-alives.
>>>>
>>>> Regards,
>>>>
>>>> Christer
>>>>
>>>> -----Original Message-----
>>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>>> Sent: Tuesday, May 05, 2009 4:53 PM
>>>> To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
>>>> Cc: SIPCORE
>>>> Subject: RE: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 
>>>> (wasRE:Document Adoption)
>>>>
>>>>  
>>>>
>>>>> -----Original Message-----
>>>>> From: sipcore-bounces@ietf.org
>>>>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
>>>>> Sent: 05 May 2009 13:04
>>>>> To: Tan, Ya Ching (NSN - DE/Munich)
>>>>> Cc: SIPCORE
>>>>> Subject: Re: [sipcore] Draft submission: 
>>>>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>>>>
>>>>>
>>>>>> 3. The draft repeatedly uses the term 'initial SIP request'.  
>>>>>> What is the definition of this as it is not a known
>>>> technical term in
>>>>
>>>>>> SIP [RFC 3261].  The sentence in Section 5 "If the UAC wishes to 
>>>>>> apply keep-alive for future calls, it MUST insert a "keep" Via 
>>>>>> parameter in the initial SIP request of those calls".
>>>> This seems to
>>>>>> imply that 'initial SIP request'
>>>>>> means the initial INVITE.  'Initial INVITE' is what RFC
>>> 3261 calls
>>>>>> the dialog-initiating INVITE.  Should the "keep"
>>>>>> parameter in Via only be used in initial REGISTER and
>>>>> initial INVITE ?
>>>>>
>>>>> I think that is the current assumption, yes, but I am not
>>>> sure whether
>>>>> we need to make that restriction.
>>>>>
>>>>> Regarding the usage of "initial SIP request", I guess it
>>> should say
>>>>> "initial SIP dialog request", or something like that. I
>>>> think similar
>>>>> wording is used in some places of the document.
>>>> [JRE] So the sender of an "initial SIP dialog request" would invoke 
>>>> the capability between itself and the next downstream SIP entity.
>>>> Presumably
>>>> a SIP proxy that does not record-route would  never invoke this 
>>>> capability. But what about the next SIP entity downstream, if that 
>>>> doesn't record-route? Presumably it too should refrain from
>>> engaging
>>>> in this capability. Mid-dialog requests would by-pass any 
>>>> non-record-routing proxies, so there would be no opportunity at the 
>>>> time of the "initial SIP dialog request" to create, and hence 
>>>> keep-alive, the connections needed for mid-dialog requests.
>>> I couldn't
>>>> find any discussion of this in the latest draft.
>>>>
>>>> John
>>>>
>>>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
> 

From christer.holmberg@ericsson.com  Thu May  7 10:02:01 2009
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Thread-Topic: [sipcore] Draft submission: draft-ietf-sipcore-keep-00	(wasRE:Document Adoption)
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00	(wasRE:Document Adoption)
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Hi,=20

>>>>I think there are cases where you may record route because you want=20
>>>>to do keep-alives.
>>>Christer - can you explain that. I agree with John that it seems
backwards. What would be a use case where the decision the do keep=20
>>>alives comes first and the need to r-r in order to be present for
those
>>>keep alives is secondary?
>>=20
>>I am not sure I understand what is meant by "first and secondary". My=20
>>point is that an entity can choose to Record-Route because it wants to

>>either send or receive keep-alives (in the same way that it may choose

>>to record-record due to whatever other reasons).
>
>I agree that it *could* do that.
>But I fail to understand *why* it would do that?
>Aren't the keepalives there to assure there is a working communication
path between them? If it has no other reason to be on a path, then why
would it want the keepalives? They don't have value in their=20
>own right.

I agree, so in that case there probably wouldn't be any keep-alives
negotiated. The receiving entity would not Record-Route, and it would
not insert "keep".

But, the comments was that an entity first decides to Record-Route, and
then decides to do keep-alives. So, when the entity has decided to
Record-Route it can insert "keep" in the same request.


>>An edge proxy, if it detects that the UA is behind a NAT, could
Record-Route in order to=20
>>allow the UA to send keep-alives towards it.
>
>Maybe its just semantics, but in that case the reason it is R-R'ing is
not because it wants to do keepalives. It is R-R'ing in order to
maintain a working path to the UA. Doing keepalives is something=20
>else might need to do to keep that working path up.

Yes.

>>But, of course, we can allow an entity to indicate that it is willing=20
>>to receive keep-alives (by inserting a "yes" value) even if it doesn't

>>record-route.
>
>Does that mean that keepalives should be sent even though the target of
the keepalives is of no use?

Well, if there is no use to send keep-alives, there will be no "yes"
inserted.

Assume I insert "keep", and forwards a request to you. No keep-alives
are going to be sent until you actually add a "yes" value. And, if there
is no reason for keep-alives you will not add "yes", and there won't be
any keep-alives.=20

It is always the receiving entity which decides (by inserting "yes")
whether there will be keep-alives or not. The sending entity only
indicates (by inserting "keep") that he is able to send keep-alives.

Regards,

Christer



> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>> But, we don't need to argue about that. I guess your point is that we

>> should allow the keep-alive negotiation also in mid-dialog request?
>>
>> Regards,
>>
>> Christer
>>
>>> -----Original Message-----
>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>> Sent: 6. toukokuuta 2009 9:18
>>> To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
>>> Cc: SIPCORE
>>> Subject: RE: [sipcore] Draft submission:=20
>>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>>
>>> Christer,
>>>
>>> That seems somewhat back-to-front. You don't record route (and hence

>>> determine the path of the dialog) just because you want to do=20
>>> keep-alives. You first decide whether you are to be on the path of a

>>> dialog, and then you should decide whether to do keep-alives with=20
>>> your neighbour. I guess where non-record-routing proxies are=20
>>> involved, this can be achieved only on the first mid-dialog request.
>>> However, for an INVITE-initiated dialog this might be the ACK=20
>>> request, but unfortunately that doesn't have a response, so it is=20
>>> not
>=20
>>> suitable. In some cases it would be an UPDATE or PRACK request=20
>>> during
>=20
>>> the early dialog. For a
>>> SUBSCRIBE- or REFER-initiated dialog it would be the first NOTIFY=20
>>> request.
>>>
>>> In extending the keep-alive principle to dialogs, we need to be very

>>> careful how we specify it.
>>>
>>> John
>>>
>>>> -----Original Message-----
>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>>> Sent: 05 May 2009 21:50
>>>> To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
>>>> Cc: SIPCORE
>>>> Subject: RE: [sipcore] Draft submission:=20
>>>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>>>
>>>>
>>>> Hi John,
>>>>
>>>> You raise a good point. The draft should probably say that a SIP=20
>>>> entity MUST record-route (unless the entity is an endpoint)
>>> if it is
>>>> going to send and/or receive keep-alives.
>>>>
>>>> Regards,
>>>>
>>>> Christer
>>>>
>>>> -----Original Message-----
>>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>>> Sent: Tuesday, May 05, 2009 4:53 PM
>>>> To: Christer Holmberg; Tan, Ya Ching (NSN - DE/Munich)
>>>> Cc: SIPCORE
>>>> Subject: RE: [sipcore] Draft submission: draft-ietf-sipcore-keep-00

>>>> (wasRE:Document Adoption)
>>>>
>>>> =20
>>>>
>>>>> -----Original Message-----
>>>>> From: sipcore-bounces@ietf.org
>>>>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
>>>>> Sent: 05 May 2009 13:04
>>>>> To: Tan, Ya Ching (NSN - DE/Munich)
>>>>> Cc: SIPCORE
>>>>> Subject: Re: [sipcore] Draft submission:=20
>>>>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>>>>
>>>>>
>>>>>> 3. The draft repeatedly uses the term 'initial SIP request'. =20
>>>>>> What is the definition of this as it is not a known
>>>> technical term in
>>>>
>>>>>> SIP [RFC 3261].  The sentence in Section 5 "If the UAC wishes to=20
>>>>>> apply keep-alive for future calls, it MUST insert a "keep" Via=20
>>>>>> parameter in the initial SIP request of those calls".
>>>> This seems to
>>>>>> imply that 'initial SIP request'
>>>>>> means the initial INVITE.  'Initial INVITE' is what RFC
>>> 3261 calls
>>>>>> the dialog-initiating INVITE.  Should the "keep"
>>>>>> parameter in Via only be used in initial REGISTER and
>>>>> initial INVITE ?
>>>>>
>>>>> I think that is the current assumption, yes, but I am not
>>>> sure whether
>>>>> we need to make that restriction.
>>>>>
>>>>> Regarding the usage of "initial SIP request", I guess it
>>> should say
>>>>> "initial SIP dialog request", or something like that. I
>>>> think similar
>>>>> wording is used in some places of the document.
>>>> [JRE] So the sender of an "initial SIP dialog request" would invoke

>>>> the capability between itself and the next downstream SIP entity.
>>>> Presumably
>>>> a SIP proxy that does not record-route would  never invoke this=20
>>>> capability. But what about the next SIP entity downstream, if that=20
>>>> doesn't record-route? Presumably it too should refrain from
>>> engaging
>>>> in this capability. Mid-dialog requests would by-pass any=20
>>>> non-record-routing proxies, so there would be no opportunity at the

>>>> time of the "initial SIP dialog request" to create, and hence=20
>>>> keep-alive, the connections needed for mid-dialog requests.
>>> I couldn't
>>>> find any discussion of this in the latest draft.
>>>>
>>>> John
>>>>
>>>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>=20

From john.elwell@siemens-enterprise.com  Thu May  7 12:59:45 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 07 May 2009 09:12
> To: Elwell, John; Tan, Ya Ching (NSN - DE/Munich)
> Cc: SIPCORE
> Subject: RE: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>=20
>=20
> Hi,=20
>=20
> > > > Hi,
> > > >=20
> > > > >>>Perhaps we might deliberately exclude the mechanism for
> > > use between
> > >=20
> > > > >>>two entities between which there was one or more
> > > non-record-routing
> > >=20
> > > > >>>proxy.
> > > > >>=20
> > > > >>I am not sure I understand. The keep-alives are always
> > > between two
> > > > >>neighbour entities. If there are other proxies (no matter
> > > > if they are
> > > > >>record-routing or not) between the two entities, thoe two
> > > entities
> > > > >>will not send keep-alives between each other. When a=20
> SIP entity=20
> > > > >>receives a request, it only checks the TOPMOST Via if=20
> > it contains=20
> > > > >>"keep", so if some other proxy has added a Via header
> > > > (without "keep")
> > > > no keep-alives will be sent.
> > > > >>
> > > > >[JRE] Suppose an INVITE request is routed:
> > > > >
> > > > >UA1 -> Proxy1 - >Proxy2 -> Proxy3 -> UA2
> > > > >
> > > > >Suppose also that Proxy1 and Proxy3 record-route, but Proxy2
> > > > does not.
> > > > >
> > > > >There is no point in doing keep-alive between Proxy1 and=20
> > Proxy2, or
> > > > between Proxy2 and Proxy3.
> > > >=20
> > > > Correct. So, in that case, if Proxy2 doesn't Record-Route,
> > > it should
> > > > not insert "yes" in the Via received from Proxy1, and it=20
> > should not=20
> > > > insert "keep" in its Via sent towards Proxy3.
> > > >=20
> > > >=20
> > > > >When the first mid-dialog request is sent, it will go=20
> > directly from
> > > > Proxy1 to Proxy3 or vice versa.=20
> > > >=20
> > > > Correct.
> > > >=20
> > > >=20
> > > > >My question was whether we want to allow the invocation of
> > > keep-alive
> > > > on that hop (Proxy1 to Proxy3) or whether we want to=20
> > exclude this?=20
> > > >=20
> > > > Technically I see no harm in allowing sending keep-alives
> > > TO an entity
> > >=20
> >>>which didn't record-route, eventhough I am not sure it is useful.
> >>>[JRE] That is not the point. I am asking about keep-alives between
> >>>Proxy1 and Proxy3, both of which record-routed.
> >>=20
> >>In that case the top Via (Proxy2) will not contain the "keep"
> parameter.
> >>So, Proxy3 would need to make sure that the last entity=20
> (Proxy1) which
>=20
> >>inserted "keep" is also the last entity which=20
> record-routed. And, I am
>=20
> >>not sure we want to start comparing Via- and Record-Route=20
> headers in=20
> >>order to figure that out.
> >[JRE] No, I would not want to do that. My point is that at=20
> >the time of the dialog-forming request, there will be no=20
> >direct transport connection between proxy1 and proxy3, so=20
> >keep-alive is not relevant. However, as a result of the first=20
> >mid-dialog request, there will be a direct transport=20
> >connection between Proxy1 and Proxy3. So the question is=20
> >whether we want to figure out a way of keeping that alive or not.
>=20
> I think there will be a direct transport connection between proxy1 and
> proxy3 already at the time of the response to the dialog-forming
> request.=20
[JRE] I don't understand how that can happen, since the response will
follow the path of the Via entries, not the path or record-routes.
Please explain.

John




> So, you can start sending keep-alives at that point.=20
> You don't
> need to wait for the first mid-dialog request.
>=20
>=20
> =20
> >>An alternative would of course be to define "keep" as a=20
> Record-Route=20
> >>and Path) parameter. Then it doesn't matter how many proxies
> >>(read: Via headers) there are between Proxy1 and Proxy3, as long as
> the top=20
> >>Record-Route contains the keep parameter.
> >>=20
> >>However, UAs don't insert Record-Route, so a UAC that=20
> supports sending
>=20
> >>of keep-alives would have to insert "keep" somewhere else.
> >[JRE] And also it would not apply to REGISTER keep-alives.=20
> >This would be a significant departure from what we are=20
> >chartered to do -  I am not sure we want to go there.
>=20
> For REGISTER you would use the Path header. Outbound also uses that.
>=20
>=20
> Regards,
>=20
> Christer
>=20
>=20
> >To summarise:
> >- The mechanism we currently have works for REGISTER-initiated
> keep-alives.
> >- It also works for dialog-based keep-alives between entities=20
> >that were adjacent at dialog-initiation time and remain on=20
> >the dialog path.
> >- It does not work for dialog-based keep-alives between=20
> >entities that were not adjacent at dialog-initiation time but=20
> >become adjacent subsequently (as a result of intermediate=20
> >proxies not record-routing).
> >=20
> >The last case does not arise if B2BUAs are used or all=20
> >proxies record-route. Does the WG want to address the last=20
> >case or not?
> >=20
> > John
> >=20
>=20

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From: "Spencer Dawkins" <spencer@wonderhamster.org>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "Christer Holmberg" <christer.holmberg@ericsson.com>, "Tan, Ya Ching \(NSN - DE/Munich\)" <ya-ching.tan@nsn.com>
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Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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>> >[JRE] No, I would not want to do that. My point is that at
>> >the time of the dialog-forming request, there will be no
>> >direct transport connection between proxy1 and proxy3, so
>> >keep-alive is not relevant. However, as a result of the first
>> >mid-dialog request, there will be a direct transport
>> >connection between Proxy1 and Proxy3. So the question is
>> >whether we want to figure out a way of keeping that alive or not.
>>
>> I think there will be a direct transport connection between proxy1 and
>> proxy3 already at the time of the response to the dialog-forming
>> request.
> [JRE] I don't understand how that can happen, since the response will
> follow the path of the Via entries, not the path or record-routes.
> Please explain.

I'm as confused as John is. For example, if the signaling path is using TCP, 
there would be a TCP connection between Proxy1 and Proxy2, and a separate 
TCP connection between Proxy2 and Proxy3, but this would not cause a 
SYN/SYN-ACK/ACK exchange between Proxy1 and Proxy3 - or am I missing 
something?

Thanks,

Spencer 



From christer.holmberg@ericsson.com  Fri May  8 03:42:23 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Spencer Dawkins" <spencer@wonderhamster.org>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "Tan, Ya Ching (NSN - DE/Munich)" <ya-ching.tan@nsn.com>
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Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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Hi,

>>>>[JRE] No, I would not want to do that. My point is that at the time=20
>>>>of the dialog-forming request, there will be no direct transport=20
>>>>connection between proxy1 and proxy3, so keep-alive is not relevant.

>>>>However, as a result of the first mid-dialog request, there will be=20
>>>>a direct transport connection between Proxy1 and Proxy3. So the=20
>>>>question is whether we want to figure out a way of keeping that=20
>>>>alive or not.
>>>
>>>I think there will be a direct transport connection between proxy1
and
>>>proxy3 already at the time of the response to the dialog-forming=20
>>>request.
>>[JRE] I don't understand how that can happen, since the response will=20
>>follow the path of the Via entries, not the path or record-routes.
>>Please explain.
>=20
>I'm as confused as John is. For example, if the signaling=20
>path is using TCP, there would be a TCP connection between=20
>Proxy1 and Proxy2, and a separate TCP connection between=20
>Proxy2 and Proxy3, but this would not cause a SYN/SYN-ACK/ACK=20
>exchange between Proxy1 and Proxy3 - or am I missing something?

No, you are not :)

The TCP connection between Proxy1 and Proxy3 will be established when
something is sent from Proxy1 to Proxy3. That would normally be a
mid-dialog request, but it COULD of course also be the keep-alive
message.

But, if we don't want the keep-alive message to establish the TCP
connection we can simply decide that keep-alives can be negotiated only
between adjacent nodes, and we can keep the Via parameter based
solution.=20

Then, it would not be possible to negotiate keep-alives between Proxy1
and Proxy3 during the initial request, since Proxy2 would still insert a
Via (it would then be possible to negotiate keep-alives between Proxy1
and Proxy3 during a mid-dialog request, becuase then they will be
adjacent).

Regards,

Christer


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Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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2009/5/8 Christer Holmberg <christer.holmberg@ericsson.com>:
>
> Hi,
>
>>>>>[JRE] No, I would not want to do that. My point is that at the time
>>>>>of the dialog-forming request, there will be no direct transport
>>>>>connection between proxy1 and proxy3, so keep-alive is not relevant.
>
>>>>>However, as a result of the first mid-dialog request, there will be
>>>>>a direct transport connection between Proxy1 and Proxy3. So the
>>>>>question is whether we want to figure out a way of keeping that
>>>>>alive or not.
>>>>
>>>>I think there will be a direct transport connection between proxy1
> and
>>>>proxy3 already at the time of the response to the dialog-forming
>>>>request.
>>>[JRE] I don't understand how that can happen, since the response will
>>>follow the path of the Via entries, not the path or record-routes.
>>>Please explain.
>>
>>I'm as confused as John is. For example, if the signaling
>>path is using TCP, there would be a TCP connection between
>>Proxy1 and Proxy2, and a separate TCP connection between
>>Proxy2 and Proxy3, but this would not cause a SYN/SYN-ACK/ACK
>>exchange between Proxy1 and Proxy3 - or am I missing something?
>
> No, you are not :)
>
> The TCP connection between Proxy1 and Proxy3 will be established when
> something is sent from Proxy1 to Proxy3. That would normally be a
> mid-dialog request, but it COULD of course also be the keep-alive
> message.
>
> But, if we don't want the keep-alive message to establish the TCP
> connection we can simply decide that keep-alives can be negotiated only
> between adjacent nodes, and we can keep the Via parameter based
> solution.
>
> Then, it would not be possible to negotiate keep-alives between Proxy1
> and Proxy3 during the initial request, since Proxy2 would still insert a
> Via (it would then be possible to negotiate keep-alives between Proxy1
> and Proxy3 during a mid-dialog request, becuase then they will be
> adjacent).

John's query is not about the possibility of negotiating keep-alives
between Proxy1 and Proxy3 during initial message exchange. He is
asking if we have a requirement to negotiate keep-alives between
proxy1 and proxy3 using mid-dialog messages.

I think its a valid scenario and we should work on a solution.

Hisham

>
> Regards,
>
> Christer
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

From drage@alcatel-lucent.com  Mon May 11 06:13:23 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Dean Willis <dean.willis@softarmor.com>, Paul Kyzivat <pkyzivat@cisco.com>
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Thread-Topic: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER responses?
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I actually suggest that the tables are insufficient, and can only be regard=
ed as a summary, therefore you do need to say precisely what is required in=
 the text as well (from both a sending and receiving perspective). And I wo=
uld note that what is being specified in these tables is actually dynamic, =
rather than some static conformance.

I do not believe a maintained web site is the way to go - our external regi=
stries are not there to specify behaviour, they are they to stop duplicate =
assignment of values. If we go this way, we immediately run into the proble=
m that the normative behaviour is meant to be specified in RFCs, so we stil=
l have a maintenance problem.

What I do suggest that in the text where you specify the usage of headers w=
ithin individual methods, you try and be as method unspecific as possible. =
If, for example, you are allowing a header to have meaning in REFER, INVITE=
 and SUBSCRIBE requests because they are all dialog creating requests, then=
 write the requirements as such. That way, if a new method is created, we c=
an easily extrapolate this sort of text to the new method. Most of our head=
er usage can be categorised fairly easily in this manner.

Francois, your proposed outbound change belongs on the SIP list as that is =
still PROTO shepherded there. Perhaps you can update with additions as I su=
ggest above.

regards

Keith

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: Wednesday, May 06, 2009 7:56 PM
> To: Dean Willis; Paul Kyzivat
> Cc: SIPCORE
> Subject: Re: [sipcore] Outbound: Flow-Timer only in REGISTER=20
> responses?
>=20
>=20
> >>There is no way out - the interaction of a new method with=20
> all headers
>=20
> >>and responses must be considered, and the interaction of a=20
> new header=20
> >>or response with all methods must be considered.
> >
> >
> >Although strictly from a documentation procedure: the current format
> wouldn't allow enough columns to appear in an internet-draft=20
> format to actually allow a new RFC to properly updated the table.
> >
> >Plus, there are a lot of "conditionals" encoded in the table as
> additional row comments. I'm pretty sure that this confuses a=20
> lot of people -- I know it confuses me. Of course, it would=20
> help if we didn't=20
> >keep defining special cases of application-affecting-transport logic
> like weird responses codes that Change the Rules.
> >
> >Perhaps some sort of formal language construct could express=20
> the logic
> in some sort of fsm-verifiable way?
>=20
> I guess we could start using the same mechanism as in 3GPP TS=20
> 24.229 Annex A. I am sure Keith would be more and happy to teach us :)
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

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--NextPart

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


	Title           : Session Initiation Protocol (SIP) Event Notification Extension for Notification Rate Control
	Author(s)       : A. Niemi, et al.
	Filename        : draft-ietf-sipcore-event-rate-control-00.txt
	Pages           : 21
	Date            : 2009-05-11

This document specifies mechanisms for adjusting the rate of Session
Initiation Protocol (SIP) event notifications.  These mechanisms can
be applied in subscriptions to all SIP event packages.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-event-rate-control-00.txt

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From salvatore.loreto@ericsson.com  Tue May 12 04:26:57 2009
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-event-rate-control-00.txt
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Hi there,

I have submitted a new version of the 
"draft-niemi-sipping-event-throttle", this time as wg item.
The draft is now named "draft-ietf-sipcore-event-rate-control"
and the title has changed in:
"Session Initiation Protocol (SIP) Event Notification Extension for 
Notification Rate Control".

in this version we have  done the  following changes to address all the 
feedbacks and suggestions
we got in the mailing list discussion and during the last IETF meeting:

- renamed the parameter names as "min-interval", "max-interval", 
"average-interval"
- addressed the split of normative and non-normative sections and 
introduced some more normative sentences where we were lacking those
- deleted some redundant text and merged some sections
- clarified the text in several places using the newly chosen 
terminology: maximum/minimum/average rate mechanism
- added some words about hidden throttling
- inserted the suggestion we got about "not to break the normal 
retransmission mechanism"
- some other more minor changes

we haven't done any changes around the Timeout formula yet; as we do 
consider the formula in the draft just a possible suggestion (we will add
some statement about this fact in the next version of the draft);
a possible change could be add some additional formulas based on the 
suggestions  we got in the mailing discussion
and leave the choice of formula for implementation.

cheers
Sal

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 Core Working Group of the IETF.
>
>
> 	Title           : Session Initiation Protocol (SIP) Event Notification Extension for Notification Rate Control
> 	Author(s)       : A. Niemi, et al.
> 	Filename        : draft-ietf-sipcore-event-rate-control-00.txt
> 	Pages           : 21
> 	Date            : 2009-05-11
>
> This document specifies mechanisms for adjusting the rate of Session
> Initiation Protocol (SIP) event notifications.  These mechanisms can
> be applied in subscriptions to all SIP event packages.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sipcore-event-rate-control-00.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
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>   
> ------------------------------------------------------------------------
>
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>   


From john.elwell@siemens-enterprise.com  Tue May 12 23:22:35 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Cullen Jennings <fluffy@cisco.com>, Jon Peterson <jon.peterson@neustar.biz>, SIPCORE <sipcore@ietf.org>
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Subject: [sipcore] Inconsistency in RFC 3325?
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What looks like an inconsistency in RFC 3325 has been identified.

The table in 9.2 states that PPI can be added ("a") by a proxy:
"      Header field         where   proxy   ACK  BYE  CAN  INV  OPT  REG
      ------------         -----   -----   ---  ---  ---  ---  ---  ---
      P-Preferred-Identity          adr     -    o    -    o    o    -


                                           SUB  NOT  REF  INF  UPD  PRA
                                           ---  ---  ---  ---  ---  ---
                                            o    o    o    -    -    -"

However, earlier in 9.2 it states:
"The P-Preferred-Identity header field is used from a user agent to a
   trusted proxy to carry the identity the user sending the SIP message
   wishes to be used for the P-Asserted-Header field value that the
   trusted element will insert."
i.e., no mention of being sent from a proxy.

Moreover, in 6 it states:
"The proxy MUST remove the user-provided P-Preferred-
   Identity header from any message it forwards."
i.e., quite explicit about not forwarding, which one might also
interpret as not being allowed to insert.

So can PPI be added by a proxy, and if so what is its meaning? Has there
been any earlier discussion on this topic?

John

From pkyzivat@cisco.com  Wed May 13 06:06:20 2009
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Inconsistency in RFC 3325?
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John,

I don't know if the proxy usage was intended or not.
But neither do I see harm in permitting a proxy to insert the header.
Perhaps there is a *really dumb* UA, with a proxy in front of it 
providing some smarts, before reaching the trusted (and secured) device 
that acts on the PPI.

	Thanks,
	Paul

Elwell, John wrote:
> What looks like an inconsistency in RFC 3325 has been identified.
> 
> The table in 9.2 states that PPI can be added ("a") by a proxy:
> "      Header field         where   proxy   ACK  BYE  CAN  INV  OPT  REG
>       ------------         -----   -----   ---  ---  ---  ---  ---  ---
>       P-Preferred-Identity          adr     -    o    -    o    o    -
> 
> 
>                                            SUB  NOT  REF  INF  UPD  PRA
>                                            ---  ---  ---  ---  ---  ---
>                                             o    o    o    -    -    -"
> 
> However, earlier in 9.2 it states:
> "The P-Preferred-Identity header field is used from a user agent to a
>    trusted proxy to carry the identity the user sending the SIP message
>    wishes to be used for the P-Asserted-Header field value that the
>    trusted element will insert."
> i.e., no mention of being sent from a proxy.
> 
> Moreover, in 6 it states:
> "The proxy MUST remove the user-provided P-Preferred-
>    Identity header from any message it forwards."
> i.e., quite explicit about not forwarding, which one might also
> interpret as not being allowed to insert.
> 
> So can PPI be added by a proxy, and if so what is its meaning? Has there
> been any earlier discussion on this topic?
> 
> John
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From dean.willis@softarmor.com  Wed May 13 07:41:57 2009
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Subject: Re: [sipcore] Inconsistency in RFC 3325?
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Paul Kyzivat wrote:
> I don't know if the proxy usage was intended or not.
> But neither do I see harm in permitting a proxy to insert the header.
> Perhaps there is a *really dumb* UA, with a proxy in front of it
> providing some smarts, before reaching the trusted (and secured) device
> that acts on the PPI.

I seem to recall discussing just exactly that use case. IIRC, somebody
in 3GPP proposed that the P-CSCF should be able to insert the default
preferred identity if the mobile terminal failed to insert one.


> 
> Elwell, John wrote:
>> What looks like an inconsistency in RFC 3325 has been identified.
>>
>> The table in 9.2 states that PPI can be added ("a") by a proxy:
>> "      Header field         where   proxy   ACK  BYE  CAN  INV  OPT  REG
>>       ------------         -----   -----   ---  ---  ---  ---  ---  ---
>>       P-Preferred-Identity          adr     -    o    -    o    o    -
>>
>>
>>                                            SUB  NOT  REF  INF  UPD  PRA
>>                                            ---  ---  ---  ---  ---  ---
>>                                             o    o    o    -    -    -"
>>

There's that benighted table again. It confuses most people. It must be
replaced.

--
Dean

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Subject: Re: [sipcore] Inconsistency in RFC 3325?
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Seems off topic to discuss 3GPP behaviour here, but just to correct: the 
P-CSCF inserts a P-Asserted-Identity based or not on a 
P-Preferred-Identity, it will remove the latter. In case there is no 
P-Preferred-Identity it will insert the default identity in a 
P-Asserted-Identity.
BR,
/Hans Erik

Dean Willis wrote:
> Paul Kyzivat wrote:
>   
>> I don't know if the proxy usage was intended or not.
>> But neither do I see harm in permitting a proxy to insert the header.
>> Perhaps there is a *really dumb* UA, with a proxy in front of it
>> providing some smarts, before reaching the trusted (and secured) device
>> that acts on the PPI.
>>     
>
> I seem to recall discussing just exactly that use case. IIRC, somebody
> in 3GPP proposed that the P-CSCF should be able to insert the default
> preferred identity if the mobile terminal failed to insert one.
>
>
>   
>> Elwell, John wrote:
>>     
>>> What looks like an inconsistency in RFC 3325 has been identified.
>>>
>>> The table in 9.2 states that PPI can be added ("a") by a proxy:
>>> "      Header field         where   proxy   ACK  BYE  CAN  INV  OPT  REG
>>>       ------------         -----   -----   ---  ---  ---  ---  ---  ---
>>>       P-Preferred-Identity          adr     -    o    -    o    o    -
>>>
>>>
>>>                                            SUB  NOT  REF  INF  UPD  PRA
>>>                                            ---  ---  ---  ---  ---  ---
>>>                                             o    o    o    -    -    -"
>>>
>>>       
>
> There's that benighted table again. It confuses most people. It must be
> replaced.
>
> --
> Dean
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>   

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Subject: Re: [sipcore] Inconsistency in RFC 3325?
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Hi John,

If you cite the complete paragraph:
"6. Hints for Multiple Identities

   If a P-Preferred-Identity header field is present in the message that
   a proxy receives from an entity that it does not trust, the proxy MAY
   use this information as a hint suggesting which of multiple valid
   identities for the authenticated user should be asserted.  If such a
   hint does not correspond to any valid identity known to the proxy for
   that user, the proxy can add a P-Asserted-Identity header of its own
   construction, or it can reject the request (for example, with a 403
   Forbidden).  The proxy MUST remove the user-provided P-Preferred-
   Identity header from any message it forwards."

then it is clear that "The proxy" is not just any proxy, but the "proxy 
that received the P-Preferred-Identity from an entity that it does not 
trust".

It does therefore not say that any proxy shall in all circumstances 
remove P-Preferred-Identity.

Hence, there is no conflict between the table and this text. but makes 
me wonder about the usefullness of these tables as they don't even say 
anything about what UA's are allowed to do or not...

/Hans Erik


Elwell, John wrote:
> What looks like an inconsistency in RFC 3325 has been identified.
>
> The table in 9.2 states that PPI can be added ("a") by a proxy:
> "      Header field         where   proxy   ACK  BYE  CAN  INV  OPT  REG
>       ------------         -----   -----   ---  ---  ---  ---  ---  ---
>       P-Preferred-Identity          adr     -    o    -    o    o    -
>
>
>                                            SUB  NOT  REF  INF  UPD  PRA
>                                            ---  ---  ---  ---  ---  ---
>                                             o    o    o    -    -    -"
>
> However, earlier in 9.2 it states:
> "The P-Preferred-Identity header field is used from a user agent to a
>    trusted proxy to carry the identity the user sending the SIP message
>    wishes to be used for the P-Asserted-Header field value that the
>    trusted element will insert."
> i.e., no mention of being sent from a proxy.
>
> Moreover, in 6 it states:
> "The proxy MUST remove the user-provided P-Preferred-
>    Identity header from any message it forwards."
> i.e., quite explicit about not forwarding, which one might also
> interpret as not being allowed to insert.
>
> So can PPI be added by a proxy, and if so what is its meaning? Has there
> been any earlier discussion on this topic?
>
> John
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>   

From dworley@nortel.com  Wed May 13 14:04:32 2009
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On Mon, 2009-05-04 at 21:06 -0500, Dean Willis wrote:
> On May 4, 2009, at 5:02 PM, Dale Worley wrote:
> >
> > The first sentence is close to what you have written, but as you note,
> > that fact needs to be stated normatively.  The second sentence talks
> > about "Require" in a response, which is far as I can tell, has no
> > defined meaning.

> "Require" in a response has a defined meaning: It means that the  
> referenced extension was applied in formulating the response.  Of  
> course, we have erroneous RFCs (including some I edited, dang it!)  
> that don't do this right.
> 
> To quote RFC 3261  8.2.4 Applying Extensions:
> > Any extensions applied to a non-421 response MUST be listed in a  
> > Require header field included in the response.

Very good point.  And now that you mention it, I have the annoying
feeling that I knew that at some point.  But this passage does mean that
the meaning of Requires in responses is quite a bit different than the
meaning in requests.

There is some subtlety to the phrase "extension applied to a response":
I believe that what is meant is not "the operations defined by the
extension were applied when processing this request" but "this response
cannot be properly understood by a UAC that does not support the
extension [whatever that means]".  That is, there are times when an
extension is *used* by the UAS when processing a request but the UAS
does not put a Requires header in the response.

The obvious example is a registrar that constructs a GRUU for a
registration of a UA, but the UA does not support GRUU, so the registrar
does not add "Require: gruu" to the response, nor put the GRUU
field-parameters in the Contact headers of the response.  (Though
ironically, if the UA ignores unknown field parameters correctly, the UA
would *not* need to support GRUU to handle the GRUU field-parameters
correctly.)

Intriguingly, the only time such a Require is useful is if the UAC needs
to know whether the extension was applied or not, and that fact is not
indicated by any other information in the response.  Do we have any
extensions for which that can happen?

> Note that the following sentence, also from that section, is under  
> debate -- I think it is not justifiable and should allow informational- 
> track RFCs.
> > Of course, the server MUST NOT apply extensions not listed in the  
> > Supported header field in the request. As a result of this, the  
> > Require header field in a response will only ever contain option  
> > tags defined in standards- track RFCs.

I'm not so worried about where the tags are defined, as a UAC can only
see tags in responses that it understands.  The central point is that
the UAC can only receive "Requires: foo" in a response if it has
declared "Supported: foo" in the request, so there can never be the
problem of "receiving a response that contains a Requires tag that it
does not understand".

On Mon, 2009-05-04 at 20:57 -0500, Dean Willis wrote: 
> How about the authorization question: Christer requires the UAS to  
> support and use the Foo: extension, but Christer is not authorized to  
> use the Foo: extension at this UAS. UAS would be willing to process  
> the request for Christer without using Foo:, but SIP provides no  
> mechanism for UAS to tell Christer not to use Foo:.

Although "Require: foo" in responses (or other indication of whether the
foo extension was applied) does tell Christer whether his request to use
foo was acted upon.

On Tue, 2009-05-05 at 07:40 +0200, Christer Holmberg wrote: 
> Currently I wonder if there is a single extension which requires the UAC
> to send Supported:foo in addition to Require:foo to indicate that the
> UAC also supports the extension. The reason is probably that most (all?)
> extensions do require some kind of support from both the UAC and UAS.

Currently, I think the extensions fall into two categories, neither of
which require such sophistication:

1) Extensions where the processing by each end is very similar, so that
we expect that the "UAS implementation" and the "UAC implementation" can
only be done together.  I suspect that session-timers are like this
(although I could be wrong, as I haven't studied session-timers).

2) Extensions where the relationship between the two UAs is inherently
asymmetrical, and the respective roles are fixed with the first request.
gruu and eventlist are like this.

In the long run, there might be cases that fall into neither of these
cases.  In that case, the "obvious" logic is that a Requires demands
that the UAS support the UAS-side of the extension.

What Supported means is less obvious.  Thinking about what has been said
above, the one thing that Supported in a request *must* mean is that the
UAC tolerates any extension-specific features in the response,
equivalently, that Requires will be tolerated in the response.  That
means that Supported in a request says that the UAC supports the
UAC-side of the extension.

Dale



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From: "Dale Worley" <dworley@nortel.com>
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Subject: [sipcore] Comments on draft-roach-sipcore-rfc3265bis-00
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I wrote:

        Notifier migration is an interesting concept and seems like it might
        be very useful.  But how does a subscriber take advantage of it?  The
        text in 4.4.2 assumes that the subscriber will send a new
        out-of-dialog SUBSCRIBE to whatever URI it did originally.  But will
        that obtain an alternative notifier for the resource whose notifier
        just ceased?  The possibilities of SIP routing make that doubtful.
        What we want is for the notifier to give the subscriber a
        globally-routable URI that will reach an alternative notifier, but
        there seems to be no mechanism to carry the URI.
        
I was thinking that a way to carry an alternative subscription address
was in a Refer-To header.  That has the right syntax and connotation.

Dale



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Subject: Re: [sipcore] Question on Require in responses
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Dale Worley wrote:

> On Tue, 2009-05-05 at 07:40 +0200, Christer Holmberg wrote: 
>> Currently I wonder if there is a single extension which requires the UAC
>> to send Supported:foo in addition to Require:foo to indicate that the
>> UAC also supports the extension. The reason is probably that most (all?)
>> extensions do require some kind of support from both the UAC and UAS.
> 
> Currently, I think the extensions fall into two categories, neither of
> which require such sophistication:
> 
> 1) Extensions where the processing by each end is very similar, so that
> we expect that the "UAS implementation" and the "UAC implementation" can
> only be done together.  I suspect that session-timers are like this
> (although I could be wrong, as I haven't studied session-timers).
> 
> 2) Extensions where the relationship between the two UAs is inherently
> asymmetrical, and the respective roles are fixed with the first request.
> gruu and eventlist are like this.
> 
> In the long run, there might be cases that fall into neither of these
> cases.  In that case, the "obvious" logic is that a Requires demands
> that the UAS support the UAS-side of the extension.
> 
> What Supported means is less obvious.  Thinking about what has been said
> above, the one thing that Supported in a request *must* mean is that the
> UAC tolerates any extension-specific features in the response,
> equivalently, that Requires will be tolerated in the response.  That
> means that Supported in a request says that the UAC supports the
> UAC-side of the extension.

Mostly I think Dale is right here. But even in these kinds of cases 
there remains a level of ambiguity. Take for instance eventlist:

If I am subscribing to some event, and include Supported:eventlist, then 
it does at least mean I can handle event lists in notifications I receive.

But does it also mean that I could provide an event list if somebody 
sent a subscription to me?

I guess you could say: try the subscribe and see what I say. That would 
be entirely appropriate if we can conclude that the significance of the 
Supported is bounded to the context in which it is used - the 
subscription in this case.

But Supported in OPTIONS makes hash of that. Including 
Supported:eventlist in an OPTIONS could mean most anything.

IMO, presence of almost anything in an OPTIONS is a very weak promise of 
support. At most I think it means: "I may support this feature in *some* 
context - your mileage may vary."

	Thanks,
	Paul

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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-event-rate-control-00.txt
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Just a few comments:

Section 3.1 is entitled: Use Case for maximizing the rate of notifications

This wording isn't quite right, since "maximizing the rate" means 
"making the rate as large as possible". I would propose "Use case for 
limiting the maximum rate of notifications", or "Use Case for bounding 
the maximum notification rate".

Similarly for section 3.2 (Use Case for minimizing the rate of 
notifications): I would suggest "Use Case for setting a minimum rate for 
notifications" or "Use Case for bounding the minimum rate of notification".

In general its a little hard to read due to frequent switches between 
minimum-interval/maximum-rate and maximum-interval/minimum-rate. It 
might be clearer to discuss everything in terms of intervals, since that 
is what the parameters are. But if that isn't popular people will still 
figure it out.

	Thanks,
	Paul




From fluffy@cisco.com  Sun May 17 09:17:29 2009
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From: Cullen Jennings <fluffy@cisco.com>
To: Hans Erik van Elburg <ietf.hanserik@gmail.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Inconsistency in RFC 3325?
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On May 13, 2009, at 12:58 PM, Hans Erik van Elburg wrote:

> Hi John,
>
> If you cite the complete paragraph:
> "6. Hints for Multiple Identities
>
>  If a P-Preferred-Identity header field is present in the message that
>  a proxy receives from an entity that it does not trust, the proxy MAY
>  use this information as a hint suggesting which of multiple valid
>  identities for the authenticated user should be asserted.  If such a
>  hint does not correspond to any valid identity known to the proxy for
>  that user, the proxy can add a P-Asserted-Identity header of its own
>  construction, or it can reject the request (for example, with a 403
>  Forbidden).  The proxy MUST remove the user-provided P-Preferred-
>  Identity header from any message it forwards."
>
> then it is clear that "The proxy" is not just any proxy, but the  
> "proxy that received the P-Preferred-Identity from an entity that it  
> does not trust".

>
>
> It does therefore not say that any proxy shall in all circumstances  
> remove P-Preferred-Identity.

Agree with Hans.


>
>
> Hence, there is no conflict between the table and this text. but  
> makes me wonder about the usefullness of these tables as they don't  
> even say anything about what UA's are allowed to do or not...
INHO these tables have turned out to be pretty much useless (in this  
RFC, in 3261, and all the rest) but that's just my 2 cents

The idea was to allow a proxy in front of a UA that did not support  
3325 to add the PPI on behalf of the phone. Folks were imaging  
something like an little proxy on a box in my house adding it before  
it sent the call up to the service provider proxy. Clearly that idea  
did not get specified in this draft but the "a" got left in the proxy  
field. Keep that in this case, the proxy in the house trusts the  
service provider proxy but the service provider proxy does not trust  
the proxy in the house.


Cullen < in my individual contributor roll >

>
>
> /Hans Erik
>
>
> Elwell, John wrote:
>> What looks like an inconsistency in RFC 3325 has been identified.
>>
>> The table in 9.2 states that PPI can be added ("a") by a proxy:
>> "      Header field         where   proxy   ACK  BYE  CAN  INV   
>> OPT  REG
>>      ------------         -----   -----   ---  ---  ---  ---  ---   
>> ---
>>      P-Preferred-Identity          adr     -    o    -    o    o    -
>>
>>
>>                                           SUB  NOT  REF  INF  UPD   
>> PRA
>>                                           ---  ---  ---  ---  ---   
>> ---
>>                                            o    o    o    -    -     
>> -"
>>
>> However, earlier in 9.2 it states:
>> "The P-Preferred-Identity header field is used from a user agent to a
>>   trusted proxy to carry the identity the user sending the SIP  
>> message
>>   wishes to be used for the P-Asserted-Header field value that the
>>   trusted element will insert."
>> i.e., no mention of being sent from a proxy.
>>
>> Moreover, in 6 it states:
>> "The proxy MUST remove the user-provided P-Preferred-
>>   Identity header from any message it forwards."
>> i.e., quite explicit about not forwarding, which one might also
>> interpret as not being allowed to insert.
>>
>> So can PPI be added by a proxy, and if so what is its meaning? Has  
>> there
>> been any earlier discussion on this topic?
>>
>> John
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>


From dean.willis@softarmor.com  Sun May 17 14:41:27 2009
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Cc: SIPCORE <sipcore@ietf.org>, Dale Worley <dworley@nortel.com>
Subject: Re: [sipcore] Question on Require in responses
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Paul Kyzivat wrote:
> If I am subscribing to some event, and include Supported:eventlist, then
> it does at least mean I can handle event lists in notifications I receive.
> 
> But does it also mean that I could provide an event list if somebody
> sent a subscription to me?

eventlist as written seems to suffer from modal ambiguity. I suggest it
requires two different option tags to resolve this ambiguity.

--
Dean

From christer.holmberg@ericsson.com  Mon May 18 05:04:57 2009
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Thread-Topic: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Hisham Khartabil" <hisham.khartabil@gmail.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
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Hi,

The requirements talk about negotiating keep-alives towards the next hop
(UAC) and the previous hop (UAS), so in my opinion that only talks about
adjacent nodes.

We can of course argue whether we talk about nodes being adjacent during
the routing of the initial INVITE, or nodes which become adjacent once
the route set has been established :)=20

However, I think the only difference which headers we use for the
negotiation.

If we talk about adjacent nodes during routing of the initial INVITE the
Via based solution works fine.

If we talk about adjacent nodes in the route set I think we need to use
e.g. Record-Route/Contact/Path.

Of course we could use a Via based solution also for the second case,
but then we need to establish the route set before we do the keep-alive
negotiation, to make sure all Vias belong to entities in the route set.
And, that also means a separate SIP transaction (UPDATE or PRACK) for
the negotiation part.

Regards,

Christer=20


> -----Original Message-----
> From: Hisham Khartabil [mailto:hisham.khartabil@gmail.com]=20
> Sent: 11. toukokuuta 2009 2:26
> To: Christer Holmberg
> Cc: Spencer Dawkins; Elwell, John; Tan, Ya Ching (NSN -=20
> DE/Munich); SIPCORE
> Subject: Re: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>=20
> 2009/5/8 Christer Holmberg <christer.holmberg@ericsson.com>:
> >
> > Hi,
> >
> >>>>>[JRE] No, I would not want to do that. My point is that=20
> at the time=20
> >>>>>of the dialog-forming request, there will be no direct transport=20
> >>>>>connection between proxy1 and proxy3, so keep-alive is=20
> not relevant.
> >
> >>>>>However, as a result of the first mid-dialog request,=20
> there will be=20
> >>>>>a direct transport connection between Proxy1 and Proxy3. So the=20
> >>>>>question is whether we want to figure out a way of keeping that=20
> >>>>>alive or not.
> >>>>
> >>>>I think there will be a direct transport connection between proxy1
> > and
> >>>>proxy3 already at the time of the response to the dialog-forming=20
> >>>>request.
> >>>[JRE] I don't understand how that can happen, since the=20
> response will=20
> >>>follow the path of the Via entries, not the path or record-routes.
> >>>Please explain.
> >>
> >>I'm as confused as John is. For example, if the signaling path is=20
> >>using TCP, there would be a TCP connection between
> >>Proxy1 and Proxy2, and a separate TCP connection between
> >>Proxy2 and Proxy3, but this would not cause a=20
> SYN/SYN-ACK/ACK exchange=20
> >>between Proxy1 and Proxy3 - or am I missing something?
> >
> > No, you are not :)
> >
> > The TCP connection between Proxy1 and Proxy3 will be=20
> established when=20
> > something is sent from Proxy1 to Proxy3. That would normally be a=20
> > mid-dialog request, but it COULD of course also be the keep-alive=20
> > message.
> >
> > But, if we don't want the keep-alive message to establish the TCP=20
> > connection we can simply decide that keep-alives can be negotiated=20
> > only between adjacent nodes, and we can keep the Via=20
> parameter based=20
> > solution.
> >
> > Then, it would not be possible to negotiate keep-alives=20
> between Proxy1=20
> > and Proxy3 during the initial request, since Proxy2 would=20
> still insert=20
> > a Via (it would then be possible to negotiate keep-alives between=20
> > Proxy1 and Proxy3 during a mid-dialog request, becuase then=20
> they will=20
> > be adjacent).
>=20
> John's query is not about the possibility of negotiating=20
> keep-alives between Proxy1 and Proxy3 during initial message=20
> exchange. He is asking if we have a requirement to negotiate=20
> keep-alives between
> proxy1 and proxy3 using mid-dialog messages.
>=20
> I think its a valid scenario and we should work on a solution.
>=20
> Hisham
>=20
> >
> > Regards,
> >
> > Christer
> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
>=20

From adam@nostrum.com  Tue May 19 16:24:30 2009
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Cc: SIPCORE <sipcore@ietf.org>, Dale Worley <dworley@nortel.com>
Subject: Re: [sipcore] Question on Require in responses
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[as an individual]

Dean Willis wrote:
> Paul Kyzivat wrote:
>   
>> If I am subscribing to some event, and include Supported:eventlist, then
>> it does at least mean I can handle event lists in notifications I receive.
>>
>> But does it also mean that I could provide an event list if somebody
>> sent a subscription to me?
>>     
>
> eventlist as written seems to suffer from modal ambiguity. I suggest it
> requires two different option tags to resolve this ambiguity.
>   

Almost every use of option tags specific to subscriptions is going to 
suffer from the same "ambiguity" to varying degrees. With 
INVITE-initiated dialogs, the behavior is symmetrical: both participants 
serve the same role in the dialog. With SUBSCRIBE/NOTIFY-initiated 
dialogs, the roles are inherently asymmetrical -- one party is a 
subscriber, while the other is a notifier.

So, if we plan to come up with some "Grand Unified Theory" of option tag 
semantics and publish it as an RFC or part of an RFC, we should probably 
include this as a base principle. I would suggest one of the 
participants in this discussion take the arguments so far and summarize 
them in an internet-draft, taking this into account.

On the other hand, if we're just grousing to pass the time, I'd prefer 
to do that on idle Friday afternoons over beer rather than on the list. ;-)

/a

From dean.willis@softarmor.com  Wed May 20 11:35:04 2009
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Adam Roach wrote:

> On the other hand, if we're just grousing to pass the time, I'd prefer
> to do that on idle Friday afternoons over beer rather than on the list. ;-)

Why would we want to interfere with something important like Friday
afternoon beer time by talking about option tags that everybody has
already gotten wrong?

--
Dean


From adam@nostrum.com  Thu May 21 15:05:26 2009
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-sec-flows-00.txt
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[as chair]

I'd like to thank Brian for taking over editorship of this document, 
which has been mostly complete (pending some syntax issues) for several 
years now. Keeping in mind that this document is an adopted SIPCORE 
working group item, I would encourage everyone to read over it, and 
provide comments to the list. Even a quick note saying something like "I 
read through this, understood it, and think the examples are correct" is 
useful.

Of course, something like "I untarred the examples and independently 
verified them with my own implementation" would be spectacular.

/a


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 Core Working Group of the IETF.
>
>
> 	Title           : Example call flows using Session Initiation Protocol (SIP) security mechanisms
> 	Author(s)       : C. Jennings, et al.
> 	Filename        : draft-ietf-sipcore-sec-flows-00.txt
> 	Pages           : 59
> 	Date            : 2009-05-04
>
> This document shows example call flows demonstrating the use of
> Transport Layer Security (TLS), and Secure/Multipurpose Internet Mail
> Extensions (S/MIME) in Session Initiation Protocol (SIP).  It also
> provides information that helps implementers build interoperable SIP
> software.  To help facilitate interoperability testing, it includes
> certificates used in the example call flows and processes to create
> certificates for testing.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sipcore-sec-flows-00.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
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>   
> ------------------------------------------------------------------------
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> https://www.ietf.org/mailman/listinfo/sipcore
>   


From adam@nostrum.com  Fri May 22 12:12:10 2009
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Subject: [sipcore] IETF 75: Agenda Time Requests
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[as chair]

In keeping with the spirit of our charter, we intend to run SIPCORE 
face-to-face meetings in a very focused fashion. To that end, requests 
for SIPCORE agenda time need to contain the following information:

  1. The name of the draft or drafts under discussion
  2. A list of open issues to be addressed
  3. A pointer to non-trivial mailing list discussion of the draft (the
     subject line of associated threads is sufficient)

Presentations are expected to be focused around the list of open issues, 
not the general contents of or changes to the document.  Tutorial 
content must be limited to explaining the open issues and potential 
resolutions to those issues.

Please note that it will be very difficult to have non-trivial mailing 
list discussion of drafts that are submitted very close to the document 
submission deadlines. We strongly encourage authors who plan to request 
agenda time to publish a revised document well ahead of the deadlines.

Currently, we have requested a total of 3.5 hours of meeting time at the 
upcoming meeting in Stockholm this July. Depending on the total number 
of open issues and the associated volume of mailing list traffic, we may 
reduce this to 2.5 hours or even 1 hour.

If you wish to request agenda time, please send these three pieces of 
information to the SIPCORE chairs on or before June 3rd. Length of 
timeslots will be determined based on the number and complexity of open 
issues in the agenda request. If a new open issue arises after 
submission of a request, please notify the chairs so that we can adjust 
timeslots accordingly.

For the sake of clarity: the discussion will not be limited to the open 
items listed in your request; it can include open items that arise after 
you submit your request. So, if your list of open items is somewhat weak 
but you expect that difficult points may arise between June 3rd and the 
Stockholm meeting, please send in a request indicating that fact.

/a

From dean.willis@softarmor.com  Sun May 24 02:05:21 2009
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Subject: [sipcore] Proto writeup for draft-ietf-sipcore-subnot-etags-02
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SIPCORE chair Adam Roach has asked me to serve as the protocol  
shepherd for the subnot-etags work. Consequently, I've prepared the  
following writeup.


As required by RFC-to-be draft-ietf-proto-wgchair-doc-shepherding,  
this is
the current template for the Document Shepherd Write-Up. Changes are
expected over time. This version is dated September 17, 2008.

     (1.a) Who is the Document Shepherd for this document? Has the
           Document Shepherd personally reviewed this version of the
           document and, in particular, does he or she believe this
           version is ready for forwarding to the IESG for publication?

The document shepherd is Dean Willis. He has reviewed this document
and believes it is ready for publication.


     (1.b) Has the document had adequate review both from key WG members
           and from key non-WG members? Does the Document Shepherd have
           any concerns about the depth or breadth of the reviews that
           have been performed?

The document has been widely reviewed within the working group. Dale
Worley in particular provided some detailed analysis, and Eric
Rescorla also performed useful review leading to subtle changes in the
document. Paul Kyzivat provided further assistance with correcting the  
ABNF
during the final review process, and Adam Roach reviewed the document  
from
perspective of RFC 3265.


     (1.c) Does the Document Shepherd have concerns that the document
           needs more review from a particular or broader perspective,
           e.g., security, operational complexity, someone familiar with
           AAA, internationalization or XML?

No.


     (1.d) Does the Document Shepherd have any specific concerns or
           issues with this document that the Responsible Area Director
           and/or the IESG should be aware of? For example, perhaps he
           or she is uncomfortable with certain parts of the document,  
or
           has concerns whether there really is a need for it. In any
           event, if the WG has discussed those issues and has indicated
           that it still wishes to advance the document, detail those
           concerns here. Has an IPR disclosure related to this document
           been filed? If so, please include a reference to the
           disclosure and summarize the WG discussion and conclusion on
           this issue.

There are no shepherd concerns, and the shepherd is unaware of  
specific IPR
disclosures.

     (1.e) How solid is the WG consensus behind this document? Does it
           represent the strong concurrence of a few individuals, with
           others being silent, or does the WG as a whole understand and
           agree with it?

The document was widely reviewed within the working group between
August, 2007 and October, 2008.

     (1.f) Has anyone threatened an appeal or otherwise indicated  
extreme
           discontent? If so, please summarise the areas of conflict in
           separate email messages to the Responsible Area Director. (It
           should be in a separate email because this questionnaire is
           entered into the ID Tracker.)

There appears to be no extreme discontent or significant conflict
surrounding the draft.

     (1.g) Has the Document Shepherd personally verified that the
           document satisfies all ID nits? (See
           http://www.ietf.org/ID-Checklist.html and
           http://tools.ietf.org/tools/idnits/). Boilerplate checks are
           not enough; this check needs to be thorough. Has the document
           met all formal review criteria it needs to, such as the MIB
           Doctor, media type and URI type reviews?

The shepherd checked idnits, validated BNF using BAP, and did the usual
visual review of the document. There are no formal criteria review for
this document.

     (1.h) Has the document split its references into normative and
           informative? Are there normative references to documents that
           are not ready for advancement or are otherwise in an unclear
           state? If such normative references exist, what is the
           strategy for their completion? Are there normative references
           that are downward references, as described in [RFC3967]? If
           so, list these downward references to support the Area
           Director in the Last Call procedure for them [RFC3967].

References appear to be appropriate.

     (1.i) Has the Document Shepherd verified that the document IANA
           consideration section exists and is consistent with the body
           of the document? If the document specifies protocol
           extensions, are reservations requested in appropriate IANA
           registries? Are the IANA registries clearly identified? If
           the document creates a new registry, does it define the
           proposed initial contents of the registry and an allocation
           procedure for future registrations? Does it suggest a
           reasonable name for the new registry? See [RFC5226]. If the
           document describes an Expert Review process has Shepherd
           conferred with the Responsible Area Director so that the IESG
           can appoint the needed Expert during the IESG Evaluation?

The IANA considerations appear to be appropriate.


     (1.j) Has the Document Shepherd verified that sections of the
           document that are written in a formal language, such as XML
           code, BNF rules, MIB definitions, etc., validate correctly in
           an automated checker?

The document contains only a trivial BNF for a SIP response code and
another very small addition for a message-header. The BNF was verified
using BAP.


     (1.k) The IESG approval announcement includes a Document
           Announcement Write-Up. Please provide such a Document
           Announcement Write-Up? Recent examples can be found in the
           "Action" announcements for approved documents. The approval
           announcement contains the following sections:


Document Writeup


           Technical Summary

The Session Initiation Protocol (SIP) events framework enables
receiving asynchronous notification of various events from other SIP
user agents. This framework defines the procedures for creating,
refreshing and terminating subscriptions, as well as fetching and
periodic polling of resource state. These SIP Events Framework
procedures have a serious deficiency in that they provide no tools to
avoid replaying event notifications that have already been received by
a user agent. This specification defines an extension to SIP events
that allows the subscriber to condition the subscription request to
whether the state has changed since the previous notification was
received. When such a condition is true, either the body of a
resulting event notification or the entire notification message is
suppressed. This "conditioning" of the subscription requests uses the
well-known concept of entity tags (eTags).


           Working Group Summary

This document received extended working group review during a WGLC
period that exceeded one year in duration. One critical clarification
came up in this review period: this specification does not provide
"versioning history" for events; rather it lets a subscriber know
whether the current event state is consistent with the event state as
currently known by that subscriber. The distinction is subtle. For
example, if the subscriber knows of event state "A", but the actual
state has changed to "B" and then back to "A", the subscriber will not
be informed about the B state through this specification.

Following WGLC, the proto review process detected and corrected a
minor error in the ABNF, and further review by Adam Roach detected and
corrected problem related to suppressed NOTIFYs on initial dialogs.



           Document Quality


The Open Mobile Alliance SIMPLE Presence specification references this
document, and has done so for several years. Note that the most recent
OMA document  the shepherd could locate references
draft-ietf-sip-subnot-etags-03.txt rather than the current  
specification.




From christer.holmberg@ericsson.com  Sun May 24 22:47:36 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "SIPCORE" <sipcore@ietf.org>
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Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00(wasRE:Document Adoption)
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=20
Hi,

My proposal would be that we keep the Via based solution.

That allows negotiation as part of of keep-alives in any transaction. A
Record-Route based solution only allowes the negotiation during the
initial dialog transaction, since the route set can't be modified later
during the dialog.

Are people ok with that?

Regards,

Christer




> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: 18. toukokuuta 2009 15:06
> To: Hisham Khartabil
> Cc: SIPCORE
> Subject: Re: [sipcore] Draft submission:=20
> draft-ietf-sipcore-keep-00(wasRE:Document Adoption)
>=20
>=20
> Hi,
>=20
> The requirements talk about negotiating keep-alives towards=20
> the next hop
> (UAC) and the previous hop (UAS), so in my opinion that only=20
> talks about adjacent nodes.
>=20
> We can of course argue whether we talk about nodes being=20
> adjacent during the routing of the initial INVITE, or nodes=20
> which become adjacent once the route set has been established :)=20
>=20
> However, I think the only difference which headers we use for=20
> the negotiation.
>=20
> If we talk about adjacent nodes during routing of the initial=20
> INVITE the Via based solution works fine.
>=20
> If we talk about adjacent nodes in the route set I think we=20
> need to use e.g. Record-Route/Contact/Path.
>=20
> Of course we could use a Via based solution also for the=20
> second case, but then we need to establish the route set=20
> before we do the keep-alive negotiation, to make sure all=20
> Vias belong to entities in the route set.
> And, that also means a separate SIP transaction (UPDATE or=20
> PRACK) for the negotiation part.
>=20
> Regards,
>=20
> Christer=20
>=20
>=20
> > -----Original Message-----
> > From: Hisham Khartabil [mailto:hisham.khartabil@gmail.com]
> > Sent: 11. toukokuuta 2009 2:26
> > To: Christer Holmberg
> > Cc: Spencer Dawkins; Elwell, John; Tan, Ya Ching (NSN - DE/Munich);=20
> > SIPCORE
> > Subject: Re: [sipcore] Draft submission:=20
> > draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
> >=20
> > 2009/5/8 Christer Holmberg <christer.holmberg@ericsson.com>:
> > >
> > > Hi,
> > >
> > >>>>>[JRE] No, I would not want to do that. My point is that
> > at the time
> > >>>>>of the dialog-forming request, there will be no direct=20
> transport=20
> > >>>>>connection between proxy1 and proxy3, so keep-alive is
> > not relevant.
> > >
> > >>>>>However, as a result of the first mid-dialog request,
> > there will be
> > >>>>>a direct transport connection between Proxy1 and=20
> Proxy3. So the=20
> > >>>>>question is whether we want to figure out a way of=20
> keeping that=20
> > >>>>>alive or not.
> > >>>>
> > >>>>I think there will be a direct transport connection=20
> between proxy1
> > > and
> > >>>>proxy3 already at the time of the response to the=20
> dialog-forming=20
> > >>>>request.
> > >>>[JRE] I don't understand how that can happen, since the
> > response will
> > >>>follow the path of the Via entries, not the path or=20
> record-routes.
> > >>>Please explain.
> > >>
> > >>I'm as confused as John is. For example, if the signaling path is=20
> > >>using TCP, there would be a TCP connection between
> > >>Proxy1 and Proxy2, and a separate TCP connection between
> > >>Proxy2 and Proxy3, but this would not cause a
> > SYN/SYN-ACK/ACK exchange
> > >>between Proxy1 and Proxy3 - or am I missing something?
> > >
> > > No, you are not :)
> > >
> > > The TCP connection between Proxy1 and Proxy3 will be
> > established when
> > > something is sent from Proxy1 to Proxy3. That would normally be a=20
> > > mid-dialog request, but it COULD of course also be the keep-alive=20
> > > message.
> > >
> > > But, if we don't want the keep-alive message to establish the TCP=20
> > > connection we can simply decide that keep-alives can be=20
> negotiated=20
> > > only between adjacent nodes, and we can keep the Via
> > parameter based
> > > solution.
> > >
> > > Then, it would not be possible to negotiate keep-alives
> > between Proxy1
> > > and Proxy3 during the initial request, since Proxy2 would
> > still insert
> > > a Via (it would then be possible to negotiate keep-alives between
> > > Proxy1 and Proxy3 during a mid-dialog request, becuase then
> > they will
> > > be adjacent).
> >=20
> > John's query is not about the possibility of negotiating=20
> keep-alives=20
> > between Proxy1 and Proxy3 during initial message exchange. He is=20
> > asking if we have a requirement to negotiate keep-alives between
> > proxy1 and proxy3 using mid-dialog messages.
> >=20
> > I think its a valid scenario and we should work on a solution.
> >=20
> > Hisham
> >=20
> > >
> > > Regards,
> > >
> > > Christer
> > >
> > > _______________________________________________
> > > sipcore mailing list
> > > sipcore@ietf.org
> > > https://www.ietf.org/mailman/listinfo/sipcore
> > >
> >=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

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From: "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
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Subject: Re: [sipcore] Draft submission:draft-ietf-sipcore-keep-00(wasRE:Document Adoption)
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From dean.willis@softarmor.com  Mon May 25 23:23:08 2009
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From: Dean Willis <dean.willis@softarmor.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00(wasRE:Document Adoption)
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I don't believe it is reasonable to change the keepalive set after the  
route-set is determined, and think it's inefficient to do an  
additional-re-INVITE in order to add keepalive to a dialog. So I think  
I prefer something on the record-route operation that lets us use the  
initial dialog formation to establish keepalive.

--
Dean


On May 25, 2009, at 12:49 AM, Christer Holmberg wrote:

>
> Hi,
>
> My proposal would be that we keep the Via based solution.
>
> That allows negotiation as part of of keep-alives in any  
> transaction. A
> Record-Route based solution only allowes the negotiation during the
> initial dialog transaction, since the route set can't be modified  
> later
> during the dialog.
>
> Are people ok with that?
>
> Regards,
>
> Christer
>
>
>
>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
>> Sent: 18. toukokuuta 2009 15:06
>> To: Hisham Khartabil
>> Cc: SIPCORE
>> Subject: Re: [sipcore] Draft submission:
>> draft-ietf-sipcore-keep-00(wasRE:Document Adoption)
>>
>>
>> Hi,
>>
>> The requirements talk about negotiating keep-alives towards
>> the next hop
>> (UAC) and the previous hop (UAS), so in my opinion that only
>> talks about adjacent nodes.
>>
>> We can of course argue whether we talk about nodes being
>> adjacent during the routing of the initial INVITE, or nodes
>> which become adjacent once the route set has been established :)
>>
>> However, I think the only difference which headers we use for
>> the negotiation.
>>
>> If we talk about adjacent nodes during routing of the initial
>> INVITE the Via based solution works fine.
>>
>> If we talk about adjacent nodes in the route set I think we
>> need to use e.g. Record-Route/Contact/Path.
>>
>> Of course we could use a Via based solution also for the
>> second case, but then we need to establish the route set
>> before we do the keep-alive negotiation, to make sure all
>> Vias belong to entities in the route set.
>> And, that also means a separate SIP transaction (UPDATE or
>> PRACK) for the negotiation part.
>>
>> Regards,
>>
>> Christer
>>
>>
>>> -----Original Message-----
>>> From: Hisham Khartabil [mailto:hisham.khartabil@gmail.com]
>>> Sent: 11. toukokuuta 2009 2:26
>>> To: Christer Holmberg
>>> Cc: Spencer Dawkins; Elwell, John; Tan, Ya Ching (NSN - DE/Munich);
>>> SIPCORE
>>> Subject: Re: [sipcore] Draft submission:
>>> draft-ietf-sipcore-keep-00 (wasRE:Document Adoption)
>>>
>>> 2009/5/8 Christer Holmberg <christer.holmberg@ericsson.com>:
>>>>
>>>> Hi,
>>>>
>>>>>>>> [JRE] No, I would not want to do that. My point is that
>>> at the time
>>>>>>>> of the dialog-forming request, there will be no direct
>> transport
>>>>>>>> connection between proxy1 and proxy3, so keep-alive is
>>> not relevant.
>>>>
>>>>>>>> However, as a result of the first mid-dialog request,
>>> there will be
>>>>>>>> a direct transport connection between Proxy1 and
>> Proxy3. So the
>>>>>>>> question is whether we want to figure out a way of
>> keeping that
>>>>>>>> alive or not.
>>>>>>>
>>>>>>> I think there will be a direct transport connection
>> between proxy1
>>>> and
>>>>>>> proxy3 already at the time of the response to the
>> dialog-forming
>>>>>>> request.
>>>>>> [JRE] I don't understand how that can happen, since the
>>> response will
>>>>>> follow the path of the Via entries, not the path or
>> record-routes.
>>>>>> Please explain.
>>>>>
>>>>> I'm as confused as John is. For example, if the signaling path is
>>>>> using TCP, there would be a TCP connection between
>>>>> Proxy1 and Proxy2, and a separate TCP connection between
>>>>> Proxy2 and Proxy3, but this would not cause a
>>> SYN/SYN-ACK/ACK exchange
>>>>> between Proxy1 and Proxy3 - or am I missing something?
>>>>
>>>> No, you are not :)
>>>>
>>>> The TCP connection between Proxy1 and Proxy3 will be
>>> established when
>>>> something is sent from Proxy1 to Proxy3. That would normally be a
>>>> mid-dialog request, but it COULD of course also be the keep-alive
>>>> message.
>>>>
>>>> But, if we don't want the keep-alive message to establish the TCP
>>>> connection we can simply decide that keep-alives can be
>> negotiated
>>>> only between adjacent nodes, and we can keep the Via
>>> parameter based
>>>> solution.
>>>>
>>>> Then, it would not be possible to negotiate keep-alives
>>> between Proxy1
>>>> and Proxy3 during the initial request, since Proxy2 would
>>> still insert
>>>> a Via (it would then be possible to negotiate keep-alives between
>>>> Proxy1 and Proxy3 during a mid-dialog request, becuase then
>>> they will
>>>> be adjacent).
>>>
>>> John's query is not about the possibility of negotiating
>> keep-alives
>>> between Proxy1 and Proxy3 during initial message exchange. He is
>>> asking if we have a requirement to negotiate keep-alives between
>>> proxy1 and proxy3 using mid-dialog messages.
>>>
>>> I think its a valid scenario and we should work on a solution.
>>>
>>> Hisham
>>>
>>>>
>>>> Regards,
>>>>
>>>> Christer
>>>>
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>>>
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From adam@nostrum.com  Tue May 26 13:07:04 2009
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Subject: Re: [sipcore] Comments on draft-roach-sipcore-rfc3265bis-00
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[as an individual]

Many thanks to Dale for reviewing the document. He caught a number of 
legacy issues that existed in 3265 that very much deserve cleanup for a 
bis version, as well as a few issues with the new text.

Several responses inline.


Dale Worley wrote:
> I'm glad we're cleaning up how we define events.  But there are a few
> things left to clean up:
>
> ** Technical
>
> * What is a resource?
>
> It looks like the draft cleans up the question of what is a resource,
> in that section 3.1.2 paragraph 2 states that the request-URI
> identifies the resource in question.  However, given the error in the
> first sentence of that paragraph (discussed below), I'd suggest
> promoting the statement that the request-URI identifies the resource
> to the beginning of the paragraph.
>   

This paragraph is actually copied verbatim from section 3.1.2 of RFC 3265.

> * What is state?
>
> There is a lot of ambiguity over what constitutes the "state" that the
> event system transmits.  These ambiguities bedevil attempts to
> implement generic event system "toolkits" as well as complicating the
> definition of new event packages.  (In the call-completion committee
> we've run into trouble with this.)
>
> In theory, we want the resource to have a certain set of state
> information, and that state information progresses through a series of
> different values over time.  Each time the state changes, every
> subscriber to that state would be notified of the new state value.  We
> further envision that a subscriber might supply a "filter" which is
> applied to every state value; then only the output of the filter would
> be sent to the subscriber, and the subscriber would be notified when
> the filtered state changed.
>
> Layered onto this nice, clean system are a set of issues, most of
> which we haven't addressed systematically:
>
> - If we want to be able to send state deltas, there needs to be a
>   defined way of taking the difference between two state values.
>   (This is well-described in the draft.)
>
> - In most event package definitions, we include the version number and
>   the partial/full indication *in* the state, rather than treating
>   them as *attributes of* the state value contained in a NOTIFY.
>   Since most definitions require the version number to start counting
>   from 0 independently for each subscription, the version number is
>   clearly not part of the state itself.
>   


What you're identifying is a general philosophy that the types of state 
deltas that make sense are better described by the event packages 
themselves, instead of some generic state-delta mechanism. I'm intrigued 
by your proposal to add versioning information to the base 
specification, and would be interested in whether you have a 
backwards-compatible proposal for doing so.


> - We haven't been clear about the difference between "state" (which
>   possesses values over time) and "events" (which happen instantly).
>   Generally, the event packages have been consistent in using state
>   rather than events.  (E.g., the dialog package's <state> element has
>   an attribute telling how a dialog entered its current state, but
>   that value is actually state, since it is "historical information",
>   a new subscriber could properly see the same value.)
>
>   State-orientation makes implementation easier, although it has some
>   unspoken implications; e.g., since notifications may be missed in
>   some way, the subscriber may see fewer state transitions than did
>   thenotifier, and so the event package must be organized so that this
>   is not a problem.  But we still have a tendency to use the word
>   "event" too much, even in the name of the mechanism.  For instance,
>   in the abstract:
>
>       The purpose of this extension is to provide an extensible framework
>       by which SIP nodes can request notification from remote nodes
>       indicating that certain events have occurred.
>
>   Similarly, 3.2.1 says:
>
>       NOTIFY bodies are expected to provide additional details about the
>       nature of the event which has occurred and the resultant resource
>       state.
>
>   The essential point is that the only *events* that can be reported
>   are the changes in an ongoing *state value*.  Though we've been
>   consistent about that fact, it's not clearly stated anywhere.
>   

This reflects the evolution of the document from its original mechanism 
(which was, indeed, more event oriented than state oriented). I have the 
same model in mind as you do -- that of conveying the state of the 
resource. I agree that cleaning up all of the "event" language to 
instead talk about states and state transitions would indeed be an 
improvement. I'll add this to my list of changes for the next revision.

> - Many event packages contain data which is expected to change
>   frequently, but we do not expect each change to cause sending of the
>   state value.  For example, the dialog event package contains
>   "duration" elements telling how long each dialog has been
>   established, in seconds.  Clearly, this value changes every second.
>   Similarly, the "reg" event package contains the CSeq value of the
>   most-recent registration of a contact.  Changes in either of these
>   values are not intended to cause notification, but that is not
>   stated in the package definitions, nor is the possibility made clear
>   in the event framework.
>   

Semantically, these are the same thing as hardwired filters on the 
state. Do you have any concrete suggestions for treatment of this in the 
base spec? Or should it simply be mentioned that some event packages 
inherently contain the semantic equivalent of hard-coded filters?

> - In principle, the state should not be affected by who has subscribed
>   to received notifications for it.  But there are circumstances where
>   we want to be aware of the set of subscribers, and to be able to
>   present differing information to them.  The first job can probably
>   be done (in principle) by examining the "winfo" state, but I don't
>   know of a solution to the second.
>   

You're venturing into the realm of things like the presence common 
policy framework, which I think cover this very well. Do you have a 
proposal for how to incorporate this into the base specification?

> * Expiration time
>
> There is a problem regarding how the expiration time of a subscription
> is indicated that I don't think has ever been resolved.  It probably
> isn't a problem in practice, but it would help if we agreed on a
> solution.
>
> The difficulty regards how the subscriber should track the
> end-of-subscription time that is established when a re-subscribe is
> done.  The subscriber will receive a response to its SUBSCRIBE, and
> also the NOTIFY that it stimulates.  It may also receive other NOTIFYs
> stimulated by state changes.  All of these carry expiration times, or
> rather, the length of time from the present that the subscription will
> last.  The various NOTIFYs can be put in order by their CSeqs, and the
> latest one clearly supersedes the others.  But the relationship
> between a received NOTIFY and a success response to a SUBSCRIBE is
> harder to discern, as the subscriber does not know the order in which
> they were sent, and so does not know which represents the most recent
> state of the notifier.
>
> I think this problem is obviated by the fact that the notifier must
> always send a NOTIFY after receiving any SUBSCRIBE, so that the
> subscriber can in pratice disregard SUBSCRIBE responses, assuming that
> the network is reliable.
>
> But if we take this interpretation, then the wording in the draft
> should be adjusted in a number of places where it gives equal respect
> to the expiration time reported in a SUBSCRIBE response and that in a
> NOTIFY.
>   

Agreed. This version of the document puts far more emphasis on NOTIFY as 
"the second leg in the three-way handshake." As such, I'm inclined to 
deprecate the conveyance of any information in SUBSCRIBE responses, and 
use NOTIFY requests exclusively. I'll do a pass of the document for 
expires use in this regard.

> If we do not assume that the subscriber can reliably get a NOTIFY from
> every SUBSCRIBE, then working out a reliable solution is more
> difficult, due to the race condition.
>
> There are related problems regarding terminology.  The "length of a
> subscription" can mean two things, the "time from present" that it
> expires, or the absolute endpoint of the subscription.  The draft says
> in various circumstances that a subscription cannot be lengthened
> without being specific what this means.  As far as I can tell: when
> processing a SUBSCRIBE, a notifier must update the subscription
> endpoint of the subscription so that the time from present is no
> larger than what the subscriber requested (which is likely later than
> the previous endpoint); a notifier may unilaterally bring sooner the
> endpoint of a subscription so long as it notifies the subscriber; a
> subscriber's refresh duration request is unconstrained (in that it may
> require bringing forward the endpoint of the subscription, as well as
> postponing it).
>   

The rules are intended to operate exactly like expiration handling for 
REGISTER. Is there any language you think is particularly ambiguous?

> * Notifier migration
>
> Notifier migration is an interesting concept and seems like it might
> be very useful.

Note that the preponderance of this text was copied verbatim from 
RFC3265, section 3.3.5.

> But how does a subscriber take advantage of it?  The
> text in 4.4.2 assumes that the subscriber will send a new
> out-of-dialog SUBSCRIBE to whatever URI it did originally.  But will
> that obtain an alternative notifier for the resource whose notifier
> just ceased?  The possibilities of SIP routing make that doubtful.
>   

Not really. There are a couple of architectures we had in mind when we 
designed this mechanism:

   1. The proxy/registrar has a "default" destination that it will send
      SUBSCRIBE requests to when a user has no registered UAs. This
      "default" destination could serve proper state for the
      corresponding resource (or even act as a simple "parking lot",
      from which subscribers will be migrated once the watched user has
      a UA registered).

   2. The notifier is part of a cluster of notifiers that can all
      service requests corresponding to the user. In order to take the
      notifier offline, it informs its clustering mechanism that it is
      no longer accepting new traffic, and then sends NOTIFY messages to
      each subscribed UA for the purpose of moving them to a notifier
      that is not going offline. Once all such subscriptions are shed,
      the node can be taken out of service. A similar mechanism can be
      used to rebalance subscriptions after adding notifiers to a cluster.

In any case, it's not the subscriber taking advantage of this mechanism; 
it's the notifier.

> What we want is for the notifier to give the subscriber a
> globally-routable URI that will reach an alternative notifier, but
> there seems to be no mechanism to carry the URI.
>   

At the risk of triggering your "REFER means everything" rant, I think 
this kind of operation would best be served by using REFER to initiate 
the equivalent of a "blind transfer" of the subscription. This looks a 
lot like figure 1 of
draft-ietf-sipping-cc-transfer-12 (except with "SUBSCRIBE" instead of 
"INVITE").

> ** Editorial
>
> * section 2
>
> The definition of "subscription" doesn't mention that the existence of
> a subscription means that the notifier will send state updates to the
> subscriber.
>   

Good point.

> * section 3.1
>
> This paragraph should say something like "The SUBSCRIBE method is used
> to requrest the creation of a subscription which will send current
> state and state updates from a remote node."
>   

Yes, that's more explicit.

> * section 3.1.2
>
>    The Request URI of a SUBSCRIBE request, most importantly, contains
>    enough information to route the request to the appropriate entity per
>    the request routing procedures outlined in SIP [RFC3261].
>
> This is not necessarily true -- the routing is also affected by any
> Route headers in the request.  In sipX, we use that feature so that we
> can present to the notifier a resource name (request-URI) that need
> not be the same as the SIP URI of the notifier.
>   

Would "The Request URI of a SUBSCRIBE request, in conjunction its Route 
header fields, contains enough information..." satisfy your concern?

> * section 4.1.3
>
>    NOTIFY requests MUST contain "Subscription-State" header fields which
>    indicate the status of the subscription.
>
> This is an instance where plurals should be made singular for clarity:
>
>    A NOTIFY request MUST contain a "Subscription-State" header field
>    which indicates the status of the subscription.
>   

Agreed.

> * section 4.1.3
>
>    giveup:  The subscription has been terminated because the notifier
>       could not obtain authorization in a timely fashion.  If a "retry-
>       after" parameter is also present, the client SHOULD wait at least
>       the number of seconds specified by that parameter before
>       attempting to re-subscribe; otherwise, the client MAY retry
>       immediately, but will likely get put back into pending state.
>
> The last clause would read better as:
>
>       the client MAY retry immediately, although the new subscription will
>       likely go into the pending state.
>
> The subscription doesn't really "get put back" into pending, since it
> is now in terminated.
>   

That's a reasonable clarification.

> * section 4.2.1.2
>
>    Note that a NOTIFY message is always sent immediately after any 200-
>    class response to a SUBSCRIBE request, regardless of whether the
>    subscription has already been authorized.
>
> Perhaps it is better to say:
>
>    Note that a NOTIFY message is always sent immediately after any
>    200-class response to a SUBSCRIBE request, regardless of the state
>    of the created subscription (e.g., pending).
>   

Okay.

> * section 4.2.1.4.  Refreshing of Subscriptions
>
>    When a notifier receives a subscription refresh, assuming that the
>    subscriber is still authorized, the notifier updates the expiration
>    time for subscription.  As with the initial subscription, the server
>    MAY shorten the amount of time until expiration, but MUST NOT
>    increase it.
>
> Here is one of the places where the discussion of "shortening
> expiration time" is ambiguous.  Reading it literally, it means that
> the notifier cannot increase the absolute expiration time *relative to
> what it was before the SUBSCRIBE arrived*, as "the amount of time
> until expiration" is an attribute of the subscription.  Of course,
> what is meant, the notifier sets a new expiration time which may be no
> longer-from-present that the duration requested in the SUBSCRIBE.
>   

I'll see if I can adapt some language from 3261 regarding REGISTER 
handling to make this clearer.

> * section 4.4.1
>
>    NOTIFY requests are matched to such SUBSCRIBE requests if they
>    contain the same "Call-ID", a "To" header field "tag" parameter which
>    matches the "From" header field "tag" parameter of the SUBSCRIBE, and
>    the same "Event" header field.  Rules for comparisons of the "Event"
>    header fields are described in Section 8.2.1.
>
> This is awkward.  How about:
>
>    NOTIFY and SUBSCRIBE requests are within the same subscription
>    usage if they contain the same "Call-ID", a "To" header field "tag"
>    parameter which matches the "From" header field "tag" parameter of
>    the SUBSCRIBE, and the same "Event" header field.  Rules for
>    comparisons of the "Event" header fields are described in Section
>    8.2.1.
>   

Actually, I think this would be best served by recasting the association 
as it relates to 3261 dialogs, and then adding on the "Event" matching. 
I'll see if I can come up with something reasonable that conveys enough 
information.

> * section 4.4.2.
>
> Notifier migration is an interesting concept.  But note that the text
> is incorrect:
>
>    Upon receipt of this NOTIFY message, the subscriber SHOULD attempt to
>    re-subscribe (as described in the preceding sections).  Note that
>    this subscription is established on a new dialog, and does not re-use
>    the route set from the previous subscription dialog.
>
> It's not correct to say the subscriber is "re-subscribing" (in the
> sense of the draft).  

I can see your point. I'll change it to something like "initate a new 
subscription to the resource."

> There is also the technical question of what
> request-URI the subscriber should use to re-subscribe.  (see above)
>   

I think the intention was always that the subscriber would use the same 
Request-URI (and, if relevant, Route) that was used to set up the 
subscription that has just terminated. I'll clarify this point.

> * section 4.5.1
>
>       Because GRUUs are guaranteed to route to a *** a specific device, this
>
> Contains duplicate word "a".
>   

I'll fix that.

> * section 5.1
>
>    When designing an event package using the methods described in this
>    document for event notification, it is important to consider: is SIP
>    an appropriate mechanism for the problem set?  Is SIP being selected
>
> The initial "is" of the clauses are not consistently capitalized.
>   

This issue isn't as simple as it appears at first. See, for example, 
<http://en.wikipedia.org/wiki/Colon_(punctuation)#Use_of_capitals>. If I 
changed it to a capital letter, there's a good chance that I'd get a 
request to change it back from someone who has a different sense of 
aesthetics than  you do (and I believe that there's a strong argument to 
be made that the form in the current document is the more common form 
for both American and British English). I've had to mediate these kinds 
of disputes before, and was kind of hoping to avoid them this time -- in 
particular, I don't want to have to add another paragraph like the final 
paragraph of section 1.2 to clarify how I have chosen to treat colons 
and capitalization.

So, unless there is an argument that can be made regarding clarity, I'm 
not accepting comments on the topic of typographical conventions. :)

> * section 5.3
>
>    Any event package that supports delta changes to states MUST include
>    a version number that increases by exactly one for each NOTIFY
>    transaction in a subscription.
>
> And it must include a full/partial state indicator.
>   

Good point.

> * section 5.4
>
>    Event packages are not required to reiterate any of the behavior
>    described in this document, although they may choose to do so for
>    clarity or emphasis.  In general, though, such packages are expected
>    to describe only the behavior that extends or modifies the behavior
>    described in this document.
>
> This reads strangely, as it suggests that an event package description
> might contradict the RFC.
>   

We certainly can't prevent it. :)

Examples of where this happens include filtering and throttling, which 
do actually change things like when NOTIFY messages are sent. 3265bis 
(and 3265) say this happens when the state changes. The throttling 
document says this happens when the state changes and/or when other 
criteria are met.

> * sectin 5.4.3
>
>    It is expected that most, but not all, event packages will define
>    syntax and semantics for SUBSCRIBE method bodies;
>
> This reads strangely, since only one defined event package has
> SUBSCRIBE bodies.
>   

I think we optimistically believed that people would define filters 
along with their event packages, instead of choosing to bolt them on 
later. But your observation that this didn't play out the way we 
expected is a good one, and I'll revise the text accordingly.

> * section 5.4.7
>
>    and whether state information is complete or deltas for
>    notifications
>
> This is awkwardly phrased.  Perhaps end the sentence before this
> clause and continue:
>
>     It also includes whether state deltas can be used, and if so, how they
>     are interpreted by the subscriber to update its record of the state;
>     see Section 5.3.
>   

Okay.

> * section 7
>
> The IANA Considerations section does not name the new SUBSCRIBE and
> NOTIFY methods, which is needs to do in order to register the methods.
> (RFC 3265 doesn't do this either, although the IANA database
> entries for SUBSCRIBE and NOTIFY references RFC 3265.)
>   

I have an open action item to figure out how the IESG wants us to handle 
IANA registration sections of bis RFCs. But this is a good catch, and 
one that I don't recall being pointed out before. I'll make a note of 
it, with action pending on the final guidance I get regarding IANA 
sections in bis docments.

/a

From dean.willis@softarmor.com  Tue May 26 17:44:56 2009
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Subject: [sipcore] Fwd: [Sip] Last Call: draft-ietf-sip-connect-reuse (Connection Reuse in the Session Initiation Protocol (SIP)) to Proposed Standard
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Please follow this discussion on the SIP list.

Begin forwarded message:

> From: The IESG <iesg-secretary@ietf.org>
> Date: May 26, 2009 9:34:01 AM CDT
> To: IETF-Announce <ietf-announce@ietf.org>
> Cc: sip@ietf.org
> Subject: [Sip] Last Call: draft-ietf-sip-connect-reuse (Connection  
> Reuse in the Session Initiation Protocol (SIP)) to Proposed Standard
> Reply-To: ietf@ietf.org
>
> The IESG has received a request from the Session Initiation Protocol  
> WG
> (sip) to consider the following document:
>
> - 'Connection Reuse in the Session Initiation Protocol (SIP) '
>   <draft-ietf-sip-connect-reuse-13.txt> as a Proposed Standard
>
> The IESG plans to make a decision in the next few weeks, and solicits
> final comments on this action.  Please send substantive comments to  
> the
> ietf@ietf.org mailing lists by 2009-06-09. Exceptionally,
> comments may be sent to iesg@ietf.org instead. In either case, please
> retain the beginning of the Subject line to allow automated sorting.
>
> The file can be obtained via
> http://www.ietf.org/internet-drafts/draft-ietf-sip-connect- 
> reuse-13.txt
>
>
> IESG discussion can be tracked via
> https://datatracker.ietf.org/public/pidtracker.cgi?command=view_id&dTag=10776&rfc_flag=0
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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 dean.willis@softarmor.com  Tue May 26 17:45:44 2009
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Subject: [sipcore] Fwd: [Sip] Draft ietf-sip-xcapevent revised, many open questions
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Begin forwarded message:

> From: Dean Willis <dean.willis@softarmor.com>
> Date: May 26, 2009 7:35:21 PM CDT
> To: "sip@ietf.org List" <sip@ietf.org>
> Cc: Robert Sparks <rjs@nostrum.com>
> Subject: [Sip] Draft ietf-sip-xcapevent revised, many open questions
>
>
> I have submitted a revised version of the XCAP Event draft based on  
> feedback from Lisa Dusseault and that incorporates a vast amount of  
> editing by Jean Mahoney, a technical writer who volunteered to help  
> clean things up after Lisa pushed the draft back from IESG review.
>
>
>
> See:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-xcapevent-05.txt
>
> and since tehre are a lot of changes, the side-by-side diff is  
> really helpful:
>
> http://tools.ietf.org/rfcdiff?url2=draft-ietf-sip-xcapevent-05
>
> There are a number of questions open in the existing document.
>
>
>
> 1)  What's the default notification mode supposed to be? "xml- 
> patching" is listed as the default and fail-safe in some text, but  
> "no-patching" is also described as the singular "must implement"  
> mode. It seems to me that if a mode is the default if no mode is  
> specified, then that mode had better be the default.
> This question is a primary source of confusion in Section 4.3, and  
> I'd appreciate some feedback. This whole section got a lot of edits,  
> you may want to usea difftool.
>
>
> 2) Aggregating definition
>
> Our current text has this definition; is it right?
>
>   Aggregating:  An XCAP client can update only a single XCAP Component
>      at a time using HTTP.  However, a notifier may be able to
>      aggregate a series of these modifications into a single
>      notification using XML-Patch- Ops semantics encoded in the XCAP-
>      Diff format.
>
>
>
> 3) SUBSCRIBE bodies
>
> Section 4.4 currently says;
>
>   The SUBSCRIBE request MAY contain an Accept header field.  If no  
> such
>   header field is present, it has a default value of "application/
>   xcap-diff+xml".  If the header field is present, it MUST include
>   "application/xcap-diff+xml", and MAY include any other types.
>
> This doesn't sound right to me.
>
>
>
> 4) Error recovery
>
> Section 4.7 says:
>
>   While the "aggregate" mode uses bandwidth most efficiently, it
>   introduces other challenges.  The initial synchronization might fail
>   with rapidly changing resources, because the "aggregate" mode
>   messages might not include the full version-history of a document  
> and
>   the base XCAP protocol does not support version-history retrievals  
> of
>   documents.  When new documents are created in subscribed collections
>   and the notifier is aggregating patches, the same issue can occur.
>   In a corner case, the notifier may not be able to provide patches
>   with the XML-Patch-Ops [RFC5261] semantics.  Therefore, if the
>   notifier has to temporarily disable diff generation and send only  
> the
>   URI references of some changed documents to the subscriber, it MUST
>   continue with the "xcap-patching" mode afterwards for these
>   resources, if the initial subscription also started with the "xcap-
>   patching" mode.  Recovery from this error condition therefore means
>   that the subscriber must refresh to a "known good" state by
>   downloading the current document if it has lost track of the  
> patching
>   operations as indicated by mismatching ETags.,
>
> (note the accidental extra comma here at the end, I just noticed it,  
> and it is my fault).
>
> Anyhow, does the "recovery" text make sense?
>
>
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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 gonzalo.camarillo@ericsson.com  Wed May 27 04:12:12 2009
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Subject: [sipcore] Status of the 199 draft
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Folks,

as you know, the 199 draft, whose current version can be found below, 
was WGLCed in the SIP WG a few months ago:

http://www.ietf.org/mail-archive/web/sip/current/msg25440.html

http://tools.ietf.org/id/draft-ietf-sipcore-199-00.txt

We intend to request its publication as soon as the author of the draft 
addresses all the comments received so far. Therefore, if you want to 
send comments on this draft, do it fast.

Cheers,

Gonzalo
SIPCORE co-chair

From gonzalo.camarillo@ericsson.com  Wed May 27 05:30:40 2009
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Folks,

this draft was WGLCed in SIPPING some time ago:

http://www.ietf.org/mail-archive/web/sipping/current/msg16089.html

We intend to request its publication as soon as all the dedicated 
reviewers give their OK. So, if you have comments on this draft, send 
them to the list as soon as possible.

Thanks,

Gonzalo
SIPCORE co-chair


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 Core Working Group of the IETF.
> 
> 
> 	Title           : Scaling Requirements for Presence in SIP/SIMPLE
> 	Author(s)       : A. Houri, et al.
> 	Filename        : draft-ietf-sipcore-presence-scaling-requirements-00.txt
> 	Pages           : 9
> 	Date            : 2009-04-20
> 
> The document provides a set of requirements for enabling interdomain
> scaling in presence for SIP/SIMPLE.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sipcore-presence-scaling-requirements-00.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
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> 
> Below is the data which will enable a MIME compliant mail reader
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> 
> 
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> _______________________________________________
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> https://www.ietf.org/mailman/listinfo/sipcore


From gonzalo.camarillo@ericsson.com  Wed May 27 05:47:27 2009
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00	(wasRE:Document Adoption)
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Hi,

 > UA1 -> Proxy1 - >Proxy2 -> Proxy3 -> UA2

John wrote:

>> [JRE] No, I would not want to do that. My point is that at 
>> the time of the dialog-forming request, there will be no 
>> direct transport connection between proxy1 and proxy3, so 
>> keep-alive is not relevant. However, as a result of the first 
>> mid-dialog request, there will be a direct transport 
>> connection between Proxy1 and Proxy3. So the question is 
>> whether we want to figure out a way of keeping that alive or not.

Christer answered:

> I think there will be a direct transport connection between proxy1 and
> proxy3 already at the time of the response to the dialog-forming
> request. So, you can start sending keep-alives at that point. You don't
> need to wait for the first mid-dialog request.

the response will not trigger such a connection because even though 
Proxy2 did not record routed, he inserted itself in the Via, which is 
used to route responses.

On a different note, the negotiation you are trying to perform here has 
somewhat similar requirements as the one specified in RFC 3486. You may 
want to have a look at that spec at some point.

Cheers,

Gonzalo


From gonzalo.camarillo@ericsson.com  Wed May 27 05:49:51 2009
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Subject: Re: [sipcore] Draft submission: draft-ietf-sipcore-keep-00	(wasRE:Document Adoption)
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Hi,

> We can of course argue whether we talk about nodes being adjacent during
> the routing of the initial INVITE, or nodes which become adjacent once
> the route set has been established :) 

well, that is actually the main point of the discussion. We need to 
understand whether or not we have that requirement. For example, in RFC 
3486, the UAC can negotiate SigComp compression with a node that will be 
adjacent to it during the dialog but that was not in the initial 
request. Do we have the same requirement here or not?

Cheers,

Gonzalo

From salvatore.loreto@ericsson.com  Wed May 27 06:51:28 2009
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Cc: Avshalom Houri <AVSHALOM@il.ibm.com>, Jonathan Rosenberg <jdrosen@cisco.com>, SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] [Fwd: Re: I-D Action:draft-ietf-sipcore-subnot-etags-02.txt]
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Hi there,

I have carefully read the last version of the draft and I don't have any 
comments to add.

I believe it is ready for publication!

/Sal

From wangxiangjun@huawei.com  Sat May 30 18:24:31 2009
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From adam@nostrum.com  Sat May 30 18:43:16 2009
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Date: Sat, 30 May 2009 20:44:44 -0500
From: Adam Roach <adam@nostrum.com>
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To: wang xiangjun <wangxiangjun@huawei.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Forking and draft-ietf-sipcore-subnot-etags
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[as an individual]

As with normal RFC 3265 processing, after the initial SUBSCRIBE, the 
subscriber is responsible for maintaining the multiple subscriptions 
separately, including the handling of etags.

I don't think we cover the exact case you're talking about. I suspect 
you're pretty much stuck with getting the body for all but one of the 
forks, and suppressing the subsequent NOTIFY messages. If this is a 
problem for a specific network, there are a host of solutions that can 
be engineered through the use of GRUUs.

/a

wang xiangjun wrote:
> Hello Adam,
>    I have read the draft and your clarification about the forking 
> situation which is a good job, but I still have a problem about it as 
> below:
>    If a subscriber sends a SUBSCRIBE request and the request is forked 
> to two notifiers by a proxy which establishes two dialogs. The 
> subscriber will receive two SIP-Etag from the two notifiers 
> respectively, is it right? Then the subscription terminates and after 
> a while the subscriber wants to resume a subscription, the problem 
> is how does the subscriber deals with the Suppress-If-Match header 
> field? Does it include one of the SIP-Etag value or both of them?
>  
> Best Regards,
> wangxiangjun
> 华为技术有限公司 huawei_logo
> 地址：深圳市龙岗区坂田华为基地 邮编：518129
> http://www.huawei.com
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>     ----- Original Message -----
>     *From:* Adam Roach <mailto:adam@nostrum.com>
>     *To:* SIPCORE <mailto:sipcore@ietf.org>
>     *Sent:* Thursday, April 23, 2009 12:13 AM
>     *Subject:* [sipcore] Forking and draft-ietf-sipcore-subnot-etags
>
>     [as an individual participant]
>
>     You've probably noticed a new version of the subnot-etags document
>     just
>     came out. This version incorporates some changes to resolve a
>     last-minute technical issue with the way this document interacts with
>     request forking. (There's also an ABNF change, but there's a separate
>     thread going on that topic).
>
>     In previous versions of the document, the language allowed complete
>     suppression of NOTIFY messages except when there were "reportable
>     changes in the NOTIFY header". This criteria was somewhat
>     ambiguous in
>     the general case of establishing a new subscription (the
>     transition of
>     subscription state to "active" technically qualifies as a "reportable
>     change in the NOTIFY header", but implementors were left to figure
>     this
>     out independently). To complicate matters, there were at least two
>     passages in the document that strongly suggested that there may be
>     circumstances in which the first NOTIFY in a subscription would be
>     completely suppressed, which causes some serious issues with forking.
>
>     The complete suppression of NOTIFY messages was also explicit for
>     resource state polling, which gives rise to similar issues.
>
>     The issue with suppressing the first NOTIFY message in a dialog is
>     that
>     a forked SUBSCRIBE message can land on several notifiers, each of
>     whom
>     may well accept the subscription. However, due to SIP forking
>     processing, only one final response to the SUBSCRIBE will arrive
>     at the
>     subscriber.
>
>     Under the 3265 model, this is resolved by the use of an immediate
>     NOTIFY
>     in the reverse direction -- the subscriber learns about all the
>     resulting subscriptions because it receives a NOTIFY message for each
>     one. Removing this aspect of subscription establishment basically
>     re-creates a variant of the HERFP problem, except much worse:
>     instead of
>     failing to learn about various request failures, the subscriber
>     fails to
>     learn about various successful dialog establishments.
>
>     The most recent version of the document addresses this by:
>
>        1. Cleaning up the passages that implied suppression of the initial
>           NOTIFY message in a dialog
>
>        2. Making explicit the previously implicit behavior of not
>           suppressing initial NOTIFY message in a dialog
>
>        3. Changing poll behavior to require empty NOTIFY messages
>     instead of
>           completely suppressed NOTIFY messages
>
>     Only the third change is an actual change in behavior. The first
>     two are
>     simply making previously implied behavior explicit.
>
>     I realize that this is a bit of deep RFC 3265 arcana, but please
>     weigh
>     in on the changes so we can move the document over the finish line
>     soon.
>     Thanks!
>
>     /a
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