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Date: Thu, 01 Apr 2010 09:36:06 +0200
From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

A good mechanism to discovery UA capabilities is essential to provide 
meaningful service:
So I am in favor of pursuing the work,
in particular the possibility to use a single OPTIONS request to 
retrieve the capabilities from ALL UAS(s) behind a forking proxy.

I agree with Christer, that as discussed in the ml and during the f2f 
meeting, the 3xx solution already provide this feature
even if with the limitations Paul has highlighted (e.g. honoring R-D is 
optional by the proxy).

I do think, it would be opportune document it in a BCP to 
document/clarify the ambiguous stuff, and I can help to co-edit it if we 
decide to write one.

Explore the possibility to use other mechanisms (e.g. presence, 
something similar to XEP-0115) and define it would be also interesting,
but IMO it would require a major effort and more cycles and I am not so 
sure there are enough people willing to accomplish such effort.

cheers
Sal



On 3/30/10 9:05 PM, Adam Roach wrote:
> [as chair]
>
> One of the topics of discussion at the recent face-to-face meeting in 
> Anaheim was the SIP OPTIONS method. We have had significant discussion 
> of some specific technical topics on the SIPCORE mailing list, such 
> that all participants should have a reasonable overview of the scope 
> that any related work may entail. Participants may also benefit from 
> reading through the minutes of the corresponding conversation in Anaheim:
>
> http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTIONS%20Method
>
> Rather than continue discussion of the specific technical topics 
> related to OPTIONS, the chairs would encourage the working group to 
> engage in a discussion around whether we should, as a group, work on 
> addressing some of the issues that have been identified.
>
> In particular, we would direct the group to consider the nature of 
> such work:
>
>     * Are we proposing to introduce new features, or fix existing bugs?
>     * If we are adding new features: is OPTIONS the best way to
>       introduce these features, or would other mechanisms (presence,
>       callee-caps) produce superior results?
>     * If we are fixing bugs: are the bugs theoretical, or can we cite
>       actual interoperability issues in the field?
>     * If there are identified interoperability issues: is the level of
>       effort involved in fixing them commensurate with the impact of
>       the problems?
>
> Finally, keep in mind that work does not happen unless participants 
> are willing to invest time. If you speak out in favor of pursuing this 
> work, please indicate the level of time commitment that you will 
> personally invest in bringing such work to a successful conclusion 
> (e.g., will author drafts; will perform dedicated reviews; etc).
>
> Thanks.
>
> /a

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Hi,<br>
<br>
A good mechanism to discovery UA capabilities is essential to provide
meaningful service:<br>
So I am in favor of pursuing the work, <br>
in particular the possibility to use a single OPTIONS request to
retrieve the capabilities from ALL UAS(s) behind a forking proxy. <br>
<br>
I agree with Christer, that as discussed in the ml and during the f2f
meeting, the 3xx solution already provide this feature<br>
even if with the limitations Paul has highlighted (e.g. honoring R-D is
optional by the proxy).<br>
<br>
I do think, it would be opportune document it in a BCP to
document/clarify the ambiguous stuff, and I can help to co-edit it if
we decide to write one.<br>
<br>
Explore the possibility to use other mechanisms (e.g. presence,
something similar to XEP-0115) and define it would be also interesting,<br>
but IMO it would require a major effort and more cycles and I am not so
sure there are enough people willing to accomplish such effort.<br>
<br>
cheers<br>
Sal<br>
<br>
<br>
<br>
On 3/30/10 9:05 PM, Adam Roach wrote:
<blockquote cite="mid:4BB24B81.1050006@nostrum.com" type="cite">
  <meta http-equiv="content-type"
 content="text/html; charset=ISO-8859-1">
[as chair]<br>
  <br>
One of the topics of discussion at the recent face-to-face meeting in
Anaheim was the SIP OPTIONS method. We have had significant discussion
of some specific technical topics on the SIPCORE mailing list, such
that all participants should have a reasonable overview of the scope
that any related work may entail. Participants may also benefit from
reading through the minutes of the corresponding conversation in
Anaheim:<br>
  <br>
  <a moz-do-not-send="true" class="moz-txt-link-freetext"
 href="http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTIONS%20Method">http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTIONS%20Method</a><br>
  <br>
Rather than continue discussion of the specific technical topics
related to OPTIONS, the chairs would encourage the working group to
engage in a discussion around whether we should, as a group, work on
addressing some of the issues that have been identified.<br>
  <br>
In particular, we would direct the group to consider the nature of such
work:<br>
  <ul>
    <li>Are we proposing to introduce new features, or fix existing
bugs?</li>
    <li>If we are adding new features: is OPTIONS the best way to
introduce these features, or would other mechanisms (presence,
callee-caps) produce superior results?<br>
    </li>
    <li>If we are fixing bugs: are the bugs theoretical, or can we cite
actual interoperability issues in the field?</li>
    <li>If there are identified interoperability issues: is the level
of
effort involved in fixing them commensurate with the impact of the
problems?</li>
  </ul>
Finally, keep in mind that work does not happen unless participants are
willing to invest time. If you speak out in favor of pursuing this
work, please indicate the level of time commitment that you will
personally invest in bringing such work to a successful conclusion
(e.g., will author drafts; will perform dedicated reviews; etc).<br>
  <br>
Thanks.<br>
  <br>
/a<br>
</blockquote>
</body>
</html>

--------------090109050401030203060309--

From adam@nostrum.com  Thu Apr  1 06:11:10 2010
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On 4/1/10 02:36, Apr 1, Salvatore Loreto wrote:
> Hi,
>
> A good mechanism to discovery UA capabilities is essential to provide 
> meaningful service:

[as an individual]

If this is true -- if it really is *essential* -- then we need something 
better than the "maybe it works, maybe it doesn't" level of 
functionality that we get from R-D. If discovery of UA capabilities is 
*essential*, then we need something that survives forking in all 
circumstances, like SUBSCRIBE/NOTIFY or something else that sends a 
request in both directions.

I suspect it's more in the "nice to have" category.

/a

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On 4/1/10 02:36, Apr 1, Salvatore Loreto wrote:
<blockquote cite="mid:4BB44CE6.2070402@ericsson.com" type="cite">
  <meta content="text/html; charset=ISO-8859-1"
 http-equiv="Content-Type">
Hi,<br>
  <br>
A good mechanism to discovery UA capabilities is essential to provide
meaningful service:<br>
</blockquote>
<br>
[as an individual]<br>
<br>
If this is true -- if it really is *essential* -- then we need
something better than the "maybe it works, maybe it doesn't" level of
functionality that we get from R-D. If discovery of UA capabilities is
*essential*, then we need something that survives forking in all
circumstances, like SUBSCRIBE/NOTIFY or something else that sends a
request in both directions.<br>
<br>
I suspect it's more in the "nice to have" category.<br>
<br>
/a<br>
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</html>

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From keith.drage@alcatel-lucent.com  Thu Apr  1 06:24:21 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Adam Roach <adam@nostrum.com>
Date: Thu, 1 Apr 2010 15:24:27 +0200
Thread-Topic: [sipcore] Consensus Call: SIP Table 2
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Consensus Call: SIP Table 2
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There are essentially two potential parts of the work here:

-	Decide on the future documentation structure
-	work out how to deal with interactions with legacy specifications.

SIPCORE is being asked only to decide on the first of these at the moment.

As Adam says, most of the existing header field usage is fully defined. The=
 problem exists with certain header fields where it appears a somewhat arbi=
trary analysis was given to the entries, and therefore the entries appear i=
nconsistent.=20

An example of this is in the RFC 3329 header fields, where I personally wou=
ld expect the semantics of these headers to be dealt with before one looks =
at the method. That means they should appear in all methods, but you will s=
ee from RFC 3329 that the headers are not valid in PRACK.

With a set of documentation rules to follow, there are no legacy issues if =
a new header field is defined. It follows one of the proforma rules, or cre=
ates a new one.

There is a legacy issue if a new method is created, but it is a long while =
since we created a new method. Yes we hit the documentation issues with the=
 update to INFO and event packages, but for existing headers we can just ca=
rry on with table 2 and 3 if we really need to. Or at that point we can cho=
ose to create a new specification that updates those RFCs that do define he=
ader fields. I would note however that that may well result in some technic=
al change, due to perceived inconsistencies in the current definitions, as =
in the example indicated above.

At the moment I do not favour the existing table method of documentation.

I do favour a set of proforma rules that specifiers can choose from.

A couple of examples of these proforma rules would be something like:

"The use of the header field is optional in all requests that are expected =
to be exchanged end-to-end between UAs. Implementations of this extension M=
UST support reception of this header field in such requests."

"The use of the header field is optional in all requests that initiate a di=
alog, or are used outside of a dialog being created. Implementations of thi=
s extension MUST support reception of this header field in such requests."

I suspect we could deal with most of the current header field usage with ab=
out 10 or so such proforma rules, and I do not expect new header fields to =
create a substantial set of new ones.

These need to specify both use (mandatory/optional) and if it is not otherw=
ise clear, support (mandatory/optional). The latter is missing from the exi=
sting tables.

regards

Keith

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: Thursday, April 01, 2010 7:45 AM
> To: Adam Roach
> Cc: SIPCORE
> Subject: Re: [sipcore] Consensus Call: SIP Table 2
>=20
>=20
> Hi,=20
>=20
> >Just to clarify: both alternatives will leave whatever is=20
> currently unclear as it is....
> >	=09
> >
> >[as individual]
> >=09
> >In truth, I don't think anything is particularly unclear --=20
> it's just=20
> >not documented the same way. The documents that didn't=20
> expand table 2 while adding new header fields all have prose=20
> that describes when the header field can appear. (Yes, I know=20
> this, because I looked at every one of them when I did the=20
> table 2 expansion that appears in the slides).
> >
> >So, you are saying that by reading the specs we would be=20
> able to complete the table you had put toghether?
> >=09
> >	[as chair]
> >=09
> >But, yes, the consensus call only deals with actions as they=20
> apply to=20
> >future documents -- such as RFC3265bis -- and does not=20
> address actions that would affect RFCs or any other documents=20
> that have already cleared IETF review.
> >=09
> >And, couldn't alternative 1 result in lots of "defined semantics" -=20
> >especially if we define a new method and may have to take a=20
> large number of header fields into consideration? Maybe=20
> you'll even end up with something that looks like Table 2...
> >
> >
> >[as individual again]
> >=09
> >I don't think so. As Keith has pointed out, statements like "both=20
> >header fields defined in this document can appear only in=20
> methods that=20
> >create new dialogs" fill out entire swaths of the table all=20
> at once. They also have the pleasantly future-proof quality=20
> that we don't need to say anything specifically about the=20
> header fields when someone creates a new method. A simple=20
> evaluation of the new method's properties against the header=20
> fields' normative text makes the answer brilliantly clear.
>=20
> Sure. That is fine when you define new headers, and when you=20
> define new methods and need to cover those new headers.
>=20
> But, when you define a new method, you still also need to=20
> consider all "old" existing header fields - for which we=20
> unfortunately do not have statements as those suggested by Keith.
>=20
> Take 3265bis as an example, and assume you remove table 2.=20
> The text will of course describe the usage of header fields=20
> directly associated with SUB/NOT, eg. Allow-Events. But, how=20
> will you deal with all the other header fields that we have=20
> defined over the years? I am sure there are header fields=20
> that we do NOT need to cover, because they are associated=20
> with specific mechanisms (e.g. RSeq and RAck). But, there is=20
> a huge number of more "generic" header fields that somehow=20
> need to be covered.
>=20
> Anyway, there is only one way to figure out: by doing the work :)
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> ________________________________
>=20
> 		From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> 		Sent: 30. maaliskuuta 2010 21:51
> 		To: SIPCORE
> 		Subject: [sipcore] Consensus Call: SIP Table 2
> 	=09
> 	=09
> 		[as chair]
> 	=09
> 		Last week, at the face-to-face meeting in=20
> Anaheim, the SIPCORE participants in attendance had a=20
> discussion regarding the future of Tables 2 and 3 in RFC=20
> 3261. A summary of the conversation, along with links to the=20
> presentations and an audio recording of the conversation, can=20
> be found in the minutes:
> 	=09
> 	=09
> http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP
> %20Table%202%20/%203
> 	=09
> 		During that conversation, the chairs polled=20
> those present for support between two options:
> 	=09
>=20
> 		1.	Produce an artifact stating that=20
> documents defining new header fields and methods define=20
> semantics in prose, and will no longer extend Table 2; or
> 		=09
> 		=09
> 		2.	Produce an RFC that defines new=20
> procedures for adding elements to Table 2 as new header=20
> fields and methods are defined=20
>=20
> 		Among those present, there was strong support=20
> for option 1. If you wish to raise an objection to this=20
> direction, please do so on the SIPCORE mailing list before April 15th.
> 	=09
> 		Thank you.
> 	=09
> 		/a
> 	=09
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From HKaplan@acmepacket.com  Thu Apr  1 15:26:22 2010
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To: Christer Holmberg <christer.holmberg@ericsson.com>, 'Adam Roach' <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Thu, 1 Apr 2010 18:25:11 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi Christer,
Inline...

________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Christer Holmberg
Sent: Wednesday, March 31, 2010 1:56 AM
To: 'Adam Roach'; SIPCORE
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?

Hi,

Adam, thanks for sending out the the e-mail!

First, a clarification regarding "new features":

When we say "new features", I assume (based on Hadriel's comment in the mee=
ting) it means the possibility to, with a single OPTIONS request, retrieve =
the capabilities from ALL UAS(s) behind a forking proxy - instead of only o=
ne UAS (whoever sends the 200 OK to the forking proxy first).

Yeah what I meant was that in my opinion your slides had really two separat=
e classes of topics: the first were "bugs", either in incorrectly or under-=
specific RFC cases/text.  Effectively things we would put into bugzilla, an=
d possibly incorporate into any errata/bis someday.  The second was the "I =
want this request to go to all contacts for the AoR and get all their respo=
nses", which is not a bug of the current specs.


The message I got from the meeting, and also previous e-mail discussions, w=
as that the "OPTIONS 3xx" solution already provideds this feature, and that=
 it can be used (without defining new protocol elements etc).

Does it actually solve it?  Sorry I wasn'f following the discussion of it o=
n the mailing list, but...What if there is more than one proxy in the path =
- which one is supposed to return 3xx?
What if the first proxy has two routes for the AoR, and one of those points=
 to another proxy that also has two (different) routes for the AoR?
Do you only want 3xx for Registered contact URI's?  What if the proxy doesn=
't know whether they're Registered contact URI's vs. static routes?

So, in my opinion the "adding" word is a little missleading. Instead I thin=
k the question is whether we want to DOCUMENT that existing feature.

Second, regarding bugs and interoperability issues in the field. Personally=
 I am not sure of any major ones at the moment, but that is probably becaus=
e OPTIONS haven't been very much deployed in the first place (it has been u=
sed for some keep-alive/ping purposes, but in those cases it really doesn't=
 matter what comes in the response etc).

Funny enough that's the only area I've ever actually seen OPTIONS interop p=
roblems - the response code.  We seem to need to map the response code to s=
omething different a lot.  Not sure why though.

Now, however, at least I am starting to get questions regarding the usage o=
f OPTIONS, so I guess we need to ask ourselves whether we think the bugs WI=
LL create interoperability issues.

And, the issue you raised regarding using the re-INVITE to re-create dialog=
s ("[sipcore] Mea Culpa: Dialog Necromancy in RFC 3261") also affects OPTIO=
NS, eventhough you want to keep them separate (which is ok).

Obviously I am in favor of pursuing the work, and will be willing to author=
 drafts :)

Regards,

Christer




________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Adam Roach
Sent: 30. maaliskuuta 2010 22:06
To: SIPCORE
Subject: [sipcore] SIP OPTIONS: Shall we work on this?
[as chair]

One of the topics of discussion at the recent face-to-face meeting in Anahe=
im was the SIP OPTIONS method. We have had significant discussion of some s=
pecific technical topics on the SIPCORE mailing list, such that all partici=
pants should have a reasonable overview of the scope that any related work =
may entail. Participants may also benefit from reading through the minutes =
of the corresponding conversation in Anaheim:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTI=
ONS%20Method

Rather than continue discussion of the specific technical topics related to=
 OPTIONS, the chairs would encourage the working group to engage in a discu=
ssion around whether we should, as a group, work on addressing some of the =
issues that have been identified.

In particular, we would direct the group to consider the nature of such wor=
k:

 *   Are we proposing to introduce new features, or fix existing bugs?
 *   If we are adding new features: is OPTIONS the best way to introduce th=
ese features, or would other mechanisms (presence, callee-caps) produce sup=
erior results?
 *   If we are fixing bugs: are the bugs theoretical, or can we cite actual=
 interoperability issues in the field?
 *   If there are identified interoperability issues: is the level of effor=
t involved in fixing them commensurate with the impact of the problems?
Finally, keep in mind that work does not happen unless participants are wil=
ling to invest time. If you speak out in favor of pursuing this work, pleas=
e indicate the level of time commitment that you will personally invest in =
bringing such work to a successful conclusion (e.g., will author drafts; wi=
ll perform dedicated reviews; etc).

Thanks.

/a

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<div class=3DSection1>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Inline&#8230;<o:p></o:p></span></font>=
</p>

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

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
color=3Dblack face=3D"Times New Roman"><span style=3D'font-size:12.0pt;colo=
r:windowtext'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 color=3Dblack face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;color:windowtext;font-weight:b=
old'>From:</span></font></b><font
size=3D2 color=3Dblack face=3DTahoma><span style=3D'font-size:10.0pt;font-f=
amily:Tahoma;
color:windowtext'> sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.or=
g] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b><st1:PersonName w:st=3D"=
on">Christer
 Holmberg</st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March 31, 2=
010
1:56 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> '<st1:PersonName w:st=3D=
"on">Adam
 Roach</st1:PersonName>'; <st1:PersonName w:st=3D"on">SIPCORE</st1:PersonNa=
me><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [sipcore] SIP
OPTIONS: Shall we work on this?</span></font><font color=3Dblack><span
style=3D'color:windowtext'><o:p></o:p></span></font></p>

</div>

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

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Hi,</span></font><o:p></o:p></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Adam, thanks for sending out the the
e-mail!</span></font><o:p></o:p></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:blue'>First,&nbsp;a clarification regarding
&quot;new features&quot;: </span></font><o:p></o:p></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:blue'>When&nbsp;we say&nbsp;&quot;new
features&quot;, I assume (based on&nbsp;Hadriel's comment in the
meeting)&nbsp;it means the possibility to, with a single OPTIONS request,&n=
bsp;retrieve
the capabilities from ALL UAS(s) behind a forking proxy -&nbsp;instead of o=
nly
one UAS&nbsp;(whoever sends the 200 OK to the forking proxy first).</span><=
/font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 color=3Dnavy face=3D"Times New Roman"><=
span
style=3D'font-size:12.0pt;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Yeah what I meant was that in my opini=
on your
slides had really two separate classes of topics: the first were &#8220;bug=
s&#8221;,
either in incorrectly or under-specific RFC cases/text. &nbsp;Effectively
things we would put into bugzilla, and possibly incorporate into any errata=
/bis
someday.&nbsp; The second was the &#8220;I want this request to go to all
contacts for the AoR and get all their responses&#8221;, which is not a bug=
 of
the current specs.&nbsp; <o:p></o:p></span></font></p>

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

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

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:blue'>The message I got from the meeting, an=
d
also previous e-mail discussions, was that the &quot;OPTIONS 3xx&quot; solu=
tion
already provideds this feature, and that it can be used (without defining n=
ew
protocol elements etc).</span></font><o:p></o:p></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Does it actually solve it? &nbsp;Sorry=
 I
wasn&#8217;f following the discussion of it on the mailing list, but&#8230;=
What
if there is more than one proxy in the path &#8211; which one is supposed t=
o
return 3xx?&nbsp; <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>What if the first proxy has two routes=
 for
the AoR, and one of those points to another proxy that also has two (differ=
ent)
routes for the AoR? &nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Do you only want 3xx for Registered
contact URI&#8217;s? &nbsp;What if the proxy doesn&#8217;t know whether the=
y&#8217;re
Registered contact URI&#8217;s vs. static routes?&nbsp; <o:p></o:p></span><=
/font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:blue'>So, in my opinion the &quot;adding&quo=
t;
word is a little missleading. Instead I think the question is whether we wa=
nt
to DOCUMENT that existing feature.</span></font><o:p></o:p></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Second, regarding bugs and
interoperability issues in the field. Personally I am not sure of any major
ones at the moment, but that is probably because OPTIONS haven't been very =
much
deployed&nbsp;in the first place (it has been used for some keep-alive/ping
purposes, but in those cases it really doesn't matter what comes in the
response etc). </span></font><font size=3D2 color=3Dnavy face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p></o:p></span><=
/font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Funny enough that&#8217;s the only are=
a I&#8217;ve
ever actually seen OPTIONS interop problems &#8211; the response code.&nbsp=
; We
seem to need to map the response code to something different a lot.&nbsp; N=
ot
sure why though. <o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Now, however, at least I am starting t=
o
get questions regarding the usage of OPTIONS, so I guess we need to ask
ourselves whether we think the bugs WILL create interoperability issues.</s=
pan></font><o:p></o:p></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:blue'>And, the issue you raised
regarding&nbsp;using the re-INVITE to re-create dialogs (&quot;[sipcore] Me=
a
Culpa: Dialog Necromancy in RFC 3261&quot;) also affects OPTIONS, eventhoug=
h
you want to keep them separate (which is ok).</span></font><o:p></o:p></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Obviously I am in favor of pursuing th=
e
work, and will be willing to author drafts :)</span></font><o:p></o:p></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Regards,</span></font><o:p></o:p></p>

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

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

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

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

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

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

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
color=3Dblack face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 color=
=3Dblack
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma;font-weigh=
t:bold'>From:</span></font></b><font
size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b><st1:PersonName w:st=3D"=
on">Adam
 Roach</st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 30. maaliskuuta 2010 2=
2:06<br>
<b><span style=3D'font-weight:bold'>To:</span></b> <st1:PersonName w:st=3D"=
on">SIPCORE</st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [sipcore] SIP OPTIO=
NS:
Shall we work on this?</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'>[as chair]<br>
<br>
One of the topics of discussion at the recent face-to-face meeting in <st1:=
City
w:st=3D"on"><st1:place w:st=3D"on">Anaheim</st1:place></st1:City> was the S=
IP
OPTIONS method. We have had significant discussion of some specific technic=
al
topics on the <st1:PersonName w:st=3D"on">SIPCORE</st1:PersonName> mailing =
list,
such that all participants should have a reasonable overview of the scope t=
hat
any related work may entail. Participants may also benefit from reading thr=
ough
the minutes of the corresponding conversation in <st1:City w:st=3D"on"><st1=
:place
 w:st=3D"on">Anaheim</st1:place></st1:City>:<br>
<br>
<a
href=3D"http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SI=
P%20OPTIONS%20Method">http://www.ietf.org/proceedings/10mar/minutes/sipcore=
.html#The%20SIP%20OPTIONS%20Method</a><br>
<br>
Rather than continue discussion of the specific technical topics related to
OPTIONS, the chairs would encourage the working group to engage in a discus=
sion
around whether we should, as a group, work on addressing some of the issues
that have been identified.<br>
<br>
In particular, we would direct the group to consider the nature of such wor=
k:<o:p></o:p></span></font></p>

<ul type=3Ddisc>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D3 color=3Dblack face=3D"Times Ne=
w Roman"><span
     style=3D'font-size:12.0pt'>Are we proposing to introduce new features,=
 or
     fix existing bugs? <o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D3 color=3Dblack face=3D"Times Ne=
w Roman"><span
     style=3D'font-size:12.0pt'>If we are adding new features: is OPTIONS t=
he
     best way to introduce these features, or would other mechanisms (prese=
nce,
     callee-caps) produce superior results?<o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D3 color=3Dblack face=3D"Times Ne=
w Roman"><span
     style=3D'font-size:12.0pt'>If we are fixing bugs: are the bugs theoret=
ical,
     or can we cite actual interoperability issues in the field? <o:p></o:p=
></span></font></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D3 color=3Dblack face=3D"Times Ne=
w Roman"><span
     style=3D'font-size:12.0pt'>If there are identified interoperability is=
sues:
     is the level of effort involved in fixing them commensurate with the
     impact of the problems? <o:p></o:p></span></font></li>
</ul>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'>Finally, keep in mind that work does not happen =
unless
participants are willing to invest time. If you speak out in favor of pursu=
ing
this work, please indicate the level of time commitment that you will
personally invest in bringing such work to a successful conclusion (e.g., w=
ill
author drafts; will perform dedicated reviews; etc).<br>
<br>
Thanks.<br>
<br>
/a<o:p></o:p></span></font></p>

</div>

</div>

</body>

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From HKaplan@acmepacket.com  Thu Apr  1 15:32:56 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 1 Apr 2010 18:33:19 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi Salvatore,
When you say "is essential to providing meaningful service" in what way do =
you mean that?
Do you mean literally that you need any UAC anywhere to be able to discover=
 the capabilities of any arbitrary UAS it might want to communicate with in=
 the World using OPTIONS?
Or do you mean within a provider's domain, having an App Server figure out =
what each registered UA endpoint supports and putting that in some database=
 is very useful? (which I could see 3GPP doing)
Because if it's the latter, then creating an options-tag for rfc3841 and pu=
tting it in Proxy-Require isn't that bad an idea - proxy-require only has d=
eployment issues when you want to use it across proxies outside of your adm=
inistrative control.

-hadriel

________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Salvatore Loreto
Sent: Thursday, April 01, 2010 3:36 AM
To: sipcore@ietf.org
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?

Hi,

A good mechanism to discovery UA capabilities is essential to provide meani=
ngful service:
So I am in favor of pursuing the work,
in particular the possibility to use a single OPTIONS request to retrieve t=
he capabilities from ALL UAS(s) behind a forking proxy.

I agree with Christer, that as discussed in the ml and during the f2f meeti=
ng, the 3xx solution already provide this feature
even if with the limitations Paul has highlighted (e.g. honoring R-D is opt=
ional by the proxy).

I do think, it would be opportune document it in a BCP to document/clarify =
the ambiguous stuff, and I can help to co-edit it if we decide to write one=
.

Explore the possibility to use other mechanisms (e.g. presence, something s=
imilar to XEP-0115) and define it would be also interesting,
but IMO it would require a major effort and more cycles and I am not so sur=
e there are enough people willing to accomplish such effort.

cheers
Sal



On 3/30/10 9:05 PM, Adam Roach wrote:
[as chair]

One of the topics of discussion at the recent face-to-face meeting in Anahe=
im was the SIP OPTIONS method. We have had significant discussion of some s=
pecific technical topics on the SIPCORE mailing list, such that all partici=
pants should have a reasonable overview of the scope that any related work =
may entail. Participants may also benefit from reading through the minutes =
of the corresponding conversation in Anaheim:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTI=
ONS%20Method

Rather than continue discussion of the specific technical topics related to=
 OPTIONS, the chairs would encourage the working group to engage in a discu=
ssion around whether we should, as a group, work on addressing some of the =
issues that have been identified.

In particular, we would direct the group to consider the nature of such wor=
k:

 *   Are we proposing to introduce new features, or fix existing bugs?
 *   If we are adding new features: is OPTIONS the best way to introduce th=
ese features, or would other mechanisms (presence, callee-caps) produce sup=
erior results?
 *   If we are fixing bugs: are the bugs theoretical, or can we cite actual=
 interoperability issues in the field?
 *   If there are identified interoperability issues: is the level of effor=
t involved in fixing them commensurate with the impact of the problems?
Finally, keep in mind that work does not happen unless participants are wil=
ling to invest time. If you speak out in favor of pursuing this work, pleas=
e indicate the level of time commitment that you will personally invest in =
bringing such work to a successful conclusion (e.g., will author drafts; wi=
ll perform dedicated reviews; etc).

Thanks.

/a

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<div class=3DSection1>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>When you say &#8220;is essential to pr=
oviding
meaningful service&#8221; in what way do you mean that?<o:p></o:p></span></=
font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Do you mean literally that you need an=
y UAC
anywhere to be able to discover the capabilities of any arbitrary UAS it mi=
ght
want to communicate with in the World using OPTIONS?<o:p></o:p></span></fon=
t></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Or do you mean within a provider&#8217=
;s
domain, having an App Server figure out what each registered UA endpoint
supports and putting that in some database is very useful? (which I could s=
ee
3GPP doing)<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Because if it&#8217;s the latter, then
creating an options-tag for rfc3841 and putting it in Proxy-Require isn&#82=
17;t
that bad an idea &#8211; proxy-require only has deployment issues when you =
want
to use it across proxies outside of your administrative control.<o:p></o:p>=
</span></font></p>

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

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

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

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
color=3Dblack face=3D"Times New Roman"><span style=3D'font-size:12.0pt;colo=
r:windowtext'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 color=3Dblack face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;color:windowtext;font-weight:b=
old'>From:</span></font></b><font
size=3D2 color=3Dblack face=3DTahoma><span style=3D'font-size:10.0pt;font-f=
amily:Tahoma;
color:windowtext'> sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.or=
g] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b><st1:PersonName w:st=3D"=
on">Salvatore
 Loreto</st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, April 01, 20=
10
3:36 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> sipcore@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [sipcore] SIP
OPTIONS: Shall we work on this?</span></font><font color=3Dblack><span
style=3D'color:windowtext'><o:p></o:p></span></font></p>

</div>

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

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'>Hi,<br>
<br>
A good mechanism to discovery UA capabilities is essential to provide
meaningful service:<br>
So I am in favor of pursuing the work, <br>
in particular the possibility to use a single OPTIONS request to retrieve t=
he
capabilities from ALL UAS(s) behind a forking proxy. <br>
<br>
I agree with Christer, that as discussed in the ml and during the f2f meeti=
ng,
the 3xx solution already provide this feature<br>
even if with the limitations Paul has highlighted (e.g. honoring R-D is
optional by the proxy).<br>
<br>
I do think, it would be opportune document it in a BCP to document/clarify =
the
ambiguous stuff, and I can help to co-edit it if we decide to write one.<br=
>
<br>
Explore the possibility to use other mechanisms (e.g. presence, something
similar to XEP-0115) and define it would be also interesting,<br>
but IMO it would require a major effort and more cycles and I am not so sur=
e
there are enough people willing to accomplish such effort.<br>
<br>
cheers<br>
Sal<br>
<br>
<br>
<br>
On 3/30/10 9:05 PM, <st1:PersonName w:st=3D"on">Adam Roach</st1:PersonName>
wrote: <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'>[as chair]<br>
<br>
One of the topics of discussion at the recent face-to-face meeting in <st1:=
City
w:st=3D"on"><st1:place w:st=3D"on">Anaheim</st1:place></st1:City> was the S=
IP
OPTIONS method. We have had significant discussion of some specific technic=
al
topics on the <st1:PersonName w:st=3D"on">SIPCORE</st1:PersonName> mailing =
list,
such that all participants should have a reasonable overview of the scope t=
hat
any related work may entail. Participants may also benefit from reading thr=
ough
the minutes of the corresponding conversation in <st1:City w:st=3D"on"><st1=
:place
 w:st=3D"on">Anaheim</st1:place></st1:City>:<br>
<br>
<a
href=3D"http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SI=
P%20OPTIONS%20Method"
moz-do-not-send=3Dtrue>http://www.ietf.org/proceedings/10mar/minutes/sipcor=
e.html#The%20SIP%20OPTIONS%20Method</a><br>
<br>
Rather than continue discussion of the specific technical topics related to
OPTIONS, the chairs would encourage the working group to engage in a discus=
sion
around whether we should, as a group, work on addressing some of the issues
that have been identified.<br>
<br>
In particular, we would direct the group to consider the nature of such wor=
k:<o:p></o:p></span></font></p>

<ul type=3Ddisc>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D3 color=3Dblack face=3D"Times Ne=
w Roman"><span
     style=3D'font-size:12.0pt'>Are we proposing to introduce new features,=
 or
     fix existing bugs?<o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D3 color=3Dblack face=3D"Times Ne=
w Roman"><span
     style=3D'font-size:12.0pt'>If we are adding new features: is OPTIONS t=
he
     best way to introduce these features, or would other mechanisms (prese=
nce,
     callee-caps) produce superior results?<o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D3 color=3Dblack face=3D"Times Ne=
w Roman"><span
     style=3D'font-size:12.0pt'>If we are fixing bugs: are the bugs theoret=
ical,
     or can we cite actual interoperability issues in the field?<o:p></o:p>=
</span></font></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D3 color=3Dblack face=3D"Times Ne=
w Roman"><span
     style=3D'font-size:12.0pt'>If there are identified interoperability is=
sues:
     is the level of effort involved in fixing them commensurate with the
     impact of the problems?<o:p></o:p></span></font></li>
</ul>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'>Finally, keep in mind that work does not happen =
unless
participants are willing to invest time. If you speak out in favor of pursu=
ing
this work, please indicate the level of time commitment that you will
personally invest in bringing such work to a successful conclusion (e.g., w=
ill
author drafts; will perform dedicated reviews; etc).<br>
<br>
Thanks.<br>
<br>
/a<o:p></o:p></span></font></p>

</div>

</div>

</body>

</html>

--_000_430FC6BDED356B4C8498F634416644A91A79E93040mail_--

From pkyzivat@cisco.com  Thu Apr  1 16:24:38 2010
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Comments inline. I trimmed out the irrelevant.
I guessed which was Hadriel - apparently the only distinction was color, 
at least for me.

Hadriel Kaplan wrote:
> Hi Christer,
> 
> Inline…
> 
> ------------------------------------------------------------------------
> *From:* sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] *On 
> Behalf Of *Christer Holmberg
> *Sent:* Wednesday, March 31, 2010 1:56 AM
> *To:* 'Adam Roach'; SIPCORE
> *Subject:* Re: [sipcore] SIP OPTIONS: Shall we work on this?
...
> The message I got from the meeting, and also previous e-mail 
> discussions, was that the "OPTIONS 3xx" solution already provideds this 
> feature, and that it can be used (without defining new protocol elements 
> etc).
> 
> Does it actually solve it?  Sorry I wasn’f following the discussion of 
> it on the mailing list, but…What if there is more than one proxy in the 
> path – which one is supposed to return 3xx? 
> 
> What if the first proxy has two routes for the AoR, and one of those 
> points to another proxy that also has two (different) routes for the AoR?  

That should all be handled by recursion on 3xx response processing.
The first proxy would return 3xx with its choices. One of those would 
lead to the 2nd proxy, that then would return its own 3xx.

> Do you only want 3xx for Registered contact URI’s?  What if the proxy 
> doesn’t know whether they’re Registered contact URI’s vs. static routes? 

IMO this is one of the big problems with the IMS approach to services. 
They aren't registered, and aren't anything that the S-CSCF could (or 
would) return as 3xx alternatives. So the caller will be limited in what 
options are visible to it.

	Thanks,
	Paul

From peter.musgrave@magorcorp.com  Fri Apr  2 03:05:30 2010
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From: Peter Musgrave <peter.musgrave@magorcorp.com>
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi all,

I agree with the assertion (Hadrials?) that there are two things here:
bugs in OPTIONS and a need to query multiple UAs.

I have not seen wide use of OPTIONS and have not encountered these
bugs, In general I favour fixing them but I am not sure if this is
more important that other work which could be done.

As for assessing multiple UA capabilities - I agree this is useful. Is
OPTIONS the only way forward here? Could I not use e.g. gruu-reg-event
and then send specifically targeted OPTIONS to each gruu? Is requiring
gruu support in UAs viewed as 'setting the bar too high?' This fails
to pick up forking due to dial plan entries - is that a requirement?

Peter Musgrave

From brett@broadsoft.com  Fri Apr  2 04:59:33 2010
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Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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> If discovery of UA capabilities is *essential*, then we need=20
> something that survives forking in all circumstances, like=20
> SUBSCRIBE/NOTIFY or something else that sends a request in=20
> both directions.

Concerning forking, I agree that SUBSCRIBE/NOTIFY usage is better than OPTI=
ONS.  However is there a routing issue concerning SUBSCRIBE?  More specific=
ally since INVITE and OPTIONS supposedly result in the same response status=
 code, OPTIONS supposedly gets routed like INVITE.  Is there currently a re=
quirement upon middle boxes to route SUBSCRIBE the same as INVITE?


From paulej@packetizer.com  Fri Apr  2 14:01:08 2010
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Cc: "'Vivek Bhargava \(vbharga\)'" <vbharga@cisco.com>, 'Gonzalo Salgueiro' <gsalguei@cisco.com>
Subject: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Folks,

I wanted to bring your attention to an Internet Draft we submitted today
related to one use of the SIP OPTIONS message.  This draft describes the use
of OPTIONS as an application-level "ping" mechanism.

It is important to point out two things:
 1) We are not defining any new extensions to SIP, but merely attempt
    to define the use of OPTIONS as a "ping" mechanism in a way that
    will be interoperable. We've found a lot of equipment that does
    this and it is done inconsistently from one vendor or product
    to the next.  We believe we need to ensure that devices
    implement an OPTIONS "ping" mechanism with some level of
    consistency and suggest either a standards track document
    or a BCP-type document.
 2) These procedures are not intended to be used in lieu of a proper
    load balancing or capacity reporting mechanism: the scope is
    deliberately narrowly focused to allowing a SIP entity to
    determine the status of another entity, usually a next-hop
    neighbor.

We would welcome any comments on the draft.

Paul

> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-
> bounces@ietf.org] On Behalf Of Internet-Drafts@ietf.org
> Sent: Friday, April 02, 2010 11:00 AM
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-jones-sip-options-ping-00.txt
> 
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> 
> 	Title           : Using OPTIONS to Query for Operational Status
> in the Session Initiation Protocol (SIP)
> 	Author(s)       : V. Bhargava, et al.
> 	Filename        : draft-jones-sip-options-ping-00.txt
> 	Pages           : 7
> 	Date            : 2010-04-02
> 
> This document describes a procedure for using the OPTIONS method
> defined for the Session Initiation Protocol (SIP) in order to allow one
> SIP entity to query the status of another SIP entity.  Through
> discovery of the status of a SIP entity, it is possible to expedite
> session establishment or provide diagnostic information to network
> administrators.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-jones-sip-options-ping-00.txt
> 
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From kpfleming@digium.com  Fri Apr  2 14:37:28 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul E. Jones wrote:

> We would welcome any comments on the draft.

This text:

> When a receiving SIP entity is experiencing heavy load, but still
>    willing to accept new dialogs, it SHOULD return a 486 response code.
>    In receipt of a 486, the querying SIP entity SHOULD make an effort
>    to avoid establishing new dialogs with the heavily loaded server
>    until it subsequently receives a 200 from that server.
> 

could conceivably be interpreted by a reader in such a way that they'd
expect to receive the 200 response without having to have sent an
additional OPTIONS; anyone who understands SIP at all should not reach
that conclusion, but a change to '... until it receives a 200 response
code to a subsequent OPTIONS message sent to that server' would
eliminate any possible confusion.

This may be more than you intend to include in your draft, but in
Asterisk we use the response time to OPTIONS pings to adjust the T1
timer for future requests to that server as well, in order to reduce
delays when attempting to route calls. Since part of the purpose of this
draft is to document a method to reduce session delivery delays, is it
worth considering documenting this behavior as well?

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Kevin,

> could conceivably be interpreted by a reader in such a way that they'd
> expect to receive the 200 response without having to have sent an
> additional OPTIONS; anyone who understands SIP at all should not reach
> that conclusion, but a change to '... until it receives a 200 response
> code to a subsequent OPTIONS message sent to that server' would
> eliminate any possible confusion.

Great suggestion.  I've made the change in my local draft to what you
propose.
 
> This may be more than you intend to include in your draft, but in
> Asterisk we use the response time to OPTIONS pings to adjust the T1
> timer for future requests to that server as well, in order to reduce
> delays when attempting to route calls. Since part of the purpose of
> this
> draft is to document a method to reduce session delivery delays, is it
> worth considering documenting this behavior as well?

I'm certainly open to what makes sense.  The objective, of course, is to try
to reach some form of consensus on how this ought to be done and to provide
guidance, as I'm seeing so much variation in something that should be so
simple.

My first target was really the response codes.  You'll note I didn't propose
changing the rules (at least as I understand them), but it amazes me as to
how many different response codes I've seen related to use of OPTIONS as a
ping mechanism.

I'm most certainly open to adding other recommendations to the draft.  I
don't mind playing editor and taking input from all interested parties.

Paul



From christer.holmberg@ericsson.com  Sat Apr  3 03:51:12 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Adam Roach' <adam@nostrum.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>
Date: Sat, 3 Apr 2010 12:51:07 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

>>	Hi,
>>=09
>>	A good mechanism to discovery UA capabilities is essential to provide me=
aningful service:
>>=09
>>
>>
>[as an individual]
>
>If this is true -- if it really is *essential* -- then we need something b=
etter than the "maybe it works, maybe it doesn't" level of functionality th=
at we get from R-D. If discovery of UA capabilities is=20
>*essential*, then we need something that survives forking in all circumsta=
nces, like SUBSCRIBE/NOTIFY or something else that sends a request in both =
directions.

SUB/NOT is only going to work if the involved entities support the associat=
ed event package. So, it's still a "maybe it works, maybe it doesn't".

It is true that we on a protocol level cannot ensure that proxies will supp=
ort this, but in reality people will implement extensions based on the serv=
ices and capabilities that they want to provide.=20

Also, the more usefulness of extensions (in this case 3841) we can show, th=
e more likely it is that people will implement/demand them.

>I suspect it's more in the "nice to have" category.

I don't agree. As Sal said, it is essential in some cases.

Regards,

Christer



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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Hadriel Kaplan' <HKaplan@acmepacket.com>, 'Adam Roach' <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Sat, 3 Apr 2010 13:00:30 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

 Adam, thanks for sending out the the e-mail!

First, a clarification regarding "new features":

When we say "new features", I assume (based on Hadriel's comment in the mee=
ting) it means the possibility to, with a single OPTIONS request, retrieve =
the capabilities from ALL UAS(s) behind a forking proxy - instead of only o=
ne UAS (whoever sends the 200 OK to the forking proxy first).

Yeah what I meant was that in my opinion your slides had really two separat=
e classes of topics: the first were "bugs", either in incorrectly or under-=
specific RFC cases/text.  Effectively things we would put into bugzilla, an=
d possibly incorporate into any errata/bis someday.  The second was the "I =
want this request to go to all contacts for the AoR and get all their respo=
nses", which is not a bug of the current specs.

<Christer> I agree. I tried to make that separation in my presentation, by =
talking about "unclear specifications"/bugs and "forking". I never intended=
 to say that the forking case was a bug, simply that the way it is normally=
 used very much limits it's usability in forking cases.

The message I got from the meeting, and also previous e-mail discussions, w=
as that the "OPTIONS 3xx" solution already provideds this feature, and that=
 it can be used (without defining new protocol elements etc).

Does it actually solve it?  Sorry I wasn'f following the discussion of it o=
n the mailing list, but...What if there is more than one proxy in the path =
- which one is supposed to return 3xx?
What if the first proxy has two routes for the AoR, and one of those points=
 to another proxy that also has two (different) routes for the AoR?
Do you only want 3xx for Registered contact URI's?  What if the proxy doesn=
't know whether they're Registered contact URI's vs. static routes?

<Christer> I have mostly been thinking about the registrar returning the 3x=
x, but there could be cases where another forking proxy also sends 3xx, and=
 maybe the UAC can already make some decissions based on that. Now, in theo=
ry this could of course lead to a series of 3xx response, but in real life =
I don't think that is going to happen. When forking occurs in the first pla=
ce, I think the majority of cases will contain one point of forking. In som=
e cases there may be two points of forking, but more than that I think is g=
oing to be extremely rare... But, if someone thinks I'm wrong I would reall=
y much like to hear about it :)

Regards,

Christer





So, in my opinion the "adding" word is a little missleading. Instead I thin=
k the question is whether we want to DOCUMENT that existing feature.

Second, regarding bugs and interoperability issues in the field. Personally=
 I am not sure of any major ones at the moment, but that is probably becaus=
e OPTIONS haven't been very much deployed in the first place (it has been u=
sed for some keep-alive/ping purposes, but in those cases it really doesn't=
 matter what comes in the response etc).

Funny enough that's the only area I've ever actually seen OPTIONS interop p=
roblems - the response code.  We seem to need to map the response code to s=
omething different a lot.  Not sure why though.

Now, however, at least I am starting to get questions regarding the usage o=
f OPTIONS, so I guess we need to ask ourselves whether we think the bugs WI=
LL create interoperability issues.

And, the issue you raised regarding using the re-INVITE to re-create dialog=
s ("[sipcore] Mea Culpa: Dialog Necromancy in RFC 3261") also affects OPTIO=
NS, eventhough you want to keep them separate (which is ok).

Obviously I am in favor of pursuing the work, and will be willing to author=
 drafts :)

Regards,

Christer




________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Adam Roach
Sent: 30. maaliskuuta 2010 22:06
To: SIPCORE
Subject: [sipcore] SIP OPTIONS: Shall we work on this?
[as chair]

One of the topics of discussion at the recent face-to-face meeting in Anahe=
im was the SIP OPTIONS method. We have had significant discussion of some s=
pecific technical topics on the SIPCORE mailing list, such that all partici=
pants should have a reasonable overview of the scope that any related work =
may entail. Participants may also benefit from reading through the minutes =
of the corresponding conversation in Anaheim:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTI=
ONS%20Method

Rather than continue discussion of the specific technical topics related to=
 OPTIONS, the chairs would encourage the working group to engage in a discu=
ssion around whether we should, as a group, work on addressing some of the =
issues that have been identified.

In particular, we would direct the group to consider the nature of such wor=
k:

 *   Are we proposing to introduce new features, or fix existing bugs?
 *   If we are adding new features: is OPTIONS the best way to introduce th=
ese features, or would other mechanisms (presence, callee-caps) produce sup=
erior results?
 *   If we are fixing bugs: are the bugs theoretical, or can we cite actual=
 interoperability issues in the field?
 *   If there are identified interoperability issues: is the level of effor=
t involved in fixing them commensurate with the impact of the problems?
Finally, keep in mind that work does not happen unless participants are wil=
ling to invest time. If you speak out in favor of pursuing this work, pleas=
e indicate the level of time commitment that you will personally invest in =
bringing such work to a successful conclusion (e.g., will author drafts; wi=
ll perform dedicated reviews; etc).

Thanks.

/a

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<DIV dir=3Dltr align=3Dleft><SPAN class=3D865345110-03042010><FONT face=3DA=
rial=20
color=3D#800000 size=3D2>Hi,</FONT></SPAN></DIV><BR>
<DIV class=3DSection1>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: medium =
none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0in; BORDER-LEFT: blue 1.5pt solid=
; PADDING-TOP: 0in; BORDER-BOTTOM: medium none">
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;</SPAN></FONT><FONT face=3DArial color=3Dbl=
ue=20
size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">A=
dam,=20
thanks for sending out the the e-mail!</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">First,&nbsp;a=20
clarification regarding "new features": </SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">When&nbsp;we=20
say&nbsp;"new features", I assume (based on&nbsp;Hadriel's comment in the=20
meeting)&nbsp;it means the possibility to, with a single OPTIONS=20
request,&nbsp;retrieve the capabilities from ALL UAS(s) behind a forking pr=
oxy=20
-&nbsp;instead of only one UAS&nbsp;(whoever sends the 200 OK to the forkin=
g=20
proxy first).</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dnavy size=3D3><=
SPAN=20
style=3D"FONT-SIZE: 12pt; COLOR: navy"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Yeah what I mean=
t was=20
that in my opinion your slides had really two separate classes of topics: t=
he=20
first were &#8220;bugs&#8221;, either in incorrectly or under-specific RFC =
cases/text.=20
&nbsp;Effectively things we would put into bugzilla, and possibly incorpora=
te=20
into any errata/bis someday.&nbsp; The second was the &#8220;I want this re=
quest to go=20
to all contacts for the AoR and get all their responses&#8221;, which is no=
t a bug of=20
the current specs.&nbsp; <o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p>&nbsp;</o:p=
></SPAN></FONT></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p><SPAN=20
class=3D865345110-03042010><FONT color=3D#800000>&lt;Christer&gt; I agree. =
I tried=20
to make that separation in my presentation, by talking about "unclear=20
specifications"/bugs and "forking". I never intended to say that the forkin=
g=20
case was a bug, simply that the way it is normally used&nbsp;very much limi=
ts=20
it's usability in forking cases.</FONT></SPAN></o:p></SPAN></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">The message I go=
t from=20
the meeting, and also previous e-mail discussions, was that the "OPTIONS 3x=
x"=20
solution already provideds this feature, and that it can be used (without=20
defining new protocol elements etc).</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Does it actually=
 solve=20
it? &nbsp;Sorry I wasn&#8217;f following the discussion of it on the mailin=
g list,=20
but&#8230;What if there is more than one proxy in the path &#8211; which on=
e is supposed to=20
return 3xx?&nbsp; <o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">What if the firs=
t proxy=20
has two routes for the AoR, and one of those points to another proxy that a=
lso=20
has two (different) routes for the AoR? &nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Do you only want=
 3xx=20
for Registered contact URI&#8217;s? &nbsp;What if the proxy doesn&#8217;t k=
now whether=20
they&#8217;re Registered contact URI&#8217;s vs. static routes?&nbsp;&nbsp;=
<FONT=20
color=3D#0000ff><SPAN=20
class=3D865345110-03042010>&nbsp;</SPAN></FONT></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN=20
class=3D865345110-03042010></SPAN></FONT></SPAN></FONT>&nbsp;</P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p><SPAN=20
class=3D865345110-03042010><FONT color=3D#800000>&lt;Christer&gt; I have mo=
stly been=20
thinking about the registrar returning the 3xx, but there could be cases wh=
ere=20
another forking proxy also sends 3xx, and maybe the UAC can already make so=
me=20
decissions based on that. Now, in theory this could of course lead to a ser=
ies=20
of 3xx response, but in real life I don't think that is going to happen. Wh=
en=20
forking occurs in the first place, I think the majority of cases will conta=
in=20
one point of forking. In some cases there may be two points of forking, but=
 more=20
than that I think is going to be extremely rare... But, if someone thinks I=
'm=20
wrong I would really much like to hear about it=20
:)</FONT></SPAN></o:p></SPAN></SPAN></FONT></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#800000><SPAN class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p><SPAN=20
class=3D865345110-03042010></SPAN></o:p></SPAN></SPAN></FONT></SPAN></FONT>=
&nbsp;</P>
<P class=3DMsoNormal><FONT><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#800000><SPAN class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p><SPAN=20
class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p><SPAN=20
class=3D865345110-03042010><FONT=20
color=3D#800000>Regards,</FONT></SPAN></o:p></SPAN></SPAN></FONT></SPAN></S=
PAN></o:p></SPAN></SPAN></FONT></SPAN></FONT></P>
<P class=3DMsoNormal><FONT><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#800000><SPAN class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p><SPAN=20
class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#800000><SPAN class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p><SPAN=20
class=3D865345110-03042010></SPAN></o:p></SPAN></SPAN></FONT></SPAN></SPAN>=
</o:p></SPAN></SPAN></FONT></SPAN></FONT>&nbsp;</P>
<P class=3DMsoNormal><FONT><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#800000><SPAN class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p><SPAN=20
class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><SPAN=20
class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p><SPAN=20
class=3D865345110-03042010>Christer</SPAN></o:p></SPAN></SPAN></SPAN></SPAN=
></o:p></SPAN></SPAN></FONT></SPAN></FONT></P>
<P class=3DMsoNormal><FONT><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#800000><SPAN class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p><SPAN=20
class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#800000><SPAN class=3D865345110-03042010><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p><SPAN=20
class=3D865345110-03042010></SPAN></o:p></SPAN></SPAN></FONT></SPAN></SPAN>=
</o:p></SPAN></SPAN></FONT></SPAN></FONT>&nbsp;</P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN=20
class=3D865345110-03042010></SPAN></FONT></SPAN></FONT>&nbsp;</P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN=20
class=3D865345110-03042010></SPAN></FONT></SPAN></FONT>&nbsp;</P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN=20
class=3D865345110-03042010>&nbsp;</SPAN><o:p></o:p></FONT></SPAN></FONT></P=
>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p>&nbsp;</o:p=
></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">So, in my opinio=
n the=20
"adding" word is a little missleading. Instead I think the question is whet=
her=20
we want to DOCUMENT that existing feature.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Second, regardin=
g bugs=20
and interoperability issues in the field. Personally I am not sure of any m=
ajor=20
ones at the moment, but that is probably because OPTIONS haven't been very =
much=20
deployed&nbsp;in the first place (it has been used for some keep-alive/ping=
=20
purposes, but in those cases it really doesn't matter what comes in the res=
ponse=20
etc). </SPAN></FONT><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p></o:p></SPA=
N></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p>&nbsp;</o:p=
></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Funny enough tha=
t&#8217;s the=20
only area I&#8217;ve ever actually seen OPTIONS interop problems &#8211; th=
e response=20
code.&nbsp; We seem to need to map the response code to something different=
 a=20
lot.&nbsp; Not sure why though. <o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p>&nbsp;</o:p=
></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Now, however, at=
 least=20
I am starting to get questions regarding the usage of OPTIONS, so I guess w=
e=20
need to ask ourselves whether we think the bugs WILL create interoperabilit=
y=20
issues.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">And, the issue y=
ou=20
raised regarding&nbsp;using the re-INVITE to re-create dialogs ("[sipcore] =
Mea=20
Culpa: Dialog Necromancy in RFC 3261") also affects OPTIONS, eventhough you=
 want=20
to keep them separate (which is ok).</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Obviously I am i=
n favor=20
of pursuing the work, and will be willing to author drafts=20
:)</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Regards,</SPAN><=
/FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Christer</SPAN><=
/FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<DIV class=3DMsoNormal style=3D"TEXT-ALIGN: center" align=3Dcenter><FONT=20
face=3D"Times New Roman" color=3Dblack size=3D3><SPAN style=3D"FONT-SIZE: 1=
2pt">
<HR tabIndex=3D-1 align=3Dcenter width=3D"100%" SIZE=3D2>
</SPAN></FONT></DIV>
<P class=3DMsoNormal style=3D"MARGIN-BOTTOM: 12pt"><B><FONT face=3DTahoma c=
olor=3Dblack=20
size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">From:</SP=
AN></FONT></B><FONT=20
face=3DTahoma size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"=
>=20
sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] <B><SPAN=20
style=3D"FONT-WEIGHT: bold">On Behalf Of </SPAN></B><st1:PersonName w:st=3D=
"on">Adam=20
Roach</st1:PersonName><BR><B><SPAN style=3D"FONT-WEIGHT: bold">Sent:</SPAN>=
</B>=20
30. maaliskuuta 2010 22:06<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPA=
N></B>=20
<st1:PersonName w:st=3D"on">SIPCORE</st1:PersonName><BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> [sipcore] SIP OPTIONS: Shal=
l we=20
work on this?</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">[as chair]<BR><BR>One of the topics of discussion=
 at the=20
recent face-to-face meeting in <st1:City w:st=3D"on"><st1:place=20
w:st=3D"on">Anaheim</st1:place></st1:City> was the SIP OPTIONS method. We h=
ave had=20
significant discussion of some specific technical topics on the <st1:Person=
Name=20
w:st=3D"on">SIPCORE</st1:PersonName> mailing list, such that all participan=
ts=20
should have a reasonable overview of the scope that any related work may en=
tail.=20
Participants may also benefit from reading through the minutes of the=20
corresponding conversation in <st1:City w:st=3D"on"><st1:place=20
w:st=3D"on">Anaheim</st1:place></st1:City>:<BR><BR><A=20
href=3D"http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SI=
P%20OPTIONS%20Method">http://www.ietf.org/proceedings/10mar/minutes/sipcore=
.html#The%20SIP%20OPTIONS%20Method</A><BR><BR>Rather=20
than continue discussion of the specific technical topics related to OPTION=
S,=20
the chairs would encourage the working group to engage in a discussion arou=
nd=20
whether we should, as a group, work on addressing some of the issues that h=
ave=20
been identified.<BR><BR>In particular, we would direct the group to conside=
r the=20
nature of such work:<o:p></o:p></SPAN></FONT></P>
<UL type=3Ddisc>
  <LI class=3DMsoNormal=20
  style=3D"mso-margin-top-alt: auto; mso-margin-bottom-alt: auto; mso-list:=
 l0 level1 lfo1"><FONT=20
  face=3D"Times New Roman" color=3Dblack size=3D3><SPAN style=3D"FONT-SIZE:=
 12pt">Are we=20
  proposing to introduce new features, or fix existing bugs?=20
  <o:p></o:p></SPAN></FONT>
  <LI class=3DMsoNormal=20
  style=3D"mso-margin-top-alt: auto; mso-margin-bottom-alt: auto; mso-list:=
 l0 level1 lfo1"><FONT=20
  face=3D"Times New Roman" color=3Dblack size=3D3><SPAN style=3D"FONT-SIZE:=
 12pt">If we=20
  are adding new features: is OPTIONS the best way to introduce these featu=
res,=20
  or would other mechanisms (presence, callee-caps) produce superior=20
  results?<o:p></o:p></SPAN></FONT>=20
  <LI class=3DMsoNormal=20
  style=3D"mso-margin-top-alt: auto; mso-margin-bottom-alt: auto; mso-list:=
 l0 level1 lfo1"><FONT=20
  face=3D"Times New Roman" color=3Dblack size=3D3><SPAN style=3D"FONT-SIZE:=
 12pt">If we=20
  are fixing bugs: are the bugs theoretical, or can we cite actual=20
  interoperability issues in the field? <o:p></o:p></SPAN></FONT>
  <LI class=3DMsoNormal=20
  style=3D"mso-margin-top-alt: auto; mso-margin-bottom-alt: auto; mso-list:=
 l0 level1 lfo1"><FONT=20
  face=3D"Times New Roman" color=3Dblack size=3D3><SPAN style=3D"FONT-SIZE:=
 12pt">If=20
  there are identified interoperability issues: is the level of effort invo=
lved=20
  in fixing them commensurate with the impact of the problems?=20
  <o:p></o:p></SPAN></FONT></LI></UL>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" color=3Dblack size=3D3>=
<SPAN=20
style=3D"FONT-SIZE: 12pt">Finally, keep in mind that work does not happen u=
nless=20
participants are willing to invest time. If you speak out in favor of pursu=
ing=20
this work, please indicate the level of time commitment that you will perso=
nally=20
invest in bringing such work to a successful conclusion (e.g., will author=
=20
drafts; will perform dedicated reviews;=20
etc).<BR><BR>Thanks.<BR><BR>/a<o:p></o:p></SPAN></FONT></P></DIV></DIV></BO=
DY></HTML>

--_000_FF84A09F50A6DC48ACB6714F4666CC745E21B5A915ESESSCMS0354e_--

From christer.holmberg@ericsson.com  Sat Apr  3 04:06:38 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Peter Musgrave' <peter.musgrave@magorcorp.com>, SIPCORE <sipcore@ietf.org>
Date: Sat, 3 Apr 2010 13:06:34 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>I agree with the assertion (Hadrials?) that there are two things here:
>bugs in OPTIONS and a need to query multiple UAs.
>
>I have not seen wide use of OPTIONS and have not encountered these bugs, I=
n general I favour fixing them but I am not sure if this is more important =
that other work which could be done.
>
>As for assessing multiple UA capabilities - I agree this is useful. Is OPT=
IONS the only way forward here? Could I not use e.g. gruu-reg-event and the=
n send specifically targeted OPTIONS to each gruu? Is=20
>requiring gruu support in UAs viewed as 'setting the bar too high?' This f=
ails to pick up forking due to dial plan entries - is that a requirement?

In the OPTIONS 3xx case, after the UAC has received the 3xx it would send i=
ndividual OPTIONS request. The 3xx may contain gruus, or normal contacts.

So, it doesn't require support of gruu. It doesn't really require any suppo=
rt from the UAS.

Regards,

Christer

From christer.holmberg@ericsson.com  Sat Apr  3 04:10:14 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Date: Sat, 3 Apr 2010 13:10:09 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>>What if the first proxy has two routes for the AoR, and one of those=20
>>points to another proxy that also has two (different) routes for the AoR?
>
>That should all be handled by recursion on 3xx response processing.
>The first proxy would return 3xx with its choices. One of those would lead=
 to the 2nd proxy, that then would return its own 3xx.

That might happen, yes.

>>Do you only want 3xx for Registered contact URI's?  What if the proxy=20
>>doesn't know whether they're Registered contact URI's vs. static routes?
>
>IMO this is one of the big problems with the IMS approach to services.=20
>They aren't registered, and aren't anything that the S-CSCF could (or
>would) return as 3xx alternatives. So the caller will be limited in what o=
ptions are visible to it.

I am not really sure I understand what you mean...

However, there are of course limitations on how much information the caller=
 will get, based on what information the UAS have registered, and what info=
rmation can be returned in the 3xx in the first place.

Regards,

Christer


From christer.holmberg@ericsson.com  Sat Apr  3 04:15:21 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Hadriel Kaplan' <HKaplan@acmepacket.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>When you say "is essential to providing meaningful service" in what way do=
 you mean that?
>
>Do you mean literally that you need any UAC anywhere to be able to discove=
r the capabilities of any arbitrary UAS it might want to communicate with i=
n the World using OPTIONS?

Anyone associated with an AOR, yes, which of course may located anywhere in=
 the world :)

>Or do you mean within a provider's domain, having an App Server figure out=
 what each registered UA endpoint supports and putting that in some databas=
e is very useful? (which I could see 3GPP doing)

I guess that could be a potential use-case also.

>Because if it's the latter, then creating an options-tag for rfc3841 and p=
utting it in Proxy-Require isn't that bad an idea - proxy-require only has =
deployment issues when you want to use it across=20
>proxies outside of your administrative control.

Far from every proxy needs to understand RFC3841, and that is the problem w=
ith Proxy-Require - it requires everyone to understand something.

(It would be really great if it would be possible to require proxies to sup=
port stuff only in certain conditions, e.g. if they act as a registrar and =
intend to do forking).

Regards,

Christer


=20

________________________________

From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Salvatore Loreto
Sent: Thursday, April 01, 2010 3:36 AM
To: sipcore@ietf.org
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?

=20

Hi,

A good mechanism to discovery UA capabilities is essential to provide meani=
ngful service:
So I am in favor of pursuing the work,=20
in particular the possibility to use a single OPTIONS request to retrieve t=
he capabilities from ALL UAS(s) behind a forking proxy.=20

I agree with Christer, that as discussed in the ml and during the f2f meeti=
ng, the 3xx solution already provide this feature
even if with the limitations Paul has highlighted (e.g. honoring R-D is opt=
ional by the proxy).

I do think, it would be opportune document it in a BCP to document/clarify =
the ambiguous stuff, and I can help to co-edit it if we decide to write one=
.

Explore the possibility to use other mechanisms (e.g. presence, something s=
imilar to XEP-0115) and define it would be also interesting,
but IMO it would require a major effort and more cycles and I am not so sur=
e there are enough people willing to accomplish such effort.

cheers
Sal



On 3/30/10 9:05 PM, Adam Roach wrote:=20

[as chair]

One of the topics of discussion at the recent face-to-face meeting in Anahe=
im was the SIP OPTIONS method. We have had significant discussion of some s=
pecific technical topics on the SIPCORE mailing list, such that all partici=
pants should have a reasonable overview of the scope that any related work =
may entail. Participants may also benefit from reading through the minutes =
of the corresponding conversation in Anaheim:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTI=
ONS%20Method

Rather than continue discussion of the specific technical topics related to=
 OPTIONS, the chairs would encourage the working group to engage in a discu=
ssion around whether we should, as a group, work on addressing some of the =
issues that have been identified.

In particular, we would direct the group to consider the nature of such wor=
k:

*	Are we proposing to introduce new features, or fix existing bugs?=20
*	If we are adding new features: is OPTIONS the best way to introduce these=
 features, or would other mechanisms (presence, callee-caps) produce superi=
or results?=20
*	If we are fixing bugs: are the bugs theoretical, or can we cite actual in=
teroperability issues in the field?=20
*	If there are identified interoperability issues: is the level of effort i=
nvolved in fixing them commensurate with the impact of the problems?=20

Finally, keep in mind that work does not happen unless participants are wil=
ling to invest time. If you speak out in favor of pursuing this work, pleas=
e indicate the level of time commitment that you will personally invest in =
bringing such work to a successful conclusion (e.g., will author drafts; wi=
ll perform dedicated reviews; etc).

Thanks.

/a


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Cc: SIPCORE <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Christer Holmberg wrote:

>> IMO this is one of the big problems with the IMS approach to services. 
>> They aren't registered, and aren't anything that the S-CSCF could (or
>> would) return as 3xx alternatives. So the caller will be limited in what options are visible to it.
> 
> I am not really sure I understand what you mean...

For instance, suppose I want to talk to your voicemail, without ringing 
your regular phone(s). I might send an OPTIONS in hope of getting a list 
of Contacts, among which the VM contact might be found.

AFAIK, IMS isn't set up in a way to have an identifiable contact for the 
VM server that it could return in the 3xx. (But I'll be pleasantly 
surprised to learn I'm wrong.)

	Thanks,
	Paul

From HKaplan@acmepacket.com  Sat Apr  3 09:20:12 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, 'Adam Roach' <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Sat, 3 Apr 2010 12:19:36 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi Christer,=20
comments inline...
(also, let's keep this plain text format :)

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: Saturday, April 03, 2010 7:01 AM
> To: Hadriel Kaplan; 'Adam Roach'; SIPCORE
> Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?
>
> I agree. I tried to make that separation in my presentation, by talking
> about "unclear specifications"/bugs and "forking". I never intended to sa=
y
> that the forking case was a bug, simply that the way it is normally
> usedvery much limits it's usability in forking cases.
=A0
No debate there - as far as I can tell rfc3841 req-disp is used in very lim=
ited and constrained cases, under specific configuration control. (when it'=
s used at all)

> I have mostly been thinking about the registrar returning the 3xx, but
> there could be cases where another forking proxy also sends 3xx, and mayb=
e
> the UAC can already make some decissions based on that. Now, in theory
> this could of course lead to a series of 3xx response, but in real life I
> don't think that is going to happen. When forking occurs in the first
> place, I think the majority of cases will contain one point of forking. I=
n
> some cases there may be two points of forking, but more than that I think
> is going to be extremely rare... But, if someone thinks I'm wrong I would
> really much like to hear about it :)

Easy answer: go beyond the local domain.  If the SIP request goes inter-dom=
ain, it's not uncommon for proxies along the path to have multiple routes. =
 For example to multiple transits, or to multiple ingress/egress point next=
-hops, or to PSTN-fallback gateways, etc.  These are all "forks" (serial fo=
rks).

Even within the domain, it's not always obvious that the reg-proxy knows wh=
ich database entries are registered contacts vs. static routes - for exampl=
e fallback routes, or voicemail routes, or invoked b2bua app-servers - and =
even if it knows, the request-disposition does not specify what types it ap=
plies to.

So ISTM that this thing will need to be configured: enabled on specific pro=
xies for specific route types, and possibly for specific methods.

-hadriel

From HKaplan@acmepacket.com  Sat Apr  3 09:41:03 2010
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To: Christer Holmberg <christer.holmberg@ericsson.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sat, 3 Apr 2010 12:40:18 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Saturday, April 03, 2010 7:15 AM
> To: Hadriel Kaplan; Salvatore Loreto; sipcore@ietf.org
>=20
> Far from every proxy needs to understand RFC3841, and that is the problem
> with Proxy-Require - it requires everyone to understand something.

No debate there - but I don't know how you get "guaranteed" usage otherwise=
.
For example, if your OPTIONS request crosses a proxy which recurses on 3xx,=
 and that proxy forwards it to a proxy which understands req-disp and sends=
 back the 3xx.  The recursing middle-proxy will handle it, not send it back=
 to you.  And across domains 3xx-recursing proxies are common.

-hadriel

From brett@broadsoft.com  Sat Apr  3 10:09:52 2010
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From: Brett Tate <brett@broadsoft.com>
To: "vbharga@cisco.com" <vbharga@cisco.com>, "paulej@packetizer.com" <paulej@packetizer.com>, "gsalguei@cisco.com" <gsalguei@cisco.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sat, 3 Apr 2010 10:08:59 -0700
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Subject: [sipcore] draft-jones-sip-options-ping-00: comments
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Glad to see someone attempting to capture the currently common use for OPTI=
ONS.

Here are a few comments.

Thanks,
Brett

---------

To help avoid receiving ignored data within 200 response, I recommend use o=
f an option-tag (such a options-ping) or something else included within OPT=
IONS to clearly indicate that the sender does not actually want all the dat=
a which can be provided within 200 response.

Section 5.1 indicates "To avoid unduly taxing a receiving SIP entity, trans=
mitters of OPTIONS messages MUST honor the Retry-After header field if rece=
ived."  The MUST requirement is likely too strong because of implications w=
hen included within a 503.  The MUST requirement is likely also too strong =
because of vulnerability to quit sending requests to that device for an ext=
remely long time because attacker or bug caused the 503 to be sent containi=
ng Retry-After with an extremely large value.

Concerning receiving 503 with Retry-After, to what does it apply:
1) OPTIONS request target ip-address,
2) OPTIONS response source ip-address,
3) both 1 and 2,
4) something else.


From HKaplan@acmepacket.com  Sat Apr  3 12:12:54 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sat, 3 Apr 2010 15:12:40 -0400
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Cc: "'Vivek Bhargava \(vbharga\)'" <vbharga@cisco.com>, 'Gonzalo Salgueiro' <gsalguei@cisco.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hey Paul,
Right, so I suspect most everyone would agree that this is the most common =
usage of OPTIONS, or at least one of 'em.  And I agree it has interop probl=
ems left and right, and requires manual provisioning and testing of specifi=
c behavior on a per hop basis.=20

The problem is I suspect it would be hard to get this to be a WG doc. :(

First, because it's an ambiguous usage of the method.  You could define an =
option-tag to make it clear, however. (along with a header to say "I'm usin=
g this option-tag's mechanism for this request") =20

But more importantly because it conflicts with two other pieces of work bei=
ng done in the IETF:
1) The keep-alive portion, which is done by RFC 5626 and also being worked =
on in draft-ietf-sipcore-keep.
2) The overload portion, for which people are trying to form a new WG out o=
f DISPATCH, based on RFC 5390 and the work of a design team implementing an=
d testing various individual drafts.  If I recall right, that work actually=
 started because the 503 Retry-After model was found to be poor.

You could, of course, just go for an individual informational publication. =
 But without the option-tag (which requires a standards-track doc right now=
), it wouldn't solve the problem of manual configuration, because different=
 devices currently do different things than what this doc recommends.

Other, specific comments on the draft:
0) You might want to consider specifying what the To/From-URIs should be se=
nt as, and that a receiver should be lenient.  We've seen interop problems =
with those URIs for OPTIONS pings.

1) Section 6 says: "When a receiving SIP entity is unable to process SIP me=
ssages, it SHOULD respond with a 503 response code."  Clearly you can't mea=
n that literally, since OPTIONS is a SIP message and responding to it would=
 require processing it. :)  But you presumably also don't mean when the ent=
ity is too busy, because you recommend using 486 for that case.  So what is=
 this intended to mean?  Unable to process new dialogs?

2) Section 6 says: "A SIP entity MUST respond to an OPTIONS method with a 2=
00 response code when it is willing and able to process SIP messages."  Aga=
in, this can't be meant literally, because two paragraphs later a 486 is in=
dicated when it is experiencing heavy load but still willing to accept new =
dialogs.  Maybe just a "...unless one of the following cases apply" needs t=
o be added at the end of the sentence.  BTW, do you really mean "SIP messag=
es", or do you mean out-of-dialog requests other than OPTIONS, or only dial=
og-forming requests?

3) Section 7 says: "If a requesting entity receives a 486 or 503
   response code, it MAY send a subsequent OPTIONS messages in order to
   detect a change in operational status, but it MUST honor the Retry-
   After header field."
Do you mean it must honor the Retry-After such that it does not send the OP=
TIONS ping based on it?  Or do you only mean it has to honor the Retry-Afte=
r for other request types?  Also, it's a nit, but the sentence should be ".=
..honor the Retry-After header field value received in the previous respons=
e."

4) Section 5.1 says: "The OPTIONS message MUST be transmitted directly to t=
he peer's IP address and not to an intermediary SIP entity."  What do you m=
ean by "intermediary SIP entity"?  Either it is the next-hop, or it's not. =
 If your next-hop is X, send it to X.  If to get to your "peer" you have to=
 use a proxy, then that proxy *is* your next-hop, and *is* your "peer".  By=
passing it won't help you in the general case. (in some specific cases, sur=
e, but not generally; and in some cases it makes things worse, so it can't =
be a MUST)

-hadriel

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Paul E. Jones
> Sent: Friday, April 02, 2010 4:47 PM
> To: sipcore@ietf.org
> Cc: 'Vivek Bhargava (vbharga)'; 'Gonzalo Salgueiro'
> Subject: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
>=20
> Folks,
>=20
> I wanted to bring your attention to an Internet Draft we submitted today
> related to one use of the SIP OPTIONS message.  This draft describes the
> use
> of OPTIONS as an application-level "ping" mechanism.
>=20
> It is important to point out two things:
>  1) We are not defining any new extensions to SIP, but merely attempt
>     to define the use of OPTIONS as a "ping" mechanism in a way that
>     will be interoperable. We've found a lot of equipment that does
>     this and it is done inconsistently from one vendor or product
>     to the next.  We believe we need to ensure that devices
>     implement an OPTIONS "ping" mechanism with some level of
>     consistency and suggest either a standards track document
>     or a BCP-type document.
>  2) These procedures are not intended to be used in lieu of a proper
>     load balancing or capacity reporting mechanism: the scope is
>     deliberately narrowly focused to allowing a SIP entity to
>     determine the status of another entity, usually a next-hop
>     neighbor.
>=20
> We would welcome any comments on the draft.
>=20
> Paul

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Brett Tate'" <brett@broadsoft.com>, <vbharga@cisco.com>, <gsalguei@cisco.com>, <sipcore@ietf.org>
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Subject: Re: [sipcore] draft-jones-sip-options-ping-00: comments
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Brett,

> Glad to see someone attempting to capture the currently common use for
> OPTIONS.

Thanks... indeed, this looked like an area that needed addressing to us.

> To help avoid receiving ignored data within 200 response, I recommend
> use of an option-tag (such a options-ping) or something else included
> within OPTIONS to clearly indicate that the sender does not actually
> want all the data which can be provided within 200 response.

I'm not opposed to this, per se, but we were trying to avoid introducing any
new signaling features at all.  If folks thought it might be useful, we can
do that.  One problem, though, is that all devices would immediately be
non-compliant ;-) 

I thought about the implications of this a little before writing the draft.
If a SIP message is sent between two SBCs, for example, quite likely the
receiving device is going to reply with something that's pre-configured
(perhaps even pre-composed) and the receiver is only going to look at the
status code of the reply.  So, I wondered whether it would even matter.

> Section 5.1 indicates "To avoid unduly taxing a receiving SIP entity,
> transmitters of OPTIONS messages MUST honor the Retry-After header
> field if received."  The MUST requirement is likely too strong because
> of implications when included within a 503.  The MUST requirement is
> likely also too strong because of vulnerability to quit sending
> requests to that device for an extremely long time because attacker or
> bug caused the 503 to be sent containing Retry-After with an extremely
> large value.

Perhaps you're right, but do you have suggestions for alternative wording?
Are you suggesting to replace "MUST" with "SHOULD"?

> Concerning receiving 503 with Retry-After, to what does it apply:
> 1) OPTIONS request target ip-address,
> 2) OPTIONS response source ip-address,
> 3) both 1 and 2,
> 4) something else.

I'm a little confused by your question. Are you asking whether it applies to
the target address to which the sender of the OPTIONS sent the request or to
the address from which the recipient returned a 503, or the address of an
intermediary?

I would say it refers to the address to which the request was sent.  If
you're concerned that a reply might come from an intermediary or a rogue
endpoint, I suppose you have reason for concern.  That said, one should not
have an intermediary in the path, as it somewhat defeats the purpose.  If
it's a security issue, I would say one needs to consider the security of
messages just as they would any other message.

Paul



From paulej@packetizer.com  Sat Apr  3 20:18:49 2010
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Hadriel Kaplan'" <HKaplan@acmepacket.com>, <sipcore@ietf.org>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hadriel,

> The problem is I suspect it would be hard to get this to be a WG doc.
> :(

That might be tough, but I'd be content with something less than that, such
an informational document that folks can agree to.  I think it's important
to get something that we can refer to and say "that's how it's supposed to
work" since I see many variants.
 
> First, because it's an ambiguous usage of the method.  You could define
> an option-tag to make it clear, however. (along with a header to say
> "I'm using this option-tag's mechanism for this request")

Is it ambiguous?  The fact so many people elected to use it suggests that it
makes sense to use.  What is ambiguous is the exact usage.
 
> But more importantly because it conflicts with two other pieces of work
> being done in the IETF:
> 1) The keep-alive portion, which is done by RFC 5626 and also being
> worked on in draft-ietf-sipcore-keep.

That document covers a different problem that's not related to an
application "ping" between two devices.  One could view it as such for
active sessions, I suppose, but it really has a different focus.

> 2) The overload portion, for which people are trying to form a new WG
> out of DISPATCH, based on RFC 5390 and the work of a design team
> implementing and testing various individual drafts.  If I recall right,
> that work actually started because the 503 Retry-After model was found
> to be poor.

Likewise, this one is also different.  We're not trying to replicate the
overload work.  Everything specified in this document, I think, is what the
SIP specs say today, except that nothing states that use of OPTIONS can be
used as an application-level "ping".  The 503 response indicating that the
device is overloaded, for example, came directly from RFCF 3261 Section
21.5.4.  If there are issues with use of 503 in a "ping" between two
adjacent devices, then we ought to consider that, but in no way do I want to
replicate the work over "overload", which has a far broader scope.
 
> You could, of course, just go for an individual informational
> publication.  But without the option-tag (which requires a standards-
> track doc right now), it wouldn't solve the problem of manual
> configuration, because different devices currently do different things
> than what this doc recommends.

I'm happy to go either way, so long as folks agree.  If we can't agree to a
set of procedures, then there's not much value in publishing even an
informational document.
 
> Other, specific comments on the draft:
> 0) You might want to consider specifying what the To/From-URIs should
> be sent as, and that a receiver should be lenient.  We've seen interop
> problems with those URIs for OPTIONS pings.

I wanted to be rather strict in saying that only sip:192.168.4.56 was
acceptable, for example, but others I consulted with thought that was too
restrictive.  So, the language is soft here.

What is the preference of the group on this?
 
> 1) Section 6 says: "When a receiving SIP entity is unable to process
> SIP messages, it SHOULD respond with a 503 response code."  Clearly you
> can't mean that literally, since OPTIONS is a SIP message and
> responding to it would require processing it. :)  But you presumably
> also don't mean when the entity is too busy, because you recommend
> using 486 for that case.  So what is this intended to mean?  Unable to
> process new dialogs?

We chose 503 because that is what section 21.5.4 of RFC 3261 says to return.
So, yeah, we meant that ;-)  Perhaps "unable" is the word that needs
changing, but that was also the word used in 3261.  What would be better?

> 2) Section 6 says: "A SIP entity MUST respond to an OPTIONS method with
> a 200 response code when it is willing and able to process SIP
> messages."  Again, this can't be meant literally, because two
> paragraphs later a 486 is indicated when it is experiencing heavy load
> but still willing to accept new dialogs.  Maybe just a "...unless one
> of the following cases apply" needs to be added at the end of the
> sentence.  BTW, do you really mean "SIP messages", or do you mean out-
> of-dialog requests other than OPTIONS, or only dialog-forming requests?

Good point.  I'll adjust the sentence to add a condition as you suggest.  We
did mean "SIP messages" since the purpose is to determine, not whether the
peer device is able to process the OPTIONS, but whether it is willing and
able to process any SIP message.  How to put that in words?  Anyway, if the
peer device is out of service, we want OPTIONS to tell us that before we
waste time trying to send an INVITE.
 
> 3) Section 7 says: "If a requesting entity receives a 486 or 503
>    response code, it MAY send a subsequent OPTIONS messages in order to
>    detect a change in operational status, but it MUST honor the Retry-
>    After header field."
> Do you mean it must honor the Retry-After such that it does not send
> the OPTIONS ping based on it?  Or do you only mean it has to honor the
> Retry-After for other request types?  Also, it's a nit, but the
> sentence should be "...honor the Retry-After header field value
> received in the previous response."

I added the nit clause to the draft :-)

As for the meaning, I'm confused with your question.  The intent is to obey
the Retry-After header.  If the reply to the OPTIONS has a Retry-After
header, then don't send another OPTIONS until that time expires.

> 4) Section 5.1 says: "The OPTIONS message MUST be transmitted directly
> to the peer's IP address and not to an intermediary SIP entity."  What
> do you mean by "intermediary SIP entity"?  Either it is the next-hop,
> or it's not.  If your next-hop is X, send it to X.  If to get to your
> "peer" you have to use a proxy, then that proxy *is* your next-hop, and
> *is* your "peer".  Bypassing it won't help you in the general case. (in
> some specific cases, sure, but not generally; and in some cases it
> makes things worse, so it can't be a MUST)

This sentence is only to reinforce what you're saying (I think).  One should
not use these procedures to do a "ping" between A and C by addressing the
message to C, but then transmitting it to a proxy B to forward to C.

Paul



From christer.holmberg@ericsson.com  Sat Apr  3 21:31:43 2010
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To: 'Hadriel Kaplan' <HKaplan@acmepacket.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sun, 4 Apr 2010 06:31:37 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>>Far from every proxy needs to understand RFC3841, and that is the=20
>>problem with Proxy-Require - it requires everyone to understand something=
.
>
>No debate there - but I don't know how you get "guaranteed" usage otherwis=
e.
>For example, if your OPTIONS request crosses a proxy which recurses on 3xx=
, and that proxy forwards it to a proxy which understands req-disp and send=
s back the 3xx.  The recursing middle-proxy will=20
>handle it, not send it back to you.  And across domains 3xx-recursing prox=
ies are common.

I agree, that might happen.

But, that is a general 3xx response "feature", and again one of thoese thin=
gs where there is very little we can do on a protocol level - we just have =
to hope people will configure their deployments in a right way. And, having=
 a document (e.g. a BCP, in this case) would help them in doing that, in my=
 opinion.

Regards,

Christer



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Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>>I agree. I tried to make that separation in my presentation, by=20
>>talking about "unclear specifications"/bugs and "forking". I never=20
>>intended to say that the forking case was a bug, simply that the way=20
>>it is normally usedvery much limits it's usability in forking cases.
>=A0
>No debate there - as far as I can tell rfc3841 req-disp is used in very li=
mited and constrained cases, under specific configuration control. (when it=
's used at all)

I think 3841 is the hidden jewel of SIP. And, eventhough I may not like eve=
ry detail init, in my opinion it can be a very useful tool.=20

>>I have mostly been thinking about the registrar returning the 3xx, but=20
>>there could be cases where another forking proxy also sends 3xx, and=20
>>maybe the UAC can already make some decissions based on that. Now, in=20
>>theory this could of course lead to a series of 3xx response, but in=20
>>real life I don't think that is going to happen. When forking occurs=20
>>in the first place, I think the majority of cases will contain one=20
>>point of forking. In some cases there may be two points of forking,=20
>>but more than that I think is going to be extremely rare... But, if=20
>>someone thinks I'm wrong I would really much like to hear about it :)
>
>Easy answer: go beyond the local domain.  If the SIP request goes inter-do=
main, it's not uncommon for proxies along the path to have multiple routes.=
  For example to multiple transits, or to multiple=20
>ingress/egress point next-hops, or to PSTN-fallback gateways, etc.  These =
are all "forks" (serial forks).

Agree. But, in this case I don't really see a big reason in those proxies s=
ending a 3xx response, since the request is forwarded towards the same dest=
ination, simply using different routes.

I think this is another case where we seem to think that the world spins ar=
ound the protocol, and operators won't be able to configure their equipment=
... :)

>Even within the domain, it's not always obvious that the reg-proxy knows w=
hich database entries are registered contacts vs. static routes - for examp=
le fallback routes, or voicemail routes, or invoked=20
>b2bua app-servers - and even if it knows, the request-disposition does not=
 specify what types it applies to.

You can define new feature tags/feature tag values, if needed, e.g. to indi=
cate "voicemail". Maybe that would be very useful, if I e.g. need to leave =
you a message, but I don't have time to talk with you in person at this mom=
ent.=20

And, local policies can be used on what contacts to return in the 3xx.

>So ISTM that this thing will need to be configured: enabled on specific pr=
oxies for specific route types, and possibly for specific methods.

For example, yes.

Regards,

Christer


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Paul E. Jones" <paulej@packetizer.com>, 'Brett Tate' <brett@broadsoft.com>, "vbharga@cisco.com" <vbharga@cisco.com>, "gsalguei@cisco.com" <gsalguei@cisco.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sun, 4 Apr 2010 00:49:35 -0400
Thread-Topic: [sipcore] draft-jones-sip-options-ping-00: comments
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Subject: Re: [sipcore] draft-jones-sip-options-ping-00: comments
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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Paul E. Jones
> Sent: Saturday, April 03, 2010 10:39 PM
>=20
> [snipped text, but topic is option-tags]
>
> I'm not opposed to this, per se, but we were trying to avoid introducing
> any
> new signaling features at all.  If folks thought it might be useful, we
> can
> do that.  One problem, though, is that all devices would immediately be
> non-compliant ;-)

I was worried about this too ('cause I also suggested option-tags and was w=
orried about code-changes), but ISTM this is basically already the case: un=
less most folks do exactly what this doc says today, they're non-compliant =
and would require changing either code or config.

The thing is, afaict, many folks DON'T do what the doc says.  Not exactly, =
which is the same thing (right?).  For example, of three rather popular ven=
dors I just looked at OPTION-ping traces of, none used the 486 for anything=
.  I don't know what they'd do if they got one.  All 3 of the vendors use 5=
03 to signal "out-of-service" back, but at least 2 of them still allow (and=
 want/expect to receive) in-dialog messages - I can't tell with the 3rd.  A=
nd one of them still _sent_ all messages even if it got a 503.=20

-hadriel

From christer.holmberg@ericsson.com  Sat Apr  3 21:51:46 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Sun, 4 Apr 2010 06:51:41 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: SIPCORE <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>>>IMO this is one of the big problems with the IMS approach to services.=20
>>>They aren't registered, and aren't anything that the S-CSCF could (or
>>>would) return as 3xx alternatives. So the caller will be limited in what=
 options are visible to it.
>>=20
>>I am not really sure I understand what you mean...
>
>For instance, suppose I want to talk to your voicemail, without ringing yo=
ur regular phone(s). I might send an OPTIONS in hope of getting a list of C=
ontacts, among which the VM contact might be found.

As I also indicated in my reply to Hadriel's e-mail, for this case I think =
it would be very useful to have a VM feature tag, or something else in the =
returned contact that indicates voicemail.

>AFAIK, IMS isn't set up in a way to have an identifiable contact for the V=
M server that it could return in the 3xx. (But I'll be pleasantly surprised=
 to learn I'm wrong.)

IMS uses 3841.

Regards,

Christer


From christer.holmberg@ericsson.com  Sat Apr  3 21:59:37 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "'Paul E. Jones'" <paulej@packetizer.com>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sun, 4 Apr 2010 06:59:33 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Cc: "'Vivek Bhargava \(vbharga\)'" <vbharga@cisco.com>, 'Gonzalo Salgueiro' <gsalguei@cisco.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,=20
=20
>>But more importantly because it conflicts with two other pieces of=20
>>work being done in the IETF:
>>1) The keep-alive portion, which is done by RFC 5626 and also being=20
>>worked on in draft-ietf-sipcore-keep.
>
>That document covers a different problem that's not related to an applicat=
ion "ping" between two devices.  One could view it as such for active sessi=
ons, I suppose, but it really has a different focus.

That is not true.

If you negotiate the keep-alive as part of the registration procedure, the =
"pings" will be sent during the duration of the registration - no matter wh=
ether you have active sessions or not.

Now, that of course only applies to nodes that are involved in registration=
, and the keep draft currently says that keep alives negotiated during sess=
ion setup are only valid for that session. However, I think that at some po=
int there was a comment that it should also be allowed to negotiate to long=
er keep-alive durations during the session setup, for proxy-to-proxy "pings=
" etc. I guess we'll get back to that discussion once we get the keep discu=
ssion up again (next week, according to my plans).

Regards,

Christer


From christer.holmberg@ericsson.com  Sat Apr  3 22:08:10 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "'DRAGE, Keith (Keith)'" <keith.drage@alcatel-lucent.com>, Adam Roach <adam@nostrum.com>
Date: Sun, 4 Apr 2010 07:08:04 +0200
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Keith,

I do agree with what you are saying. I am not proposing another way forward=
 - I simply had some issues/questions. But, we'll learn as we walk.

Instead of having text describing the proforma rules, maybe we should addop=
t Annex A from 3GPP TS 24.229 ;)=20

Regards,

Christer

-----Original Message-----
From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]=20
Sent: 1. huhtikuuta 2010 16:24
To: Christer Holmberg; Adam Roach
Cc: SIPCORE
Subject: RE: [sipcore] Consensus Call: SIP Table 2

There are essentially two potential parts of the work here:

-	Decide on the future documentation structure
-	work out how to deal with interactions with legacy specifications.

SIPCORE is being asked only to decide on the first of these at the moment.

As Adam says, most of the existing header field usage is fully defined. The=
 problem exists with certain header fields where it appears a somewhat arbi=
trary analysis was given to the entries, and therefore the entries appear i=
nconsistent.=20

An example of this is in the RFC 3329 header fields, where I personally wou=
ld expect the semantics of these headers to be dealt with before one looks =
at the method. That means they should appear in all methods, but you will s=
ee from RFC 3329 that the headers are not valid in PRACK.

With a set of documentation rules to follow, there are no legacy issues if =
a new header field is defined. It follows one of the proforma rules, or cre=
ates a new one.

There is a legacy issue if a new method is created, but it is a long while =
since we created a new method. Yes we hit the documentation issues with the=
 update to INFO and event packages, but for existing headers we can just ca=
rry on with table 2 and 3 if we really need to. Or at that point we can cho=
ose to create a new specification that updates those RFCs that do define he=
ader fields. I would note however that that may well result in some technic=
al change, due to perceived inconsistencies in the current definitions, as =
in the example indicated above.

At the moment I do not favour the existing table method of documentation.

I do favour a set of proforma rules that specifiers can choose from.

A couple of examples of these proforma rules would be something like:

"The use of the header field is optional in all requests that are expected =
to be exchanged end-to-end between UAs. Implementations of this extension M=
UST support reception of this header field in such requests."

"The use of the header field is optional in all requests that initiate a di=
alog, or are used outside of a dialog being created. Implementations of thi=
s extension MUST support reception of this header field in such requests."

I suspect we could deal with most of the current header field usage with ab=
out 10 or so such proforma rules, and I do not expect new header fields to =
create a substantial set of new ones.

These need to specify both use (mandatory/optional) and if it is not otherw=
ise clear, support (mandatory/optional). The latter is missing from the exi=
sting tables.

regards

Keith

> -----Original Message-----
> From: sipcore-bounces@ietf.org
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: Thursday, April 01, 2010 7:45 AM
> To: Adam Roach
> Cc: SIPCORE
> Subject: Re: [sipcore] Consensus Call: SIP Table 2
>=20
>=20
> Hi,
>=20
> >Just to clarify: both alternatives will leave whatever is
> currently unclear as it is....
> >	=09
> >
> >[as individual]
> >=09
> >In truth, I don't think anything is particularly unclear --
> it's just
> >not documented the same way. The documents that didn't
> expand table 2 while adding new header fields all have prose that=20
> describes when the header field can appear. (Yes, I know this, because=20
> I looked at every one of them when I did the table 2 expansion that=20
> appears in the slides).
> >
> >So, you are saying that by reading the specs we would be
> able to complete the table you had put toghether?
> >=09
> >	[as chair]
> >=09
> >But, yes, the consensus call only deals with actions as they
> apply to
> >future documents -- such as RFC3265bis -- and does not
> address actions that would affect RFCs or any other documents that=20
> have already cleared IETF review.
> >=09
> >And, couldn't alternative 1 result in lots of "defined semantics" -=20
> >especially if we define a new method and may have to take a
> large number of header fields into consideration? Maybe you'll even=20
> end up with something that looks like Table 2...
> >
> >
> >[as individual again]
> >=09
> >I don't think so. As Keith has pointed out, statements like "both=20
> >header fields defined in this document can appear only in
> methods that
> >create new dialogs" fill out entire swaths of the table all
> at once. They also have the pleasantly future-proof quality that we=20
> don't need to say anything specifically about the header fields when=20
> someone creates a new method. A simple evaluation of the new method's=20
> properties against the header fields' normative text makes the answer=20
> brilliantly clear.
>=20
> Sure. That is fine when you define new headers, and when you define=20
> new methods and need to cover those new headers.
>=20
> But, when you define a new method, you still also need to consider all=20
> "old" existing header fields - for which we unfortunately do not have=20
> statements as those suggested by Keith.
>=20
> Take 3265bis as an example, and assume you remove table 2.=20
> The text will of course describe the usage of header fields directly=20
> associated with SUB/NOT, eg. Allow-Events. But, how will you deal with=20
> all the other header fields that we have defined over the years? I am=20
> sure there are header fields that we do NOT need to cover, because=20
> they are associated with specific mechanisms (e.g. RSeq and RAck).=20
> But, there is a huge number of more "generic" header fields that=20
> somehow need to be covered.
>=20
> Anyway, there is only one way to figure out: by doing the work :)
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> ________________________________
>=20
> 		From: sipcore-bounces@ietf.org
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> 		Sent: 30. maaliskuuta 2010 21:51
> 		To: SIPCORE
> 		Subject: [sipcore] Consensus Call: SIP Table 2
> 	=09
> 	=09
> 		[as chair]
> 	=09
> 		Last week, at the face-to-face meeting in Anaheim, the SIPCORE=20
> participants in attendance had a discussion regarding the future of=20
> Tables 2 and 3 in RFC 3261. A summary of the conversation, along with=20
> links to the presentations and an audio recording of the conversation,=20
> can be found in the minutes:
> 	=09
> 	=09
> http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP
> %20Table%202%20/%203
> 	=09
> 		During that conversation, the chairs polled those present for=20
> support between two options:
> 	=09
>=20
> 		1.	Produce an artifact stating that=20
> documents defining new header fields and methods define semantics in=20
> prose, and will no longer extend Table 2; or
> 		=09
> 		=09
> 		2.	Produce an RFC that defines new=20
> procedures for adding elements to Table 2 as new header fields and=20
> methods are defined
>=20
> 		Among those present, there was strong support for option 1. If you=20
> wish to raise an objection to this direction, please do so on the=20
> SIPCORE mailing list before April 15th.
> 	=09
> 		Thank you.
> 	=09
> 		/a
> 	=09
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From brett@broadsoft.com  Sun Apr  4 07:23:40 2010
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From: Brett Tate <brett@broadsoft.com>
To: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sun, 4 Apr 2010 07:22:45 -0700
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> As for the meaning, I'm confused with your question. =20
> The intent is to obey the Retry-After header.  If the=20
> reply to the OPTIONS has a Retry-After header, then=20
> don't send another OPTIONS until that time expires.

Your response is appropriate for responses like 486 and 500 with Retry-Afte=
r; however a 503 with Retry-After applies to more than OPTIONS.  Per RFC 32=
61 section 21.5.4, it means you SHOULD NOT send any requests (INVITE, BYE, =
ACK, CANCEL, REGISTER, ...) during that interval.


From brett@broadsoft.com  Sun Apr  4 08:37:18 2010
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From: Brett Tate <brett@broadsoft.com>
To: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sun, 4 Apr 2010 08:36:29 -0700
Thread-Topic: [sipcore] draft-jones-sip-options-ping-00: comments
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> > To help avoid receiving ignored data within 200 response, I recommend
> > use of an option-tag (such a options-ping) or something else included
> > within OPTIONS to clearly indicate that the sender does not actually
> > want all the data which can be provided within 200 response.
>=20
> I'm not opposed to this, per se, but we were trying to avoid
> introducing any new signaling features at all.  If folks thought=20
> it might be useful, we can do that. =20

Yes; it would be useful for both the sender and receiver.


> One problem, though, is that all devices would immediately be non-complia=
nt ;-)

That is okay with me.


> > Section 5.1 indicates "To avoid unduly taxing a receiving SIP=20
> > entity, transmitters of OPTIONS messages MUST honor the=20
> > Retry-After header field if received."  The MUST requirement=20
> > is likely too strong because of implications when included=20
> > within a 503.  The MUST requirement is likely also too strong=20
> > because of vulnerability to quit sending requests to that device=20
> > for an extremely long time because attacker or bug caused the=20
> > 503 to be sent containing Retry-After with an extremely
> > large value.
>=20
> Perhaps you're right, but do you have suggestions for alternative
> wording?  Are you suggesting to replace "MUST" with "SHOULD"?

The current wording requires honoring the 503 with Retry-After overload mec=
hanism.  RFC 3261 section 21.5.4 is at the SHOULD level instead of a MUST. =
=20

If you only intended the Retry-After to apply to OPTIONS (instead of all re=
quests), you can't use a 503 response (unless maybe you used an option-tag =
or something to indicate a meaning other than RFC 3261 section 21.5.4).


> > Concerning receiving 503 with Retry-After, to what does it apply:
> > 1) OPTIONS request target ip-address,
> > 2) OPTIONS response source ip-address,
> > 3) both 1 and 2,
> > 4) something else.
>=20
> I'm a little confused by your question. Are you asking whether=20
> it applies to the target address to which the sender of the=20
> OPTIONS sent the request or to the address from which the=20
> recipient returned a 503, or the address of an intermediary?

I'm not referring to intermediaries; however yes to other parts of the ques=
tion (although also asking if applies to more or both). =20

I'm referring to devices with multiple ip-addresses such as 1 or more IPv4 =
addresses and/or 1 or more IPv6 addresses.  Thus the source ip-address of 5=
03 response may be different from where UAC sent the OPTIONS; and the sende=
r of 503 may have additional ip-addresses used for SIP communication betwee=
n these two servers.  Since rfc3261 was somewhat vague, I'm attempting to e=
nsure the draft is clear since it is requiring or recommending the use of 5=
03 with Retry-After.


From christer.holmberg@ericsson.com  Mon Apr  5 04:06:32 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Mon, 5 Apr 2010 13:06:25 +0200
Thread-Topic: rfc3265bis: SIP events redux [was [sipcore] Minutes Posted]
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Subject: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Hi,

For the 3265bis thing the following is stated:

"Unless there are objections on the list, the draft will go into WGLC with =
these changes."

I think there are still a few issues we need to solve before we should go t=
o WGLC.

1. As indicated in the meeting (also stated in the report), we need to disc=
uss when REFER implicit subscriptions are created - on REFER 200 OK or on N=
OTIFY.

2. There is still an open issue in the draft whether timer N applies to pro=
xies. I may have missed it, but I don't see any conclusion on that.=20

3. A while ago there was a discussion on what happens in case timer N expir=
es. There was a suggestion that the subscription would be terminated, but I=
 am not sure whether there is a decission on that. Also, I think that is im=
pacted by the etag issue I raised on the list a few days ago ('RFC3265/3265=
bis and draft-ietf-sipcore-subnot-etags issue regarding notification suppre=
ssion').

If Adam's agenda allows it, having dedicated phone meetings to discuss thes=
e issue could be a way to move things forward, in order to be able to initi=
ate the WGLC.

Regards,

Christer



________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Adam=
 Roach [adam@nostrum.com]
Sent: Tuesday, March 30, 2010 9:38 PM
To: SIPCORE
Subject: [sipcore] Minutes Posted

[as chair]

A draft version of the minutes from the recent IETF meeting in Anaheim
has been posted to the proceedings site:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html

Please review the minutes and submit any corrections to the chairs no
later than April 15th. Thank you.

/a
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore=

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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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[as an individual]

Paul E. Jones wrote:

>> Other, specific comments on the draft:
>> 0) You might want to consider specifying what the To/From-URIs should
>> be sent as, and that a receiver should be lenient.  We've seen interop
>> problems with those URIs for OPTIONS pings.
> 
> I wanted to be rather strict in saying that only sip:192.168.4.56 was
> acceptable, for example, but others I consulted with thought that was too
> restrictive.  So, the language is soft here.

I believe I was one of the "others".

My personal concern was that making such a mandate starts to interfere 
with the layering of the sip stack. If you "know" your neighbors via DNS 
names, then the translation to ip addresses occurs in the code that 
implements 3263, which is way below where you construct, send, and 
receive responses to messages.

I didn't want to see a set of recommendations that require violation of 
such layering.

But I'd be interested in hearing other points of view on this.

	Thanks,
	Paul

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Cc: SIPCORE <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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[as individual]

Christer Holmberg wrote:
> Hi, 
> 
>>>> IMO this is one of the big problems with the IMS approach to services. 
>>>> They aren't registered, and aren't anything that the S-CSCF could (or
>>>> would) return as 3xx alternatives. So the caller will be limited in what options are visible to it.
>>> I am not really sure I understand what you mean...
>> For instance, suppose I want to talk to your voicemail, without ringing your regular phone(s). I might send an OPTIONS in hope of getting a list of Contacts, among which the VM contact might be found.
> 
> As I also indicated in my reply to Hadriel's e-mail, for this case I think it would be very useful to have a VM feature tag, or something else in the returned contact that indicates voicemail.

Its already there, defined in 3840: actor="msg-taker".

>> AFAIK, IMS isn't set up in a way to have an identifiable contact for the VM server that it could return in the 3xx. (But I'll be pleasantly surprised to learn I'm wrong.)
> 
> IMS uses 3841.

Yeah, I know that it uses 3841, but the last I knew it only did so to 
select among the registered contacts, after it has already applied all 
of the configured services. And its my impression that VM is applied as 
a service, that sits in the signaling path, observes the responses from 
UAs, and then eventually routes to VM (using a configured URI). I don't 
see how 3841 preferences (either to prefer or reject VM) would ever be 
operable in this environment.

But I'd be pleased to hear that I'm wrong about this.

	Thanks,
	Paul

From pkyzivat@cisco.com  Mon Apr  5 06:36:31 2010
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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[as individual]

Christer Holmberg wrote:

> 3. A while ago there was a discussion on what happens in case timer N expires. 
>    There was a suggestion that the subscription would be terminated,
>    but I am not sure whether there is a decission on that. 

Are you asking about *initial* subscriptions, or resubscribes?

For initial subscription requests, there is no subscription to terminate.

For resubscribes, its currently an open issue in section B.3.

Are you just asking to have the open issue resolved?

	Thanks,
	Paul

From adam@nostrum.com  Mon Apr  5 08:07:02 2010
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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On 4/5/10 8:36 AM, Paul Kyzivat wrote:
> [as individual]
>
> Christer Holmberg wrote:
>
>> 3. A while ago there was a discussion on what happens in case timer N 
>> expires.    There was a suggestion that the subscription would be 
>> terminated,
>>    but I am not sure whether there is a decission on that. 
>
> Are you asking about *initial* subscriptions, or resubscribes?
>
> For initial subscription requests, there is no subscription to terminate.
>
> For resubscribes, its currently an open issue in section B.3.
>
> Are you just asking to have the open issue resolved?

[as a document author]

We had discussion around this point in Stockholm, and drove it to a 
conclusion on list afterwards. I put it up in Anaheim one more time, 
with a summary of what that conversation concluded. If someone wants to 
raise an objection to the decision to treat Timer N expiration the same 
way we treat SUBSCRIBE transaction timeouts, I'm happy to engage in that 
discussion.

/a

From adam@nostrum.com  Mon Apr  5 08:16:28 2010
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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On 4/5/10 6:06 AM, Christer Holmberg wrote:
> Hi,
>
> For the 3265bis thing the following is stated:
>
> "Unless there are objections on the list, the draft will go into WGLC with these changes."
>
> I think there are still a few issues we need to solve before we should go to WGLC.
>    

[as document author]

I think you're misinterpreting that statement. It's not saying that the 
document is ready for WGLC -- the proposed resolution for the various 
issues still needs to go into the document. The point of the note is 
that, barring any concrete objections to the proposed resolutions, I'll 
be updating the document to reflect those resolutions, and will treat 
them as closed. In other words, it's the last time I plan on bringing 
them up for discussion. I consider a yearlong effort of waving these 
issues at the working group to be sufficient time to let anyone's 
concerns be expressed.

One point I would like you to clarify:
> 2. There is still an open issue in the draft whether timer N applies to proxies. I may have missed it, but I don't see any conclusion on that.
>    

Where do you see this in the draft?

In any case, proxies don't inherently care about subscription state. If 
you're doing something in your proxy that is attempting to figure out 
the state of a subscription, you need to take care to figure out what's 
going on. In that case, yeah, Timer N is going to be of interest to you. 
But that's not something we need to specify -- it's actually very 
application-specific.

/a

From christer.holmberg@ericsson.com  Mon Apr  5 12:02:56 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 5 Apr 2010 21:02:50 +0200
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Cc: SIPCORE <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

>>>>>> IMO this is one of the big problems with the IMS approach to service=
s.
>>>>> They aren't registered, and aren't anything that the S-CSCF could (or
>>>>> would) return as 3xx alternatives. So the caller will be limited in w=
hat options are visible to it.
>>>> I am not really sure I understand what you mean...
>>> For instance, suppose I want to talk to your voicemail, without ringing=
 your regular phone(s). I might send an OPTIONS in hope of getting a list o=
f Contacts, among which the VM contact might be found.
>>
>> As I also indicated in my reply to Hadriel's e-mail, for this case I thi=
nk it would be very useful to have a VM feature tag, or something else in t=
he returned contact that indicates voicemail.
>
>Its already there, defined in 3840: actor=3D"msg-taker".

True.

>>>AFAIK, IMS isn't set up in a way to have an identifiable contact for the=
 VM server that it could return in the 3xx. (But I'll be pleasantly surpris=
ed to learn I'm wrong.)
>>
>>IMS uses 3841.
>
>Yeah, I know that it uses 3841, but the last I knew it only did so to sele=
ct among the registered contacts, after it has already applied all of the c=
onfigured services.
>
>And its my impression that VM is applied as a service, that sits in the si=
gnaling path, observes the responses from
>UAs, and then eventually routes to VM (using a configured URI). I don't se=
e how 3841 preferences (either to prefer or reject VM) would ever be
>operable in this environment.

As far as I know you can use ANY information, including feature tag values,=
 to choose the application servers to which an initial request is to be rou=
ted (the procedure is called initial filter criteria in IMS language) - or =
decide not to route the request to any ASs. And, I don't think there is any=
thing in the IMS specifications that forbids a user to have a "local" voice=
 mail service.

Regards,

Christer=

From christer.holmberg@ericsson.com  Mon Apr  5 12:14:14 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 5 Apr 2010 21:14:09 +0200
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Hi,

>>>3. A while ago there was a discussion on what happens in case timer N
>>>expires.    There was a suggestion that the subscription would be
>>>terminated, but I am not sure whether there is a decission on that.
>>
>>Are you asking about *initial* subscriptions, or resubscribes?

Resubscribes.

>>For initial subscription requests, there is no subscription to terminate.

Agree.

>>For resubscribes, its currently an open issue in section B.3.

Yes.

>>Are you just asking to have the open issue resolved?

It seems like I missunderstood the report regarding WGLC (see Adam's e-mail=
), so I said that in my opinion we should solve the issue before we initiat=
e WGLC.

>[as a document author]
>
>We had discussion around this point in Stockholm, and drove it to a
>conclusion on list afterwards. I put it up in Anaheim one more time,
>with a summary of what that conversation concluded. If someone wants to
>raise an objection to the decision to treat Timer N expiration the same
>way we treat SUBSCRIBE transaction timeouts, I'm happy to engage in that
>discussion.

One alternative would be to fallback whatever state was used before the SUB=
SCRIBE was sent. However, I don't really have any strong opinions about tha=
t as such.

My issue was that, IF there are intermediaries that maintain subscription s=
tate, and they don't support subnot, they could terminate the subscription =
in cases where the NOTIFY is suppressed. Of course, in case we would choose=
 that timer N expiration would cause a fallback there could be some state u=
nsynchs between entities.

So, a related question to that was: since the NOTIFY request is so essentia=
l to 3265bis, should we allow it to be suppressed in the first place?=20

Regards,

Christer=

From christer.holmberg@ericsson.com  Mon Apr  5 12:37:25 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>
Date: Mon, 5 Apr 2010 21:37:20 +0200
Thread-Topic: rfc3265bis: SIP events redux [was [sipcore] Minutes Posted]
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Hi,

>>2. There is still an open issue in the draft whether timer N applies to p=
roxies. I may have missed it, but I don't see any conclusion on that.
>
>
>Where do you see this in the draft?

It is not indicated in the draft. I meant that it is one of the issues I ha=
ve.

>In any case, proxies don't inherently care about subscription state. If
>you're doing something in your proxy that is attempting to figure out
>the state of a subscription, you need to take care to figure out what's
>going on. In that case, yeah, Timer N is going to be of interest to you.
>But that's not something we need to specify -- it's actually very
>application-specific.

The draft says that proxies no not need to support anything in addition to =
3261, so from that point of view you are correct.

However, at the same time it talks about proxies adding Record-Route if the=
y are interested in the SUBs and NOTs, so I assumed that means that they ar=
e "allowed" to actually maintain subscription dialog state - in the same wa=
y as they might maintain invite dialog state.

Regards,

Christer=

From HKaplan@acmepacket.com  Mon Apr  5 14:31:55 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Mon, 5 Apr 2010 17:31:50 -0400
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One problem I've seen is when something like a UA or IP-PBX is OPTIONS ping=
ing "sip:sip.ssp.com" - that will resolve in DNS to an ingress proxy, which=
 actually is the real target of the ping request - but the ingress proxy it=
self may think the OPTIONS needs to get routed to the authoritative server =
for sip.ssp.com, which may well be something _after_ the proxy. =20

Setting max-forwards to 1 helps avoid that, but in some proxies that will g=
et back a 483 too many hops, which is actually ok in my book - after all yo=
u know its alive and processing requests - but in some devices that increme=
nts error counters which people don't like.

-hadriel


> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Monday, April 05, 2010 8:52 AM
> To: Paul E. Jones
> Cc: Hadriel Kaplan; sipcore@ietf.org; 'Vivek Bhargava (vbharga)'; 'Gonzal=
o
> Salgueiro'
> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
>=20
> [as an individual]
>=20
> Paul E. Jones wrote:
>=20
> >> Other, specific comments on the draft:
> >> 0) You might want to consider specifying what the To/From-URIs should
> >> be sent as, and that a receiver should be lenient.  We've seen interop
> >> problems with those URIs for OPTIONS pings.
> >
> > I wanted to be rather strict in saying that only sip:192.168.4.56 was
> > acceptable, for example, but others I consulted with thought that was
> too
> > restrictive.  So, the language is soft here.
>=20
> I believe I was one of the "others".
>=20
> My personal concern was that making such a mandate starts to interfere
> with the layering of the sip stack. If you "know" your neighbors via DNS
> names, then the translation to ip addresses occurs in the code that
> implements 3263, which is way below where you construct, send, and
> receive responses to messages.
>=20
> I didn't want to see a set of recommendations that require violation of
> such layering.
>=20
> But I'd be interested in hearing other points of view on this.
>=20
> 	Thanks,
> 	Paul

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On Apr 5, 2010, at 8:03 AM, Paul Kyzivat wrote:
> g-taker".
>
>>> AFAIK, IMS isn't set up in a way to have an identifiable contact  
>>> for the VM server that it could return in the 3xx. (But I'll be  
>>> pleasantly surprised to learn I'm wrong.)
>> IMS uses 3841.
>
> Yeah, I know that it uses 3841, but the last I knew it only did so  
> to select among the registered contacts, after it has already  
> applied all of the configured services. And its my impression that  
> VM is applied as a service, that sits in the signaling path,  
> observes the responses from UAs, and then eventually routes to VM  
> (using a configured URI). I don't see how 3841 preferences (either  
> to prefer or reject VM) would ever be operable in this environment.

AFAIK, the AS MAY divert the INVITE to the configured VM URI if there  
is one, the user's profile allows this, and the request indicates a  
preference for VM.
--
Dean



From pkyzivat@cisco.com  Mon Apr  5 16:10:11 2010
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Cc: SIPCORE <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Dean Willis wrote:
> 
> On Apr 5, 2010, at 8:03 AM, Paul Kyzivat wrote:
>> g-taker".
>>
>>>> AFAIK, IMS isn't set up in a way to have an identifiable contact for 
>>>> the VM server that it could return in the 3xx. (But I'll be 
>>>> pleasantly surprised to learn I'm wrong.)
>>> IMS uses 3841.
>>
>> Yeah, I know that it uses 3841, but the last I knew it only did so to 
>> select among the registered contacts, after it has already applied all 
>> of the configured services. And its my impression that VM is applied 
>> as a service, that sits in the signaling path, observes the responses 
>> from UAs, and then eventually routes to VM (using a configured URI). I 
>> don't see how 3841 preferences (either to prefer or reject VM) would 
>> ever be operable in this environment.
> 
> AFAIK, the AS MAY divert the INVITE to the configured VM URI if there is 
> one, the user's profile allows this, and the request indicates a 
> preference for VM.

My *impression* is that the AS would receive the call initially, note 
that VM is configured, set a timer for itself, Record-Route itself, and 
forward the call back to the S-CSCF. Eventually the S-CSCF will get 
around to considering registered contacts. If it honors 
Request-Disposition:redirect, then it will return a 3xx with the 
registered contacts.

That 3xx will be received by the AS. If it is clever, it will realize 
the best thing it can do is add its VM contact to the 3xx, with a lower 
priority than all the others.

If so, then the caller should indeed see all the possibilities, with 
their capabilities, and be able to make its own choice.

That's actually the best outcome I have ever seen for this with IMS.
I hope it all works that way. Without Request-Disposition:redirect I 
have not seen how this would ever work right for IMS.

Chalk up another success for redirect rather than forking.

	Thanks,
	Paul

From dean.willis@softarmor.com  Mon Apr  5 17:09:10 2010
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Cc: SIPCORE <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On Apr 5, 2010, at 6:09 PM, Paul Kyzivat wrote:
>>
>
> My *impression* is that the AS would receive the call initially,  
> note that VM is configured, set a timer for itself, Record-Route  
> itself, and forward the call back to the S-CSCF. Eventually the S- 
> CSCF will get around to considering registered contacts. If it  
> honors Request-Disposition:redirect, then it will return a 3xx with  
> the registered contacts.
>
> That 3xx will be received by the AS. If it is clever, it will  
> realize the best thing it can do is add its VM contact to the 3xx,  
> with a lower priority than all the others.
>
> If so, then the caller should indeed see all the possibilities, with  
> their capabilities, and be able to make its own choice.
>
> That's actually the best outcome I have ever seen for this with IMS.
> I hope it all works that way. Without Request-Disposition:redirect I  
> have not seen how this would ever work right for IMS.
>
> Chalk up another success for redirect rather than forking.

Hmm.I thought that the S-CSCF would chain the INVITE through a series  
of AS, and if an AS's service needed to, it would just divert the call  
to VM instead of sending it to the next AS. At least that's the way we  
wrote the stuff originally, and it's how Dynamicsoft's stuff worked.

In other words, an AS can generate the 200 OK on an initial INVITE,  
short-circuiting the other ASs (as well as any Registered Contacts)  
that might have been invoked by the profile. This 200OK follows the  
inverted route back through any other AS that had been invoked on the  
call, on to the S-CSCF, and back towards the originating UA.

So, for example if the first AS invoked is set up to do a "Busy  
Everywhere", that's what gets returned to the caller, no matter what  
other AS might have been in the profile or what contacts might be  
registered. It makes for some careful service composition logic -- yo  
ucan shoot yourself in the foot -- but the composition mechanics are a  
lot easier than what I think you;re describing.

Did somebody change this stuff while I wasn't looking?

--
Dean


From christer.holmberg@ericsson.com  Mon Apr  5 22:35:24 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 6 Apr 2010 07:35:17 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: SIPCORE <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>>My *impression* is that the AS would receive the call initially, note=20
>>that VM is configured, set a timer for itself, Record-Route itself,=20
>>and forward the call back to the S-CSCF. Eventually the S-=20
>>CSCF will get around to considering registered contacts. If it honors=20
>>Request-Disposition:redirect, then it will return a 3xx with the=20
>>registered contacts.
>>
>>That 3xx will be received by the AS. If it is clever, it=20
>>will realize the best thing it can do is add its VM contact to the 3xx, w=
ith a=20
>>lower priority than all the others.
>>
>>If so, then the caller should indeed see all the possibilities, with=20
>>their capabilities, and be able to make its own choice.
>>
>>That's actually the best outcome I have ever seen for this with IMS.
>>I hope it all works that way. Without Request-Disposition:redirect I=20
>>have not seen how this would ever work right for IMS.
>>
>>Chalk up another success for redirect rather than forking.
>=20
>Hmm.I thought that the S-CSCF would chain the INVITE through=20
>a series of AS, and if an AS's service needed to, it would=20
>just divert the call to VM instead of sending it to the next=20
>AS. At least that's the way we wrote the stuff originally,=20
>and it's how Dynamicsoft's stuff worked.
>=20
>In other words, an AS can generate the 200 OK on an initial=20
>INVITE, short-circuiting the other ASs (as well as any=20
>Registered Contacts) that might have been invoked by the=20
>profile. This 200OK follows the inverted route back through=20
>any other AS that had been invoked on the call, on to the=20
>S-CSCF, and back towards the originating UA.
>=20
>So, for example if the first AS invoked is set up to do a=20
>"Busy Everywhere", that's what gets returned to the caller,=20
>no matter what other AS might have been in the profile or=20
>what contacts might be registered. It makes for some careful=20
>service composition logic -- yo ucan shoot yourself in the=20
>foot -- but the composition mechanics are a lot easier than=20
>what I think you;re describing.
>=20
>Did somebody change this stuff while I wasn't looking?

You haven't been looking for a long time :)

Anyway, I don't think it has been changed. But, I think you are talking abo=
ut an INVITE 3xx use-case, not an OPTIONS 3xx use-case.

OPTIONS is used in order to query capabilities, and I think it is a policy/=
implementation issue whether the operator wants to return contacts pointing=
 to specific services.

Regards,

Christer

From pkyzivat@cisco.com  Tue Apr  6 05:49:30 2010
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Cc: SIPCORE <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Dean Willis wrote:
> 
> On Apr 5, 2010, at 6:09 PM, Paul Kyzivat wrote:
>>>
>>
>> My *impression* is that the AS would receive the call initially, note 
>> that VM is configured, set a timer for itself, Record-Route itself, 
>> and forward the call back to the S-CSCF. Eventually the S-CSCF will 
>> get around to considering registered contacts. If it honors 
>> Request-Disposition:redirect, then it will return a 3xx with the 
>> registered contacts.
>>
>> That 3xx will be received by the AS. If it is clever, it will realize 
>> the best thing it can do is add its VM contact to the 3xx, with a 
>> lower priority than all the others.
>>
>> If so, then the caller should indeed see all the possibilities, with 
>> their capabilities, and be able to make its own choice.
>>
>> That's actually the best outcome I have ever seen for this with IMS.
>> I hope it all works that way. Without Request-Disposition:redirect I 
>> have not seen how this would ever work right for IMS.
>>
>> Chalk up another success for redirect rather than forking.
> 
> Hmm.I thought that the S-CSCF would chain the INVITE through a series of 
> AS, and if an AS's service needed to, it would just divert the call to 
> VM instead of sending it to the next AS. At least that's the way we 
> wrote the stuff originally, and it's how Dynamicsoft's stuff worked.

My knowledge is probably no more recent than yours.
AFAIK it *could* do that. But in the case of VM, it first has to know 
that VM is required on the call. Normally that doesn't happen until the 
registered contacts are tried. And VM isn't one of the registered contacts.

So I *think* the AS just lurks as a proxy to see what else will happen, 
and then later might divert the call to VM.

> In other words, an AS can generate the 200 OK on an initial INVITE, 
> short-circuiting the other ASs (as well as any Registered Contacts) that 
> might have been invoked by the profile. This 200OK follows the inverted 
> route back through any other AS that had been invoked on the call, on to 
> the S-CSCF, and back towards the originating UA.

It could do that if it was able to conclude that VM was the desired 
outcome without trying any of the other candidates. But how would it 
figure that out?

> So, for example if the first AS invoked is set up to do a "Busy 
> Everywhere", that's what gets returned to the caller, no matter what 
> other AS might have been in the profile or what contacts might be 
> registered. It makes for some careful service composition logic -- yo 
> ucan shoot yourself in the foot -- but the composition mechanics are a 
> lot easier than what I think you;re describing.
> 
> Did somebody change this stuff while I wasn't looking?

Probably not.

But either we should stop talking about it here, or else we should get 
somebody who really knows to tell us the answer.

	Thanks,
	Paul

From Martin.Thomson@andrew.com  Mon Apr  5 22:26:13 2010
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From: "Thomson, Martin" <Martin.Thomson@andrew.com>
To: Simple WG <simple@ietf.org>, "Peterson, Jon" <jon.peterson@neustar.biz>, "James M. Polk" <jmpolk@cisco.com>
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Tue, 6 Apr 2010 16:09:16 +0200
Thread-Topic: [sipcore] Consensus Call: SIP Table 2
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--_000_A444A0F8084434499206E78C106220CADE203FD4MCHP058Aglobala_
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No objection.

John

________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Adam Roach
Sent: 30 March 2010 19:51
To: SIPCORE
Subject: [sipcore] Consensus Call: SIP Table 2

[as chair]

Last week, at the face-to-face meeting in Anaheim, the SIPCORE participants=
 in attendance had a discussion regarding the future of Tables 2 and 3 in R=
FC 3261. A summary of the conversation, along with links to the presentatio=
ns and an audio recording of the conversation, can be found in the minutes:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Table%202%=
20/%203

During that conversation, the chairs polled those present for support betwe=
en two options:

 1.  Produce an artifact stating that documents defining new header fields =
and methods define semantics in prose, and will no longer extend Table 2; o=
r

 2.  Produce an RFC that defines new procedures for adding elements to Tabl=
e 2 as new header fields and methods are defined

Among those present, there was strong support for option 1. If you wish to =
raise an objection to this direction, please do so on the SIPCORE mailing l=
ist before April 15th.

Thank you.

/a

--_000_A444A0F8084434499206E78C106220CADE203FD4MCHP058Aglobala_
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<BODY text=3D#000000 bgColor=3D#ffffff>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D156050914-06042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>No objection.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D156050914-06042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D156050914-06042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>John</FONT></SPAN></DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sipcore-bounces@ietf.org=20
  [mailto:sipcore-bounces@ietf.org] <B>On Behalf Of </B>Adam=20
  Roach<BR><B>Sent:</B> 30 March 2010 19:51<BR><B>To:</B>=20
  SIPCORE<BR><B>Subject:</B> [sipcore] Consensus Call: SIP Table=20
  2<BR></FONT><BR></DIV>
  <DIV></DIV>[as chair]<BR><BR>Last week, at the face-to-face meeting in=20
  Anaheim, the SIPCORE participants in attendance had a discussion regardin=
g the=20
  future of Tables 2 and 3 in RFC 3261. A summary of the conversation, alon=
g=20
  with links to the presentations and an audio recording of the conversatio=
n,=20
  can be found in the minutes:<BR><BR><A class=3Dmoz-txt-link-freetext=20
  href=3D"http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20=
Table%202%20/%203">http://www.ietf.org/proceedings/10mar/minutes/sipcore.ht=
ml#SIP%20Table%202%20/%203</A><BR><BR>During=20
  that conversation, the chairs polled those present for support between tw=
o=20
  options:<BR>
  <OL>
    <LI>Produce an artifact stating that documents defining new header fiel=
ds=20
    and methods define semantics in prose, and will no longer extend Table =
2;=20
    or<BR><BR>
    <LI>Produce an RFC that defines new procedures for adding elements to T=
able=20
    2 as new header fields and methods are defined </LI></OL>Among those pr=
esent,=20
  there was strong support for option 1. If you wish to raise an objection =
to=20
  this direction, please do so on the SIPCORE mailing list before April=20
  15th.<BR><BR>Thank you.<BR><BR>/a<BR></BLOCKQUOTE></BODY></HTML>

--_000_A444A0F8084434499206E78C106220CADE203FD4MCHP058Aglobala_--

From pkyzivat@cisco.com  Tue Apr  6 07:16:43 2010
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Subject: Re: [sipcore] Consensus Call: SIP Table 2
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+1

Elwell, John wrote:
> No objection.
>  
> John
> 
>     ------------------------------------------------------------------------
>     *From:* sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org]
>     *On Behalf Of *Adam Roach
>     *Sent:* 30 March 2010 19:51
>     *To:* SIPCORE
>     *Subject:* [sipcore] Consensus Call: SIP Table 2
> 
>     [as chair]
> 
>     Last week, at the face-to-face meeting in Anaheim, the SIPCORE
>     participants in attendance had a discussion regarding the future of
>     Tables 2 and 3 in RFC 3261. A summary of the conversation, along
>     with links to the presentations and an audio recording of the
>     conversation, can be found in the minutes:
> 
>     http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Table%202%20/%203
> 
>     During that conversation, the chairs polled those present for
>     support between two options:
> 
>        1. Produce an artifact stating that documents defining new header
>           fields and methods define semantics in prose, and will no
>           longer extend Table 2; or
> 
>        2. Produce an RFC that defines new procedures for adding elements
>           to Table 2 as new header fields and methods are defined 
> 
>     Among those present, there was strong support for option 1. If you
>     wish to raise an objection to this direction, please do so on the
>     SIPCORE mailing list before April 15th.
> 
>     Thank you.
> 
>     /a
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From pkyzivat@cisco.com  Tue Apr  6 07:18:02 2010
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Subject: Re: [sipcore] Consensus Call: SIP Table 2
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[this was as an individual]

Paul Kyzivat wrote:
> +1
> 
> Elwell, John wrote:
>> No objection.
>>  
>> John
>>
>>     
>> ------------------------------------------------------------------------
>>     *From:* sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org]
>>     *On Behalf Of *Adam Roach
>>     *Sent:* 30 March 2010 19:51
>>     *To:* SIPCORE
>>     *Subject:* [sipcore] Consensus Call: SIP Table 2
>>
>>     [as chair]
>>
>>     Last week, at the face-to-face meeting in Anaheim, the SIPCORE
>>     participants in attendance had a discussion regarding the future of
>>     Tables 2 and 3 in RFC 3261. A summary of the conversation, along
>>     with links to the presentations and an audio recording of the
>>     conversation, can be found in the minutes:
>>
>>     
>> http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Table%202%20/%203 
>>
>>
>>     During that conversation, the chairs polled those present for
>>     support between two options:
>>
>>        1. Produce an artifact stating that documents defining new header
>>           fields and methods define semantics in prose, and will no
>>           longer extend Table 2; or
>>
>>        2. Produce an RFC that defines new procedures for adding elements
>>           to Table 2 as new header fields and methods are defined
>>     Among those present, there was strong support for option 1. If you
>>     wish to raise an objection to this direction, please do so on the
>>     SIPCORE mailing list before April 15th.
>>
>>     Thank you.
>>
>>     /a
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> 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 brett@broadsoft.com  Tue Apr  6 10:56:41 2010
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From: Brett Tate <brett@broadsoft.com>
To: Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Tue, 6 Apr 2010 10:55:43 -0700
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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> If someone wants to raise an objection to the decision to=20
> treat Timer N expiration the same way we treat SUBSCRIBE=20
> transaction timeouts, I'm happy to engage in that discussion.

The decision has an interesting interaction concerning overload.  More spec=
ifically if SUBSCRIBE received from device that the notifier still thinks i=
s in overload, notifier will send 200 response but not the NOTIFY until lat=
er.  Or an overloaded middle box throttling some traffic might reject the N=
OTIFY with 500/503 Retry-After.

Concerning the overload interactions, is the decision concerning Timer N ex=
piration acceptable since not allowing SUBSCRIBE 200 response to confirm th=
e subscription creation?


From dworley@avaya.com  Tue Apr  6 12:31:52 2010
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From: "WORLEY, DALE R (DALE)" <dworley@avaya.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Adam Roach <adam@nostrum.com>
Date: Tue, 6 Apr 2010 15:26:04 -0400
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Chri=
ster Holmberg [christer.holmberg@ericsson.com]

>In any case, proxies don't inherently care about subscription state. If
>you're doing something in your proxy that is attempting to figure out
>the state of a subscription, you need to take care to figure out what's
>going on. In that case, yeah, Timer N is going to be of interest to you.
>But that's not something we need to specify -- it's actually very
>application-specific.

The draft says that proxies no not need to support anything in addition to =
3261, so from that point of view you are correct.

However, at the same time it talks about proxies adding Record-Route if the=
y are interested in the SUBs and NOTs, so I assumed that means that they ar=
e "allowed" to actually maintain subscription dialog state - in the same wa=
y as they might maintain invite dialog state.
_______________________________________________

Even if a proxy is keeping subscription-state information, I don't see Time=
r N introducing a problem.  The proxy sees all the SUBSCRIBEs and NOTIFYs a=
nd their responses.  Under 3265bis, the NOTIFYs, actually, the the *respons=
es* to the NOTIFYs show all of the subscription state.  In particular, in s=
eciton 4.1.2.4, I see:

   Until Timer N expires, several NOTIFY messages may arrive from
   different destinations (see Section 4.4.1).  Each of these messages
   establish a new dialog and a new subscription.  After the expiration
   of Timer N, the subscriber SHOULD reject any such NOTIFY messages
   that would otherwise establish a new dialog with a "481" response
   code.

Dale

From dworley@avaya.com  Tue Apr  6 13:01:31 2010
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From: "WORLEY, DALE R (DALE)" <dworley@avaya.com>
To: Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 6 Apr 2010 16:01:26 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Salv=
atore Loreto [salvatore.loreto@ericsson.com]
Sent: Thursday, April 01, 2010 3:36 AM

A good mechanism to discovery UA capabilities is essential to provide meani=
ngful service:
So I am in favor of pursuing the work,
in particular the possibility to use a single OPTIONS request to retrieve t=
he capabilities from ALL UAS(s) behind a forking proxy.
_______________________________________

I think it is a misconception to use UA discovery to provide services.  The=
 difficulty is that the set of available UAs for a given destination URI ch=
anges unpredictably -- elements become inacessible (mobile phone out of ran=
ge!), call routing changes due to time of day, etc.  Instead, SIP provides =
a remarkably effective system for selecting the best target of a call *at t=
he moment the call is initiated* -- the caller declares his preferences, an=
d the various intermediate elements combine the caller's desires with the r=
outing information that they possess to forward the call closer to the pref=
erred destination.

As a developer, I've had to wrestle with this.  In order to provide clean "=
Is extension 1234 busy?" information, our "BLF agent" maintains subscriptio=
ns to the "reg" event package for the AOR "sip:1234@example.com".  For each=
 reported contact for the AOR, it maintains a dialog subscription to the co=
ntact.  Only by tracking the "reg" events can it keep track of the UAs avai=
lable for that AOR.  And even then, it is at the mercy of network failures.

Dale

From christer.holmberg@ericsson.com  Tue Apr  6 13:02:31 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Adam Roach <adam@nostrum.com>
Date: Tue, 6 Apr 2010 22:02:25 +0200
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Hi,

>>In any case, proxies don't inherently care about subscription state. If
>>you're doing something in your proxy that is attempting to figure out
>>the state of a subscription, you need to take care to figure out what's
>>going on. In that case, yeah, Timer N is going to be of interest to you.
>>But that's not something we need to specify -- it's actually very
>>application-specific.
>
>The draft says that proxies no not need to support anything in addition to=
 3261, so from that point of view you are correct.
>
>However, at the same time it talks about proxies adding Record-Route if th=
ey are interested in the SUBs and NOTs, so I assumed that means that they a=
re "allowed" to actually maintain subscription dialog state - in the same w=
ay as they might maintain invite dialog state.
>_______________________________________________
>
>Even if a proxy is keeping subscription-state information, I don't see Tim=
er N introducing a problem.  The proxy sees all the SUBSCRIBEs and NOTIFYs =
and their responses.  Under 3265bis, the NOTIFYs, actually, the the *respon=
ses* to the NOTIFYs show all of the=20
>subscription state.  In particular, in seciton 4.1.2.4, I see:
>
>   Until Timer N expires, several NOTIFY messages may arrive from
>   different destinations (see Section 4.4.1).  Each of these messages
>   establish a new dialog and a new subscription.  After the expiration
>   of Timer N, the subscriber SHOULD reject any such NOTIFY messages
>   that would otherwise establish a new dialog with a "481" response
>   code.

Exactly, and that's why I think draft-ietf-sipcore-subnot-etags could cause=
 issues, because it allows the NOTIFY to be suppressed. Now, if the proxy d=
oes not support that draft, and it doesn't receive a NOTIFY (since it has b=
een suppressed) it could terminate the subsciption dialog state.

Please also note that my issue is related to re/de-subscriptions, when all =
dialogs already exist.

Regards,

Christer=

From christer.holmberg@ericsson.com  Tue Apr  6 13:07:23 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Brett Tate <brett@broadsoft.com>, Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Tue, 6 Apr 2010 22:04:58 +0200
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Hi,

Overload, that would "delay" the NOTIFY, is in fact one of the scenarios I =
also had in mind, and got me thinking whether it would not simply be better=
 to maintain the pre-SUB dialog state, instead of terminating the dialog.

Regards,

Christer

________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Bret=
t Tate [brett@broadsoft.com]
Sent: Tuesday, April 06, 2010 8:55 PM
To: Adam Roach; SIPCORE
Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]

> If someone wants to raise an objection to the decision to
> treat Timer N expiration the same way we treat SUBSCRIBE
> transaction timeouts, I'm happy to engage in that discussion.

The decision has an interesting interaction concerning overload.  More spec=
ifically if SUBSCRIBE received from device that the notifier still thinks i=
s in overload, notifier will send 200 response but not the NOTIFY until lat=
er.  Or an overloaded middle box throttling some traffic might reject the N=
OTIFY with 500/503 Retry-After.

Concerning the overload interactions, is the decision concerning Timer N ex=
piration acceptable since not allowing SUBSCRIBE 200 response to confirm th=
e subscription creation?

_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore=

From christer.holmberg@ericsson.com  Tue Apr  6 13:10:03 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 6 Apr 2010 22:08:04 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi Dale,

>From my perspective, the main use-case is not to discover services, but to =
discover capabilities in order to determine what services can be applied.

The idea is also that the OPTIONS request would be sent prior to the INVITE=
, to try to make sure that the returned contacts are actually available.

Regards,

Christer

________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of WORL=
EY, DALE R (DALE) [dworley@avaya.com]
Sent: Tuesday, April 06, 2010 11:01 PM
To: Salvatore Loreto; sipcore@ietf.org
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?

________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Salv=
atore Loreto [salvatore.loreto@ericsson.com]
Sent: Thursday, April 01, 2010 3:36 AM

A good mechanism to discovery UA capabilities is essential to provide meani=
ngful service:
So I am in favor of pursuing the work,
in particular the possibility to use a single OPTIONS request to retrieve t=
he capabilities from ALL UAS(s) behind a forking proxy.
_______________________________________

I think it is a misconception to use UA discovery to provide services.  The=
 difficulty is that the set of available UAs for a given destination URI ch=
anges unpredictably -- elements become inacessible (mobile phone out of ran=
ge!), call routing changes due to time of day, etc.  Instead, SIP provides =
a remarkably effective system for selecting the best target of a call *at t=
he moment the call is initiated* -- the caller declares his preferences, an=
d the various intermediate elements combine the caller's desires with the r=
outing information that they possess to forward the call closer to the pref=
erred destination.

As a developer, I've had to wrestle with this.  In order to provide clean "=
Is extension 1234 busy?" information, our "BLF agent" maintains subscriptio=
ns to the "reg" event package for the AOR "sip:1234@example.com".  For each=
 reported contact for the AOR, it maintains a dialog subscription to the co=
ntact.  Only by tracking the "reg" events can it keep track of the UAs avai=
lable for that AOR.  And even then, it is at the mercy of network failures.

Dale
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore=

From dworley@avaya.com  Tue Apr  6 13:10:26 2010
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From: "WORLEY, DALE R (DALE)" <dworley@avaya.com>
To: SIPCORE <sipcore@ietf.org>
Date: Tue, 6 Apr 2010 16:10:20 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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No matter what we do here, it is inevitable that the implementions of OPTIO=
NS in the network will be very uneven.  Uneven to the degree that it is unl=
ikely that any SIP element will be able to depend on another SIP element's =
OPTIONS response to automatically decide how to proceed.  (Possibly with th=
e exception of "If I send OPTIONS to an IP address and get a response, then=
 the SIP element at that IP address is functional.")  The sipXecs system th=
at I work on uses OPTIONS for liveness testing when we are manually monitor=
ing a system, but the failure responses are handed over to a human for comp=
rehension; forking et al. considers only responses to "real" requests.

Within that context, if we want to define robust ways to probe the SIP netw=
ork, we should define new protocol elements, preferably new methods, with c=
arefully defined behaviors.

On the other hand, I see benefit in defining not best practices (which we c=
onsider should exclude all others) but "good practices" (guides to implemen=
tors of OPTIONS) and especially "warnings to users" (caveats to users of OP=
TIONS that it may not function as they desire).

Dale

From dworley@avaya.com  Tue Apr  6 13:17:24 2010
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From: "WORLEY, DALE R (DALE)" <dworley@avaya.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 6 Apr 2010 16:12:05 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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> From: Christer Holmberg [christer.holmberg@ericsson.com]

> From my perspective, the main use-case is not to discover services,
> but to discover capabilities in order to determine what services can be a=
pplied.

That's very difficult, though, as you need to monitor when the contacts bec=
ome available
and unavailable.  It's more reliable to attempt the call with the service a=
ctive, and if it fails,
fall back to attempting the call without the service.

> The idea is also that the OPTIONS request would be sent prior to the INVI=
TE,
> to try to make sure that the returned contacts are actually available.

Why not just send the INVITE and see if it fails?  In the most common case =
(success), that
saves one round-trip.

Dale

From christer.holmberg@ericsson.com  Tue Apr  6 13:58:52 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 6 Apr 2010 22:55:04 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

>>From my perspective, the main use-case is not to discover services,
>>but to discover capabilities in order to determine what services can be a=
pplied.
>
>That's very difficult, though, as you need to monitor when the contacts be=
come available
>and unavailable.  It's more reliable to attempt the call with the service =
active, and if it fails,
>fall back to attempting the call without the service.
>
>>The idea is also that the OPTIONS request would be sent prior to the INVI=
TE,
>>to try to make sure that the returned contacts are actually available.
>
>Why not just send the INVITE and see if it fails?  In the most common case=
 (success), that saves one round-trip.

Because that is crappy protocol design. And, the INVITE might trigger diffe=
rent resources etc to be reserved, so you don't want to keep sending those.

Please not that functionwise I am not trying to "enhance" OPTIONS as such -=
 I am simply trying to solve the forking case.

Otherwise, if we think OPTIONS is useless, it should be removed from the pr=
otocol...

Regards,

Christer=

From christer.holmberg@ericsson.com  Tue Apr  6 14:05:46 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "WORLEY, DALE R (DALE)" <dworley@avaya.com>, SIPCORE <sipcore@ietf.org>
Date: Tue, 6 Apr 2010 23:05:40 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

>No matter what we do here, it is inevitable that the implementions of OPTI=
ONS in the network will be very uneven.  Uneven to the degree that it is un=
likely that any SIP element will be able to depend on another SIP element's=
 OPTIONS response to automatically decide=20
>how to proceed.  (Possibly with the exception of "If I send OPTIONS to an =
IP address and get a response, then the SIP element at that IP address is f=
unctional.")

I don't agree. There is much usable information you can receive in the OPTI=
ONS response.

>The sipXecs system that I work on uses OPTIONS for liveness testing when w=
e are manually monitoring a system, but the failure responses are handed ov=
er to a human for comprehension; forking et al. considers only responses to=
 "real" requests.

So, maybe you can now start using OPTIONS for what it was intented to be us=
ed for :)

>Within that context, if we want to define robust ways to probe the SIP net=
work, we should define new protocol elements, preferably new methods, with =
carefully defined behaviors.

Sure, if someone comes up with cases that OPTIONS can't be used for. At thi=
s point I think we are only speculating, especially since OPTIONS hasn't be=
en used very much for capability quering. And, the forking issue has at lea=
st been one reason for that.

>On the other hand, I see benefit in defining not best practices (which we =
consider should exclude all others) but "good practices" (guides to impleme=
ntors of OPTIONS) and especially "warnings to users" (caveats to users of O=
PTIONS that it may not function as they=20
>desire).

Whatever you wanna call it :)

Regards,

Christer=

From pkyzivat@cisco.com  Tue Apr  6 14:49:37 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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[as individual]

I'm having trouble understanding the issue here.
I presume we are talking about *initial* subscribe here, not resubscribe.

IIUC the case under discussion is:

- Bob sends Alice some message
- Alice responds with 503, retry-after
   because she is overloaded
- Alice then sends Bob a SUBSCRIBE
- Bob replies 200 to that
- Bob delays the NOTIFY for the subscribe due to
   the retry-after timer
- Timer-N expires so Alice considers the subscribe a failure.
- Eventually Bob sends the NOTIFY, but Alice rejects it
   with 481
   OR
   Bob realizes that Timer-N has expired and so does not
   send the NOTIFY.

Is there *any* problem here? This seems to be working about as well as 
anyone might hope. What would you expect to happen?

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi,
> 
> Overload, that would "delay" the NOTIFY, is in fact one of the scenarios I also had in mind, and got me thinking whether it would not simply be better to maintain the pre-SUB dialog state, instead of terminating the dialog.
> 
> Regards,
> 
> Christer
> 
> ________________________________________
> From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Brett Tate [brett@broadsoft.com]
> Sent: Tuesday, April 06, 2010 8:55 PM
> To: Adam Roach; SIPCORE
> Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
> 
>> If someone wants to raise an objection to the decision to
>> treat Timer N expiration the same way we treat SUBSCRIBE
>> transaction timeouts, I'm happy to engage in that discussion.
> 
> The decision has an interesting interaction concerning overload.  More specifically if SUBSCRIBE received from device that the notifier still thinks is in overload, notifier will send 200 response but not the NOTIFY until later.  Or an overloaded middle box throttling some traffic might reject the NOTIFY with 500/503 Retry-After.
> 
> Concerning the overload interactions, is the decision concerning Timer N expiration acceptable since not allowing SUBSCRIBE 200 response to confirm the subscription creation?
> 
> _______________________________________________
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> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
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> 

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On 4/6/10 16:49, Apr 6, Paul Kyzivat wrote:
> [as individual]
>
> I'm having trouble understanding the issue here.
> I presume we are talking about *initial* subscribe here, not resubscribe.

[also as individual]

No, we know how initial subscribe behaves. The issue under discussion 
is: what happens if Timer N expires during a resubscription attempt?

/a

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:) My understanding is that both are being discussed since relevant to both=
 situations.

> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Tuesday, April 06, 2010 6:00 PM
> To: Paul Kyzivat
> Cc: Christer Holmberg; Brett Tate; SIPCORE
> Subject: Re: [sipcore] rfc3265bis: SIP events redux [was Minutes
> Posted]
>=20
> On 4/6/10 16:49, Apr 6, Paul Kyzivat wrote:
> > [as individual]
> >
> > I'm having trouble understanding the issue here.
> > I presume we are talking about *initial* subscribe here, not
> resubscribe.
>=20
> [also as individual]
>=20
> No, we know how initial subscribe behaves. The issue under discussion
> is: what happens if Timer N expires during a resubscription attempt?
>=20
> /a

From adam@nostrum.com  Tue Apr  6 15:44:59 2010
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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[as individual]

Ah, okay.

For initial subscriptions: the problem that Timer N solves is a very 
real one. I think it is reasonable to trade-off the ambiguity around 
subscription establishment for the extremely rare failure in overload 
conditions. IMHO, the failure to timer supervise the period between an 
initial SUBSCRIBE and an initial NOTIFY was the largest flaw in RFC 3265 
when we published it.

It's worth mentioning in the bis draft that there can be circumstances 
-- notably, related to congestion conditions -- that might lead to these 
kinds of failures. I'll add something to this effect in the next draft. 
But I think removing Timer N would be a huge step backwards.

Regarding the behavior of Timer N on a *re*subscribe: I don't strongly 
care one way or another. But it needs to be unambiguous and documented. 
I'm happy to say "Timer N expiration on a resubscribe destroys the 
subscription." I'm also happy to say "Timer N isn't even run for 
resubscriptions." I'm slightly biased towards the first case, since the 
situation clearly means something has gone wrong in the network.

Your overload scenario is an interesting one, and we should certainly 
consider how the system should behave in this case. On first blush, it 
seems to me that if the network is so congested that it can't get the 
NOTIFY through within a 32-second window, everyone might benefit from 
the subscription going away anyway.

/a



On 4/6/10 17:03, Apr 6, Brett Tate wrote:
> :) My understanding is that both are being discussed since relevant to both situations.
>
>    
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: Tuesday, April 06, 2010 6:00 PM
>> To: Paul Kyzivat
>> Cc: Christer Holmberg; Brett Tate; SIPCORE
>> Subject: Re: [sipcore] rfc3265bis: SIP events redux [was Minutes
>> Posted]
>>
>> On 4/6/10 16:49, Apr 6, Paul Kyzivat wrote:
>>      
>>> [as individual]
>>>
>>> I'm having trouble understanding the issue here.
>>> I presume we are talking about *initial* subscribe here, not
>>>        
>> resubscribe.
>>
>> [also as individual]
>>
>> No, we know how initial subscribe behaves. The issue under discussion
>> is: what happens if Timer N expires during a resubscription attempt?
>>
>> /a
>>      


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> Is there *any* problem here? This seems to be working about=20
> as well as anyone might hope.=20

It works okay excluding the non backward compatibility with RFC 3265 (conce=
rning SUBSCRIBE's 2xx response impacts) and continuing to uselessly repeat =
the Timer N expiration's subscription teardown process until related overlo=
ad issue resolved.


> What would you expect to happen?

Potentially allow the SUBSCRIBE's 200 response to create the subscription/d=
ialog similar to RFC 3265. =20

The draft's Timer N (and lack of related NOTIFY) text is potentially worded=
 in a way which does not require the tear down (and non creation) of subscr=
iption.


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Brett Tate wrote:
>> Is there *any* problem here? This seems to be working about 
>> as well as anyone might hope. 
> 
> It works okay excluding the non backward compatibility with RFC 3265 (concerning SUBSCRIBE's 2xx response impacts) and continuing to uselessly repeat the Timer N expiration's subscription teardown process until related overload issue resolved.
> 
> 
>> What would you expect to happen?
> 
> Potentially allow the SUBSCRIBE's 200 response to create the subscription/dialog similar to RFC 3265.  

Then what would happen in a polling subscribe (Expires:0) if the 200 is 
received and the NOTIFY is not received? How long should you wait before 
giving up? ISTM you still need timer-N to tell you it failed, so you can 
give up waiting.

	Thanks,
	Paul

From adam@nostrum.com  Tue Apr  6 16:31:32 2010
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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On 4/6/10 15:02, Apr 6, Christer Holmberg wrote:
> Hi,
>
>    
>>> In any case, proxies don't inherently care about subscription state. If
>>> you're doing something in your proxy that is attempting to figure out
>>> the state of a subscription, you need to take care to figure out what's
>>> going on. In that case, yeah, Timer N is going to be of interest to you.
>>> But that's not something we need to specify -- it's actually very
>>> application-specific.
>>>        
>> The draft says that proxies no not need to support anything in addition to 3261, so from that point of view you are correct.
>>
>> However, at the same time it talks about proxies adding Record-Route if they are interested in the SUBs and NOTs, so I assumed that means that they are "allowed" to actually maintain subscription dialog state - in the same way as they might maintain invite dialog state.
>> _______________________________________________
>>
>> Even if a proxy is keeping subscription-state information, I don't see Timer N introducing a problem.  The proxy sees all the SUBSCRIBEs and NOTIFYs and their responses.  Under 3265bis, the NOTIFYs, actually, the the *responses* to the NOTIFYs show all of the
>> subscription state.  In particular, in seciton 4.1.2.4, I see:
>>
>>    Until Timer N expires, several NOTIFY messages may arrive from
>>    different destinations (see Section 4.4.1).  Each of these messages
>>    establish a new dialog and a new subscription.  After the expiration
>>    of Timer N, the subscriber SHOULD reject any such NOTIFY messages
>>    that would otherwise establish a new dialog with a "481" response
>>    code.
>>      
> Exactly, and that's why I think draft-ietf-sipcore-subnot-etags could cause issues, because it allows the NOTIFY to be suppressed.

You're conflating *initial* NOTIFYs with *refresh* NOTIFYs.

The subnot-etags document explicitly does not allow suppression of 
initial NOTIFY messages. They are critical to establishing a dialog.

The text Dale cites relates exclusively to handling of *initial* 
NOTIFYs. It is unrelated to the kinds of NOTIFY messages that 
subnot-etags is allows to suppress.

/a

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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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On 4/6/10 17:46, Apr 6, Brett Tate wrote:
>> Is there *any* problem here? This seems to be working about
>> as well as anyone might hope.
>>      
> It works okay excluding the non backward compatibility with RFC 3265 (concerning SUBSCRIBE's 2xx response impacts)...

I don't think it's accurate to characterize the change as 
non-backwards-compatible. The only wire-visible behavior that one could 
conceivable observe is that a subscriber might make a different decision 
about which subscription to accept if the SUBSCRIBE forks and the 
subscriber wants only one subscription.

What we *do* gain is a simplification for implementors. Because the 
behavior of establishing the dialog when a NOTIFY shows up (instead of 
"the first of a 2xx or a NOTIFY) is much easier to code, people will 
make fewer mistakes, and have fewer bugs. But you'd be hard pressed to 
see a difference on the wire.

> The draft's Timer N (and lack of related NOTIFY) text is potentially 
> worded in a way which does not require the tear down (and non 
> creation) of subscription.

Tear-down of existing dialogs is still an open issue, sure. I don't get 
where you see any wiggle room in what happens if Timer N fires for the 
first SUBSCRIBE/NOTIFY exchange.

/a

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Hi,

While considering one problem in our productions' interoperability 
testing, I re-read some parts of offeranswer draft and correlative part of 
RFC3261. And I find a potential description problem in RFC3261 
incidentally.

//correlative part of RFC3261
o  If the initial offer is in an INVITE, the answer MUST be in a
         reliable non-failure message from UAS back to UAC which is
         correlated to that INVITE.  For this specification, that is
         only the final 2xx response to that INVITE.  That same exact
         answer MAY also be placed in any provisional responses sent
         prior to the answer.  The UAC MUST treat the first session
         description it receives as the answer, and MUST ignore any
         session descriptions in subsequent responses to the initial
         INVITE.

As the text allow UAS send SDP before the final answer, the first session 
description UAC receives can not be treated as answer in such cases. And 
the current text defines that UAC MUST ignore any SDP in subsequent 
responses to the initial INVITE. whick may make the UAC ignore the real 
answer in some case.

I think the description should be: 
The UAC MUST treat the first session description which is in a reliable 
non-failure message it receives as the answer, and MUST ignore any session 
descriptions in subsequent responses to the initial INVITE.

And if the problem is exist, where should be the correction's location?
I think draft-ietf-sipcore-reinvite-03 is not proper, as it aims for 
Re-INVITE. And a new draft for such tiny *potential* problem is also not 
proper.

Thanks,

Gao

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<br><font size=2 face="sans-serif">Hi,</font>
<br>
<br><font size=2 face="sans-serif">While considering one problem in our
productions' interoperability testing, I re-read some parts of offeranswer
draft and correlative part of RFC3261. And I find a potential description
problem in RFC3261 incidentally.</font>
<br>
<br><font size=2 face="sans-serif">//correlative part of RFC3261</font>
<br><font size=2 face="Courier New">o &nbsp;If the initial offer is in
an INVITE, the answer MUST be in a</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;reliable
non-failure message from UAS back to UAC which is</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;correlated
to that INVITE. &nbsp;For this specification, that is</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;only
the final 2xx response to that INVITE. &nbsp;That same exact</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;answer
MAY also be placed in any provisional responses sent</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;prior
to the answer. &nbsp;The UAC MUST treat the <b>first session</b></font>
<br><font size=2 face="Courier New"><b>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;description</b>
it receives as the answer, and MUST ignore any</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;session
descriptions in subsequent responses to the initial</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;INVITE.</font>
<br>
<br><font size=2 face="sans-serif">As the text allow UAS send SDP before
the final answer, the first session description UAC receives can not be
treated as answer in such cases. And the current text defines that UAC
MUST ignore any SDP in subsequent responses to the initial INVITE. whick
may make the UAC ignore the real answer in some case.</font>
<br>
<br><font size=2 face="sans-serif">I think the description should be: </font>
<br><font size=2 face="sans-serif">The UAC MUST treat the first session
description <b>which is in a reliable non-failure message</b> it receives
as the answer, and MUST ignore any session descriptions in subsequent responses
to the initial INVITE.</font>
<br>
<br><font size=2 face="sans-serif">And if the problem is exist, where should
be the correction's location?</font>
<br><font size=2 face="sans-serif">I think draft-ietf-sipcore-reinvite-03
is not proper, as it aims for Re-INVITE. And a new draft for such tiny
*potential* problem is also not proper.</font>
<br>
<br><font size=2 face="sans-serif">Thanks,</font>
<br>
<br><font size=2 face="sans-serif">Gao</font>
<br>
<br><font size=2 face="sans-serif">===================================<br>
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 Tel &nbsp; &nbsp;: 87211<br>
 Tel2 &nbsp; :(+86)-025-52877211<br>
 e_mail : gao.yang2@zte.com.cn<br>
===================================</font><br><pre>
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From christer.holmberg@ericsson.com  Tue Apr  6 22:33:54 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>
Date: Wed, 7 Apr 2010 07:33:35 +0200
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Hi,=20

>>>>In any case, proxies don't inherently care about subscription state.=20
>>>>If you're doing something in your proxy that is attempting to figure=20
>>>>out the state of a subscription, you need to take care to=20
>>>>figure out what's going on. In that case, yeah, Timer N is going to=20
>>>>be of interest to you.
>>>>But that's not something we need to specify -- it's actually very=20
>>>>application-specific.
>>>>       =20
>>>The draft says that proxies no not need to support=20
>>>anything in addition to 3261, so from that point of view you=20
>>>are correct.
>>>
>>>However, at the same time it talks about proxies adding=20
>>>Record-Route if they are interested in the SUBs and NOTs, so=20
>>>I assumed that means that they are "allowed" to actually=20
>>>maintain subscription dialog state - in the same way as they=20
>>>might maintain invite dialog state.
>>> _______________________________________________
>>>
>>> Even if a proxy is keeping subscription-state information, I don't=20
>>> see Timer N introducing a problem.  The proxy sees all the=20
>>> SUBSCRIBEs and NOTIFYs and their responses.  Under 3265bis,=20
>>> the NOTIFYs, actually, the the *responses* to the NOTIFYs=20
>>> show all of the subscription state.  In particular, in=20
>>> seciton 4.1.2.4, I see:
>>>
>>>    Until Timer N expires, several NOTIFY messages may arrive from
>>>    different destinations (see Section 4.4.1).  Each of these messages
>>>    establish a new dialog and a new subscription.  After the expiration
>>>    of Timer N, the subscriber SHOULD reject any such NOTIFY messages
>>>    that would otherwise establish a new dialog with a "481" response
>>>    code.
>>>     =20
>>Exactly, and that's why I think draft-ietf-sipcore-subnot-etags could cau=
se issues, because=20
>>it allows the NOTIFY to be suppressed.
>=20
>You're conflating *initial* NOTIFYs with *refresh* NOTIFYs.
>=20
>The subnot-etags document explicitly does not allow=20
>suppression of initial NOTIFY messages. They are critical to=20
>establishing a dialog.
>
>The text Dale cites relates exclusively to handling of=20
>*initial* NOTIFYs. It is unrelated to the kinds of NOTIFY=20
>messages that subnot-etags is allows to suppress.

Correct. I've been trying to say that, but I appologise if it didn't apply =
to Dale's case.

Regards,

Christer


From christer.holmberg@ericsson.com  Tue Apr  6 23:20:48 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, Brett Tate <brett@broadsoft.com>
Date: Wed, 7 Apr 2010 08:20:38 +0200
Thread-Topic: Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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Cc: SIPCORE <sipcore@ietf.org>
Subject: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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Hi,

I haven't double checked whether it is said in 3265bis, but we also need to=
 make clear whether it is the NOTIFY request, or the NOT 200 response, that=
 "acknowledges" the (re-)subscription.

Because, a NOTIFY request can also be rejected (e.g. due to overload), so i=
n order to avoid sub state unsynch every entity in the path needs to know w=
hat a NOT non-200 response means.

Regards,

Christer


=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> Sent: 7. huhtikuuta 2010 2:37
> To: Brett Tate
> Cc: SIPCORE
> Subject: Re: [sipcore] rfc3265bis: SIP events redux [was=20
> Minutes Posted]
>=20
> On 4/6/10 17:46, Apr 6, Brett Tate wrote:
> >> Is there *any* problem here? This seems to be working=20
> about as well=20
> >> as anyone might hope.
> >>     =20
> > It works okay excluding the non backward compatibility with=20
> RFC 3265 (concerning SUBSCRIBE's 2xx response impacts)...
>=20
> I don't think it's accurate to characterize the change as=20
> non-backwards-compatible. The only wire-visible behavior that=20
> one could conceivable observe is that a subscriber might make=20
> a different decision about which subscription to accept if=20
> the SUBSCRIBE forks and the subscriber wants only one subscription.
>=20
> What we *do* gain is a simplification for implementors.=20
> Because the behavior of establishing the dialog when a NOTIFY=20
> shows up (instead of "the first of a 2xx or a NOTIFY) is much=20
> easier to code, people will make fewer mistakes, and have=20
> fewer bugs. But you'd be hard pressed to see a difference on the wire.
>=20
> > The draft's Timer N (and lack of related NOTIFY) text is=20
> potentially=20
> > worded in a way which does not require the tear down (and non
> > creation) of subscription.
>=20
> Tear-down of existing dialogs is still an open issue, sure. I=20
> don't get where you see any wiggle room in what happens if=20
> Timer N fires for the first SUBSCRIBE/NOTIFY exchange.
>=20
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From christer.holmberg@ericsson.com  Tue Apr  6 23:27:29 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, Brett Tate <brett@broadsoft.com>
Date: Wed, 7 Apr 2010 08:27:22 +0200
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Hi,=20

>>>Is there *any* problem here? This seems to be working about as well=20
>>>as anyone might hope.
>>>     =20
>>It works okay excluding the non backward compatibility with RFC 3265 (con=
cerning SUBSCRIBE's 2xx response impacts)...
>=20
>I don't think it's accurate to characterize the change as=20
>non-backwards-compatible. The only wire-visible behavior that=20
>one could conceivable observe is that a subscriber might make=20
>a different decision about which subscription to accept if=20
>the SUBSCRIBE forks and the subscriber wants only one subscription.
>=20
>What we *do* gain is a simplification for implementors.=20
>Because the behavior of establishing the dialog when a NOTIFY=20
>shows up (instead of "the first of a 2xx or a NOTIFY) is much=20
>easier to code, people will make fewer mistakes, and have=20
>fewer bugs. But you'd be hard pressed to see a difference on the wire.

I agree with Adam.

Of course, for resubscriptions we don't have the forking issue, but I still=
 think we should have the same behavior, ie the NOTIFY acknowledges the res=
ub.

...and, again, that is why I have an issue with allowing the suppression of=
 NOTIFYs. It's not a problem for endpoints, but for stateful intermediaries=
 (proxies ARE allowed to Record-Route SUB requests, so)


>>The draft's Timer N (and lack of related NOTIFY) text is potentially=20
>>worded in a way which does not require the tear down (and non
>>creation) of subscription.
>=20
>Tear-down of existing dialogs is still an open issue, sure. I=20
>don't get where you see any wiggle room in what happens if=20
>Timer N fires for the first SUBSCRIBE/NOTIFY exchange.

For the initial SUB/NOT, I think that tearing down the dialog is the only t=
hing you can do.

Regards,

Christer


From john.elwell@siemens-enterprise.com  Tue Apr  6 23:32:05 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 7 Apr 2010 08:31:57 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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I agree with Dale. What happens to an OPTIONS request will not necessarily =
be reflected when an INVITE request is sent shortly afterwards. As a furthe=
r examples, in contact centres, it is usual to distribute calls cyclically =
or randomly between qualified and available agents. Sending an INVITE reque=
st to a UA determined by responses to an OPTIONS request would break that m=
odel - the issuer of the OPTIONS and INVITE requests cannot be expected to =
obey the same rotation rules as the domain proxy that routes INVITE request=
s. Issuing an INVITE request to the AOR is unlikely to result in it being d=
elivered to the UA that might be expected from analysis of responses to an =
OPTIONS request.

I don't think we should work on trying to make OPTIONS work for this purpos=
e. If we really need to do anything with OPTIONS, we should just issue warn=
ings about its unsuitability for purposes other than a simple ping.

John


> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of WORLEY, DALE R (DALE)
> Sent: 06 April 2010 21:01
> To: Salvatore Loreto; sipcore@ietf.org
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
> ________________________________________
> From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On=20
> Behalf Of Salvatore Loreto [salvatore.loreto@ericsson.com]
> Sent: Thursday, April 01, 2010 3:36 AM
>=20
> A good mechanism to discovery UA capabilities is essential to=20
> provide meaningful service:
> So I am in favor of pursuing the work,
> in particular the possibility to use a single OPTIONS request=20
> to retrieve the capabilities from ALL UAS(s) behind a forking proxy.
> _______________________________________
>=20
> I think it is a misconception to use UA discovery to provide=20
> services.  The difficulty is that the set of available UAs=20
> for a given destination URI changes unpredictably -- elements=20
> become inacessible (mobile phone out of range!), call routing=20
> changes due to time of day, etc.  Instead, SIP provides a=20
> remarkably effective system for selecting the best target of=20
> a call *at the moment the call is initiated* -- the caller=20
> declares his preferences, and the various intermediate=20
> elements combine the caller's desires with the routing=20
> information that they possess to forward the call closer to=20
> the preferred destination.
>=20
> As a developer, I've had to wrestle with this.  In order to=20
> provide clean "Is extension 1234 busy?" information, our "BLF=20
> agent" maintains subscriptions to the "reg" event package for=20
> the AOR "sip:1234@example.com".  For each reported contact=20
> for the AOR, it maintains a dialog subscription to the=20
> contact.  Only by tracking the "reg" events can it keep track=20
> of the UAs available for that AOR.  And even then, it is at=20
> the mercy of network failures.
>=20
> Dale
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From christer.holmberg@ericsson.com  Tue Apr  6 23:37:37 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 7 Apr 2010 08:37:29 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>I agree with Dale. What happens to an OPTIONS request will=20
>not necessarily be reflected when an INVITE request is sent=20
>shortly afterwards. As a further examples, in contact=20
>centres, it is usual to distribute calls cyclically or=20
>randomly between qualified and available agents.

Sure, there are cases when it may not be feasible.

>Sending an INVITE request to a UA determined by responses to an OPTIONS=20
>request would break that model - the issuer of the OPTIONS=20
>and INVITE requests cannot be expected to obey the same=20
>rotation rules as the domain proxy that routes INVITE=20
>requests. Issuing an INVITE request to the AOR is unlikely to=20
>result in it being delivered to the UA that might be expected=20
>from analysis of responses to an OPTIONS request.

I think the main use-case is when you want to contact end users.=20

>I don't think we should work on trying to make OPTIONS work=20
>for this purpose. If we really need to do anything with=20
>OPTIONS, we should just issue warnings about its=20
>unsuitability for purposes other than a simple ping.

Again, protocolwise we don't need to do anything for OPTIONS 3xx, because w=
e already have the tools to do it :)

The question is whether it would be good to document it.

Regards,

Christer

> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of WORLEY, DALE R (DALE)
> > Sent: 06 April 2010 21:01
> > To: Salvatore Loreto; sipcore@ietf.org
> > Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> >=20
> > ________________________________________
> > From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org]=20
> On Behalf Of=20
> > Salvatore Loreto [salvatore.loreto@ericsson.com]
> > Sent: Thursday, April 01, 2010 3:36 AM
> >=20
> > A good mechanism to discovery UA capabilities is essential=20
> to provide=20
> > meaningful service:
> > So I am in favor of pursuing the work, in particular the=20
> possibility=20
> > to use a single OPTIONS request to retrieve the=20
> capabilities from ALL=20
> > UAS(s) behind a forking proxy.
> > _______________________________________
> >=20
> > I think it is a misconception to use UA discovery to=20
> provide services. =20
> > The difficulty is that the set of available UAs for a given=20
> > destination URI changes unpredictably -- elements become=20
> inacessible=20
> > (mobile phone out of range!), call routing changes due to=20
> time of day,=20
> > etc.  Instead, SIP provides a remarkably effective system for=20
> > selecting the best target of a call *at the moment the call is=20
> > initiated* -- the caller declares his preferences, and the various=20
> > intermediate elements combine the caller's desires with the routing=20
> > information that they possess to forward the call closer to the=20
> > preferred destination.
> >=20
> > As a developer, I've had to wrestle with this.  In order to provide=20
> > clean "Is extension 1234 busy?" information, our "BLF=20
> agent" maintains=20
> > subscriptions to the "reg" event package for the AOR=20
> > "sip:1234@example.com".  For each reported contact for the AOR, it=20
> > maintains a dialog subscription to the contact.  Only by=20
> tracking the=20
> > "reg" events can it keep track of the UAs available for=20
> that AOR.  And=20
> > even then, it is at the mercy of network failures.
> >=20
> > Dale
> > _______________________________________________
> > 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
> =

From john.elwell@siemens-enterprise.com  Tue Apr  6 23:56:04 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 7 Apr 2010 08:55:57 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 07 April 2010 07:37
> To: Elwell, John; WORLEY, DALE R (DALE); Salvatore Loreto;=20
> sipcore@ietf.org
> Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
>=20
> Hi,=20
>=20
> >I agree with Dale. What happens to an OPTIONS request will=20
> >not necessarily be reflected when an INVITE request is sent=20
> >shortly afterwards. As a further examples, in contact=20
> >centres, it is usual to distribute calls cyclically or=20
> >randomly between qualified and available agents.
>=20
> Sure, there are cases when it may not be feasible.
>=20
> >Sending an INVITE request to a UA determined by responses to=20
> an OPTIONS=20
> >request would break that model - the issuer of the OPTIONS=20
> >and INVITE requests cannot be expected to obey the same=20
> >rotation rules as the domain proxy that routes INVITE=20
> >requests. Issuing an INVITE request to the AOR is unlikely to=20
> >result in it being delivered to the UA that might be expected=20
> >from analysis of responses to an OPTIONS request.
>=20
> I think the main use-case is when you want to contact end users.=20
[JRE] A contact centre agent is indeed an end user.

>=20
> >I don't think we should work on trying to make OPTIONS work=20
> >for this purpose. If we really need to do anything with=20
> >OPTIONS, we should just issue warnings about its=20
> >unsuitability for purposes other than a simple ping.
>=20
> Again, protocolwise we don't need to do anything for OPTIONS=20
> 3xx, because we already have the tools to do it :)
[JRE] Yes, but a forking proxy at present probably doesn't issue 3xx respon=
ses, so for the mechanism to be usable we would need to mandate this behavi=
our. Then we would still be left with legacy forking proxies that don't sup=
port the newly mandated behaviour, so do we have to specify an option tag a=
nd Proxy-Require? I am not convinced we already have the tools.

>=20
> The question is whether it would be good to document it.
[JRE] As somebody else pointed out, if you want to obtain OPTIONS responses=
 from individual UAs registered against an AOR, you can use the registratio=
n event package to identify them and send OPTIONS requests to them individu=
ally. If we are going to document a practice (which I do not necessarily ag=
ree with), we should at least consider alternatives.

John


>=20
> Regards,
>=20
> Christer
>=20
> > > -----Original Message-----
> > > From: sipcore-bounces@ietf.org
> > > [mailto:sipcore-bounces@ietf.org] On Behalf Of WORLEY,=20
> DALE R (DALE)
> > > Sent: 06 April 2010 21:01
> > > To: Salvatore Loreto; sipcore@ietf.org
> > > Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> > >=20
> > > ________________________________________
> > > From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org]=20
> > On Behalf Of=20
> > > Salvatore Loreto [salvatore.loreto@ericsson.com]
> > > Sent: Thursday, April 01, 2010 3:36 AM
> > >=20
> > > A good mechanism to discovery UA capabilities is essential=20
> > to provide=20
> > > meaningful service:
> > > So I am in favor of pursuing the work, in particular the=20
> > possibility=20
> > > to use a single OPTIONS request to retrieve the=20
> > capabilities from ALL=20
> > > UAS(s) behind a forking proxy.
> > > _______________________________________
> > >=20
> > > I think it is a misconception to use UA discovery to=20
> > provide services. =20
> > > The difficulty is that the set of available UAs for a given=20
> > > destination URI changes unpredictably -- elements become=20
> > inacessible=20
> > > (mobile phone out of range!), call routing changes due to=20
> > time of day,=20
> > > etc.  Instead, SIP provides a remarkably effective system for=20
> > > selecting the best target of a call *at the moment the call is=20
> > > initiated* -- the caller declares his preferences, and=20
> the various=20
> > > intermediate elements combine the caller's desires with=20
> the routing=20
> > > information that they possess to forward the call closer to the=20
> > > preferred destination.
> > >=20
> > > As a developer, I've had to wrestle with this.  In order=20
> to provide=20
> > > clean "Is extension 1234 busy?" information, our "BLF=20
> > agent" maintains=20
> > > subscriptions to the "reg" event package for the AOR=20
> > > "sip:1234@example.com".  For each reported contact for=20
> the AOR, it=20
> > > maintains a dialog subscription to the contact.  Only by=20
> > tracking the=20
> > > "reg" events can it keep track of the UAs available for=20
> > that AOR.  And=20
> > > even then, it is at the mercy of network failures.
> > >=20
> > > Dale
> > > _______________________________________________
> > > 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
> > =

From christer.holmberg@ericsson.com  Wed Apr  7 00:01:55 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 7 Apr 2010 09:01:48 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>>I think the main use-case is when you want to contact end users.=20
>[JRE] A contact centre agent is indeed an end user.

Yes, but not the same kind of end user :)

>>Again, protocolwise we don't need to do anything for OPTIONS 3xx,=20
>>because we already have the tools to do it :)
>[JRE] Yes, but a forking proxy at present probably doesn't=20
>issue 3xx responses, so for the mechanism to be usable we=20
>would need to mandate this behaviour. Then we would still be=20
>left with legacy forking proxies that don't support the newly=20
>mandated behaviour, so do we have to specify an option tag=20
>and Proxy-Require? I am not convinced we already have the tools.

We have the Request-Disposition "redirect" feature tag.

No, I know that not all proxies support it, but the tool exists.

There is NO mechanism which will ensure that it always works. If the remote=
 end doesn't support it - it won't work.

>>The question is whether it would be good to document it.
>[JRE] As somebody else pointed out, if you want to obtain=20
>OPTIONS responses from individual UAs registered against an=20
>AOR, you can use the registration event package to identify=20
>them and send OPTIONS requests to them individually. If we=20
>are going to document a practice (which I do not necessarily=20
>agree with), we should at least consider alternatives.

I have no problem to also look at other solutions, and we can even allow di=
fferent alternatives. It would be good to document the pros and cons of eac=
h alternatives, and people can then choose what to use based on their use-c=
ases and requirements etc.

And, keep in mind that the event package solution requires support of the e=
vent package, so it won't always work either :)

Regards,

Christer



> > Regards,
> >=20
> > Christer
> >=20
> > > > -----Original Message-----
> > > > From: sipcore-bounces@ietf.org
> > > > [mailto:sipcore-bounces@ietf.org] On Behalf Of WORLEY,
> > DALE R (DALE)
> > > > Sent: 06 April 2010 21:01
> > > > To: Salvatore Loreto; sipcore@ietf.org
> > > > Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> > > >=20
> > > > ________________________________________
> > > > From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org]
> > > On Behalf Of
> > > > Salvatore Loreto [salvatore.loreto@ericsson.com]
> > > > Sent: Thursday, April 01, 2010 3:36 AM
> > > >=20
> > > > A good mechanism to discovery UA capabilities is essential
> > > to provide
> > > > meaningful service:
> > > > So I am in favor of pursuing the work, in particular the
> > > possibility
> > > > to use a single OPTIONS request to retrieve the
> > > capabilities from ALL
> > > > UAS(s) behind a forking proxy.
> > > > _______________________________________
> > > >=20
> > > > I think it is a misconception to use UA discovery to
> > > provide services. =20
> > > > The difficulty is that the set of available UAs for a given=20
> > > > destination URI changes unpredictably -- elements become
> > > inacessible
> > > > (mobile phone out of range!), call routing changes due to
> > > time of day,
> > > > etc.  Instead, SIP provides a remarkably effective system for=20
> > > > selecting the best target of a call *at the moment the call is
> > > > initiated* -- the caller declares his preferences, and
> > the various
> > > > intermediate elements combine the caller's desires with
> > the routing
> > > > information that they possess to forward the call closer to the=20
> > > > preferred destination.
> > > >=20
> > > > As a developer, I've had to wrestle with this.  In order
> > to provide
> > > > clean "Is extension 1234 busy?" information, our "BLF
> > > agent" maintains
> > > > subscriptions to the "reg" event package for the AOR=20
> > > > "sip:1234@example.com".  For each reported contact for
> > the AOR, it
> > > > maintains a dialog subscription to the contact.  Only by
> > > tracking the
> > > > "reg" events can it keep track of the UAs available for
> > > that AOR.  And
> > > > even then, it is at the mercy of network failures.
> > > >=20
> > > > Dale
> > > > _______________________________________________
> > > > 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
> > > =

From john.elwell@siemens-enterprise.com  Wed Apr  7 00:09:53 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 7 Apr 2010 09:09:44 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 07 April 2010 08:02
> To: Elwell, John; WORLEY, DALE R (DALE); Salvatore Loreto;=20
> sipcore@ietf.org
> Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
>=20
> Hi,=20
>=20
> >>I think the main use-case is when you want to contact end users.=20
> >[JRE] A contact centre agent is indeed an end user.
>=20
> Yes, but not the same kind of end user :)
[JRE] So what kind of end user is within scope of this work? How does the i=
ssuer of OPTIONS/INVITE requests know whether he is targeting the right sor=
t of end user for the mechanism to work? I think it is dangerous to base me=
chanisms on assumptions about particular types of end user.

John

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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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--0014852f5770bbb1f30483a07162
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This is a special case. And the application you are talking about is a
B2BUA, right?

In this case the AOR addressed is an application address without registered
contacts. One can argue that the OPTIONS request should return the general
capabilities of the application. No OPTIONS forking is happening in this
case, no registered contacts. No issue?

In reality what the B2BUA will do with the OPTION request (forward or
answer) depends on the fantasy of the B2BUA developer.

I dont think this scenario is actually usefull in discussing whether or not
to write a BCP on OPTIONS handling. Or maybe it is as implication of B2BUA
behaviours can be taken into account.

/Hans Erik van Elburg


On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John <
john.elwell@siemens-enterprise.com> wrote:

> I agree with Dale. What happens to an OPTIONS request will not necessarily
> be reflected when an INVITE request is sent shortly afterwards. As a further
> examples, in contact centres, it is usual to distribute calls cyclically or
> randomly between qualified and available agents. Sending an INVITE request
> to a UA determined by responses to an OPTIONS request would break that model
> - the issuer of the OPTIONS and INVITE requests cannot be expected to obey
> the same rotation rules as the domain proxy that routes INVITE requests.
> Issuing an INVITE request to the AOR is unlikely to result in it being
> delivered to the UA that might be expected from analysis of responses to an
> OPTIONS request.
>
> I don't think we should work on trying to make OPTIONS work for this
> purpose. If we really need to do anything with OPTIONS, we should just issue
> warnings about its unsuitability for purposes other than a simple ping.
>
> John
>
>
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of WORLEY, DALE R (DALE)
> > Sent: 06 April 2010 21:01
> > To: Salvatore Loreto; sipcore@ietf.org
> > Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> >
> > ________________________________________
> > From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On
> > Behalf Of Salvatore Loreto [salvatore.loreto@ericsson.com]
> > Sent: Thursday, April 01, 2010 3:36 AM
> >
> > A good mechanism to discovery UA capabilities is essential to
> > provide meaningful service:
> > So I am in favor of pursuing the work,
> > in particular the possibility to use a single OPTIONS request
> > to retrieve the capabilities from ALL UAS(s) behind a forking proxy.
> > _______________________________________
> >
> > I think it is a misconception to use UA discovery to provide
> > services.  The difficulty is that the set of available UAs
> > for a given destination URI changes unpredictably -- elements
> > become inacessible (mobile phone out of range!), call routing
> > changes due to time of day, etc.  Instead, SIP provides a
> > remarkably effective system for selecting the best target of
> > a call *at the moment the call is initiated* -- the caller
> > declares his preferences, and the various intermediate
> > elements combine the caller's desires with the routing
> > information that they possess to forward the call closer to
> > the preferred destination.
> >
> > As a developer, I've had to wrestle with this.  In order to
> > provide clean "Is extension 1234 busy?" information, our "BLF
> > agent" maintains subscriptions to the "reg" event package for
> > the AOR "sip:1234@example.com <sip%3A1234@example.com>".  For each
> reported contact
> > for the AOR, it maintains a dialog subscription to the
> > contact.  Only by tracking the "reg" events can it keep track
> > of the UAs available for that AOR.  And even then, it is at
> > the mercy of network failures.
> >
> > Dale
> > _______________________________________________
> > 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
>

--0014852f5770bbb1f30483a07162
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

This is a special case. And the application you are talking about is a B2BU=
A, right?<br><br>In this case the AOR addressed is an application address w=
ithout registered contacts. One can argue that the OPTIONS request should r=
eturn the general capabilities of the application. No OPTIONS forking is ha=
ppening in this case, no registered contacts. No issue?<br>
<br>In reality what the B2BUA will do with the OPTION request (forward or a=
nswer) depends on the fantasy of the B2BUA developer.<br><br>I dont think t=
his scenario is actually usefull in discussing whether or not to write a BC=
P on OPTIONS handling. Or maybe it is as implication of B2BUA behaviours ca=
n be taken into account. <br>
<br clear=3D"all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Wed, Apr 7, 2010 at 8:31 AM, Elwell, =
John <span dir=3D"ltr">&lt;<a href=3D"mailto:john.elwell@siemens-enterprise=
.com" target=3D"_blank">john.elwell@siemens-enterprise.com</a>&gt;</span> w=
rote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; borde=
r-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
I agree with Dale. What happens to an OPTIONS request will not necessarily =
be reflected when an INVITE request is sent shortly afterwards. As a furthe=
r examples, in contact centres, it is usual to distribute calls cyclically =
or randomly between qualified and available agents. Sending an INVITE reque=
st to a UA determined by responses to an OPTIONS request would break that m=
odel - the issuer of the OPTIONS and INVITE requests cannot be expected to =
obey the same rotation rules as the domain proxy that routes INVITE request=
s. Issuing an INVITE request to the AOR is unlikely to result in it being d=
elivered to the UA that might be expected from analysis of responses to an =
OPTIONS request.<br>


<br>
I don&#39;t think we should work on trying to make OPTIONS work for this pu=
rpose. If we really need to do anything with OPTIONS, we should just issue =
warnings about its unsuitability for purposes other than a simple ping.<br>


<font color=3D"#888888"><br>
John<br>
</font><div><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sipcore-bounces@ietf.org" target=3D"_blank">si=
pcore-bounces@ietf.org</a><br>
</div><div>&gt; [mailto:<a href=3D"mailto:sipcore-bounces@ietf.org" target=
=3D"_blank">sipcore-bounces@ietf.org</a>] On Behalf Of WORLEY, DALE R (DALE=
)<br>
&gt; Sent: 06 April 2010 21:01<br>
&gt; To: Salvatore Loreto; <a href=3D"mailto:sipcore@ietf.org" target=3D"_b=
lank">sipcore@ietf.org</a><br>
&gt; Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?<br>
&gt;<br>
&gt; ________________________________________<br>
</div><div><div></div><div>&gt; From: <a href=3D"mailto:sipcore-bounces@iet=
f.org" target=3D"_blank">sipcore-bounces@ietf.org</a> [<a href=3D"mailto:si=
pcore-bounces@ietf.org" target=3D"_blank">sipcore-bounces@ietf.org</a>] On<=
br>

&gt; Behalf Of Salvatore Loreto [<a href=3D"mailto:salvatore.loreto@ericsso=
n.com" target=3D"_blank">salvatore.loreto@ericsson.com</a>]<br>
&gt; Sent: Thursday, April 01, 2010 3:36 AM<br>
&gt;<br>
&gt; A good mechanism to discovery UA capabilities is essential to<br>
&gt; provide meaningful service:<br>
&gt; So I am in favor of pursuing the work,<br>
&gt; in particular the possibility to use a single OPTIONS request<br>
&gt; to retrieve the capabilities from ALL UAS(s) behind a forking proxy.<b=
r>
&gt; _______________________________________<br>
&gt;<br>
&gt; I think it is a misconception to use UA discovery to provide<br>
&gt; services. =A0The difficulty is that the set of available UAs<br>
&gt; for a given destination URI changes unpredictably -- elements<br>
&gt; become inacessible (mobile phone out of range!), call routing<br>
&gt; changes due to time of day, etc. =A0Instead, SIP provides a<br>
&gt; remarkably effective system for selecting the best target of<br>
&gt; a call *at the moment the call is initiated* -- the caller<br>
&gt; declares his preferences, and the various intermediate<br>
&gt; elements combine the caller&#39;s desires with the routing<br>
&gt; information that they possess to forward the call closer to<br>
&gt; the preferred destination.<br>
&gt;<br>
&gt; As a developer, I&#39;ve had to wrestle with this. =A0In order to<br>
&gt; provide clean &quot;Is extension 1234 busy?&quot; information, our &qu=
ot;BLF<br>
&gt; agent&quot; maintains subscriptions to the &quot;reg&quot; event packa=
ge for<br>
&gt; the AOR &quot;<a href=3D"mailto:sip%3A1234@example.com" target=3D"_bla=
nk">sip:1234@example.com</a>&quot;. =A0For each reported contact<br>
&gt; for the AOR, it maintains a dialog subscription to the<br>
&gt; contact. =A0Only by tracking the &quot;reg&quot; events can it keep tr=
ack<br>
&gt; of the UAs available for that AOR. =A0And even then, it is at<br>
&gt; the mercy of network failures.<br>
&gt;<br>
&gt; Dale<br>
&gt; _______________________________________________<br>
&gt; sipcore mailing list<br>
&gt; <a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">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>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</div></div></blockquote></div><br>

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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On Wed, Apr 7, 2010 at 9:09 AM, Elwell, John <
john.elwell@siemens-enterprise.com> wrote:

> > >>I think the main use-case is when you want to contact end users.
> > >[JRE] A contact centre agent is indeed an end user.
> >
> > Yes, but not the same kind of end user :)
> [JRE] So what kind of end user is within scope of this work? How does the
> issuer of OPTIONS/INVITE requests know whether he is targeting the right
> sort of end user for the mechanism to work? I think it is dangerous to base
> mechanisms on assumptions about particular types of end user.
>
> [HE] I guess as long as the AOR addressed represents the user and the AOR's
domain portion is owned by the proxy at which the user registered with the
AOR's user-portion. Then there is no need to distinguish end users.

Throwing in any B2BUA or other anomaly will of course break things. But that
is in the nature of SIP's design.   And maybe not really relevant for this
discussion.

/Hans Erik van Elburg

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<br clear=3D"all"><br>
<br><br><div class=3D"gmail_quote">On Wed, Apr 7, 2010 at 9:09 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"margin: 0pt 0pt 0pt 0.8ex; border-left: 1=
px solid rgb(204, 204, 204); padding-left: 1ex;">
<div class=3D"im">&gt; &gt;&gt;I think the main use-case is when you want t=
o contact end users.<br>
&gt; &gt;[JRE] A contact centre agent is indeed an end user.<br>
&gt;<br>
&gt; Yes, but not the same kind of end user :)<br>
</div>[JRE] So what kind of end user is within scope of this work? How does=
 the issuer of OPTIONS/INVITE requests know whether he is targeting the rig=
ht sort of end user for the mechanism to work? I think it is dangerous to b=
ase mechanisms on assumptions about particular types of end user.<br>

<font color=3D"#888888"></font><br></blockquote><div>[HE] I guess as long a=
s the AOR addressed represents the user and the AOR&#39;s domain portion is=
 owned by the proxy at which the user registered with the AOR&#39;s user-po=
rtion. Then there is no need to distinguish end users.<br>
<br>Throwing in any B2BUA or other anomaly will of course break things. But=
 that is in the nature of SIP&#39;s design. =A0 And maybe not really releva=
nt for this discussion.<br><br>/Hans Erik van Elburg<br></div></div>

--00c09ffb4e23dfb1910483a09612--

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To: Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Wed, 7 Apr 2010 10:41:53 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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=20

> -----Original Message-----
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
> Sent: 07 April 2010 08:24
> To: Elwell, John
> Cc: WORLEY, DALE R (DALE); Salvatore Loreto; sipcore@ietf.org
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
> This is a special case. And the application you are talking=20
> about is a B2BUA, right?
[JRE] Not necessarily. A proxy can be scripted to cycle between multiple re=
gistered contacts. This doesn't make it a B2BUA.

John

>=20
> In this case the AOR addressed is an application address=20
> without registered contacts. One can argue that the OPTIONS=20
> request should return the general capabilities of the=20
> application. No OPTIONS forking is happening in this case, no=20
> registered contacts. No issue?
>=20
> In reality what the B2BUA will do with the OPTION request=20
> (forward or answer) depends on the fantasy of the B2BUA developer.
>=20
> I dont think this scenario is actually usefull in discussing=20
> whether or not to write a BCP on OPTIONS handling. Or maybe=20
> it is as implication of B2BUA behaviours can be taken into account.=20
>=20
> /Hans Erik van Elburg
>=20
>=20
>=20
> On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John=20
> <john.elwell@siemens-enterprise.com> wrote:
>=20
>=20
> 	I agree with Dale. What happens to an OPTIONS request=20
> will not necessarily be reflected when an INVITE request is=20
> sent shortly afterwards. As a further examples, in contact=20
> centres, it is usual to distribute calls cyclically or=20
> randomly between qualified and available agents. Sending an=20
> INVITE request to a UA determined by responses to an OPTIONS=20
> request would break that model - the issuer of the OPTIONS=20
> and INVITE requests cannot be expected to obey the same=20
> rotation rules as the domain proxy that routes INVITE=20
> requests. Issuing an INVITE request to the AOR is unlikely to=20
> result in it being delivered to the UA that might be expected=20
> from analysis of responses to an OPTIONS request.
> =09
> 	I don't think we should work on trying to make OPTIONS=20
> work for this purpose. If we really need to do anything with=20
> OPTIONS, we should just issue warnings about its=20
> unsuitability for purposes other than a simple ping.
> =09
> 	John
> =09
>=20
>=20
> 	> -----Original Message-----
> 	> From: sipcore-bounces@ietf.org
> =09
> 	> [mailto:sipcore-bounces@ietf.org] On Behalf Of=20
> WORLEY, DALE R (DALE)
> 	> Sent: 06 April 2010 21:01
> 	> To: Salvatore Loreto; sipcore@ietf.org
> 	> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> 	>
> 	> ________________________________________
> =09
> 	> From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On
> 	> Behalf Of Salvatore Loreto [salvatore.loreto@ericsson.com]
> 	> Sent: Thursday, April 01, 2010 3:36 AM
> 	>
> 	> A good mechanism to discovery UA capabilities is essential to
> 	> provide meaningful service:
> 	> So I am in favor of pursuing the work,
> 	> in particular the possibility to use a single OPTIONS request
> 	> to retrieve the capabilities from ALL UAS(s) behind a=20
> forking proxy.
> 	> _______________________________________
> 	>
> 	> I think it is a misconception to use UA discovery to provide
> 	> services.  The difficulty is that the set of available UAs
> 	> for a given destination URI changes unpredictably -- elements
> 	> become inacessible (mobile phone out of range!), call routing
> 	> changes due to time of day, etc.  Instead, SIP provides a
> 	> remarkably effective system for selecting the best target of
> 	> a call *at the moment the call is initiated* -- the caller
> 	> declares his preferences, and the various intermediate
> 	> elements combine the caller's desires with the routing
> 	> information that they possess to forward the call closer to
> 	> the preferred destination.
> 	>
> 	> As a developer, I've had to wrestle with this.  In order to
> 	> provide clean "Is extension 1234 busy?" information, our "BLF
> 	> agent" maintains subscriptions to the "reg" event package for
> 	> the AOR "sip:1234@example.com=20
> <mailto:sip%3A1234@example.com> ".  For each reported contact
> 	> for the AOR, it maintains a dialog subscription to the
> 	> contact.  Only by tracking the "reg" events can it keep track
> 	> of the UAs available for that AOR.  And even then, it is at
> 	> the mercy of network failures.
> 	>
> 	> Dale
> 	> _______________________________________________
> 	> 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
> =09
>=20
>=20
> =

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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Of course you can implement all kind of policies, but that makes this AOR
quite of a special user. I would argue that this is some kind of service AOR
and not a "normal" user AOR. The service AOR representing a group of
different and dynamically changing users...

Demonstrating that this is not a "normal" user.

I expect that in practice one would solve this differently... making the
case rather artificial.

/Hans Erik van Elburg

On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John <
john.elwell@siemens-enterprise.com> wrote:

>
>
> > -----Original Message-----
> > From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> > Sent: 07 April 2010 08:24
> > To: Elwell, John
> > Cc: WORLEY, DALE R (DALE); Salvatore Loreto; sipcore@ietf.org
> > Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> >
> > This is a special case. And the application you are talking
> > about is a B2BUA, right?
> [JRE] Not necessarily. A proxy can be scripted to cycle between multiple
> registered contacts. This doesn't make it a B2BUA.
>
> John
>
> >
> > In this case the AOR addressed is an application address
> > without registered contacts. One can argue that the OPTIONS
> > request should return the general capabilities of the
> > application. No OPTIONS forking is happening in this case, no
> > registered contacts. No issue?
> >
> > In reality what the B2BUA will do with the OPTION request
> > (forward or answer) depends on the fantasy of the B2BUA developer.
> >
> > I dont think this scenario is actually usefull in discussing
> > whether or not to write a BCP on OPTIONS handling. Or maybe
> > it is as implication of B2BUA behaviours can be taken into account.
> >
> > /Hans Erik van Elburg
> >
> >
> >
> > On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John
> > <john.elwell@siemens-enterprise.com> wrote:
> >
> >
> >       I agree with Dale. What happens to an OPTIONS request
> > will not necessarily be reflected when an INVITE request is
> > sent shortly afterwards. As a further examples, in contact
> > centres, it is usual to distribute calls cyclically or
> > randomly between qualified and available agents. Sending an
> > INVITE request to a UA determined by responses to an OPTIONS
> > request would break that model - the issuer of the OPTIONS
> > and INVITE requests cannot be expected to obey the same
> > rotation rules as the domain proxy that routes INVITE
> > requests. Issuing an INVITE request to the AOR is unlikely to
> > result in it being delivered to the UA that might be expected
> > from analysis of responses to an OPTIONS request.
> >
> >       I don't think we should work on trying to make OPTIONS
> > work for this purpose. If we really need to do anything with
> > OPTIONS, we should just issue warnings about its
> > unsuitability for purposes other than a simple ping.
> >
> >       John
> >
> >
> >
> >       > -----Original Message-----
> >       > From: sipcore-bounces@ietf.org
> >
> >       > [mailto:sipcore-bounces@ietf.org] On Behalf Of
> > WORLEY, DALE R (DALE)
> >       > Sent: 06 April 2010 21:01
> >       > To: Salvatore Loreto; sipcore@ietf.org
> >       > Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> >       >
> >       > ________________________________________
> >
> >       > From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On
> >       > Behalf Of Salvatore Loreto [salvatore.loreto@ericsson.com]
> >       > Sent: Thursday, April 01, 2010 3:36 AM
> >       >
> >       > A good mechanism to discovery UA capabilities is essential to
> >       > provide meaningful service:
> >       > So I am in favor of pursuing the work,
> >       > in particular the possibility to use a single OPTIONS request
> >       > to retrieve the capabilities from ALL UAS(s) behind a
> > forking proxy.
> >       > _______________________________________
> >       >
> >       > I think it is a misconception to use UA discovery to provide
> >       > services.  The difficulty is that the set of available UAs
> >       > for a given destination URI changes unpredictably -- elements
> >       > become inacessible (mobile phone out of range!), call routing
> >       > changes due to time of day, etc.  Instead, SIP provides a
> >       > remarkably effective system for selecting the best target of
> >       > a call *at the moment the call is initiated* -- the caller
> >       > declares his preferences, and the various intermediate
> >       > elements combine the caller's desires with the routing
> >       > information that they possess to forward the call closer to
> >       > the preferred destination.
> >       >
> >       > As a developer, I've had to wrestle with this.  In order to
> >       > provide clean "Is extension 1234 busy?" information, our "BLF
> >       > agent" maintains subscriptions to the "reg" event package for
> >       > the AOR "sip:1234@example.com <sip%3A1234@example.com>
> > <mailto:sip%3A1234@example.com <sip%253A1234@example.com>> ".  For each
> reported contact
> >       > for the AOR, it maintains a dialog subscription to the
> >       > contact.  Only by tracking the "reg" events can it keep track
> >       > of the UAs available for that AOR.  And even then, it is at
> >       > the mercy of network failures.
> >       >
> >       > Dale
> >       > _______________________________________________
> >       > 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
> >
> >
> >
> >
>

--00148531a91acb71940483a20ab4
Content-Type: text/html; charset=ISO-8859-1
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Of course you can implement all kind of policies, but that makes this AOR q=
uite of a special user. I would argue that this is some kind of service AOR=
 and not a &quot;normal&quot; user AOR. The service AOR representing a grou=
p of different and dynamically changing users...<br>
<br>Demonstrating that this is not a &quot;normal&quot; user.<br><br>I expe=
ct that in practice one would solve this differently... making the case rat=
her artificial.<br><br clear=3D"all">/Hans Erik van Elburg<br>
<br><div class=3D"gmail_quote">On Wed, Apr 7, 2010 at 10:41 AM, Elwell, Joh=
n <span dir=3D"ltr">&lt;<a href=3D"mailto:john.elwell@siemens-enterprise.co=
m">john.elwell@siemens-enterprise.com</a>&gt;</span> wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px =
solid rgb(204, 204, 204); padding-left: 1ex;">
<div class=3D"im"><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: Hans Erik van Elburg [mailto:<a href=3D"mailto:ietf.hanserik@gma=
il.com">ietf.hanserik@gmail.com</a>]<br>
&gt; Sent: 07 April 2010 08:24<br>
&gt; To: Elwell, John<br>
&gt; Cc: WORLEY, DALE R (DALE); Salvatore Loreto; <a href=3D"mailto:sipcore=
@ietf.org">sipcore@ietf.org</a><br>
&gt; Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?<br>
&gt;<br>
</div><div class=3D"im">&gt; This is a special case. And the application yo=
u are talking<br>
&gt; about is a B2BUA, right?<br>
</div>[JRE] Not necessarily. A proxy can be scripted to cycle between multi=
ple registered contacts. This doesn&#39;t make it a B2BUA.<br>
<br>
John<br>
<div><div></div><div class=3D"h5"><br>
&gt;<br>
&gt; In this case the AOR addressed is an application address<br>
&gt; without registered contacts. One can argue that the OPTIONS<br>
&gt; request should return the general capabilities of the<br>
&gt; application. No OPTIONS forking is happening in this case, no<br>
&gt; registered contacts. No issue?<br>
&gt;<br>
&gt; In reality what the B2BUA will do with the OPTION request<br>
&gt; (forward or answer) depends on the fantasy of the B2BUA developer.<br>
&gt;<br>
&gt; I dont think this scenario is actually usefull in discussing<br>
&gt; whether or not to write a BCP on OPTIONS handling. Or maybe<br>
&gt; it is as implication of B2BUA behaviours can be taken into account.<br=
>
&gt;<br>
&gt; /Hans Erik van Elburg<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John<br>
&gt; &lt;<a href=3D"mailto:john.elwell@siemens-enterprise.com">john.elwell@=
siemens-enterprise.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 I agree with Dale. What happens to an OPTIONS request<br>
&gt; will not necessarily be reflected when an INVITE request is<br>
&gt; sent shortly afterwards. As a further examples, in contact<br>
&gt; centres, it is usual to distribute calls cyclically or<br>
&gt; randomly between qualified and available agents. Sending an<br>
&gt; INVITE request to a UA determined by responses to an OPTIONS<br>
&gt; request would break that model - the issuer of the OPTIONS<br>
&gt; and INVITE requests cannot be expected to obey the same<br>
&gt; rotation rules as the domain proxy that routes INVITE<br>
&gt; requests. Issuing an INVITE request to the AOR is unlikely to<br>
&gt; result in it being delivered to the UA that might be expected<br>
&gt; from analysis of responses to an OPTIONS request.<br>
&gt;<br>
&gt; =A0 =A0 =A0 I don&#39;t think we should work on trying to make OPTIONS=
<br>
&gt; work for this purpose. If we really need to do anything with<br>
&gt; OPTIONS, we should just issue warnings about its<br>
&gt; unsuitability for purposes other than a simple ping.<br>
&gt;<br>
&gt; =A0 =A0 =A0 John<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 &gt; -----Original Message-----<br>
&gt; =A0 =A0 =A0 &gt; From: <a href=3D"mailto:sipcore-bounces@ietf.org">sip=
core-bounces@ietf.org</a><br>
&gt;<br>
&gt; =A0 =A0 =A0 &gt; [mailto:<a href=3D"mailto:sipcore-bounces@ietf.org">s=
ipcore-bounces@ietf.org</a>] On Behalf Of<br>
&gt; WORLEY, DALE R (DALE)<br>
&gt; =A0 =A0 =A0 &gt; Sent: 06 April 2010 21:01<br>
&gt; =A0 =A0 =A0 &gt; To: Salvatore Loreto; <a href=3D"mailto:sipcore@ietf.=
org">sipcore@ietf.org</a><br>
&gt; =A0 =A0 =A0 &gt; Subject: Re: [sipcore] SIP OPTIONS: Shall we work on =
this?<br>
&gt; =A0 =A0 =A0 &gt;<br>
&gt; =A0 =A0 =A0 &gt; ________________________________________<br>
&gt;<br>
&gt; =A0 =A0 =A0 &gt; From: <a href=3D"mailto:sipcore-bounces@ietf.org">sip=
core-bounces@ietf.org</a> [<a href=3D"mailto:sipcore-bounces@ietf.org">sipc=
ore-bounces@ietf.org</a>] On<br>
&gt; =A0 =A0 =A0 &gt; Behalf Of Salvatore Loreto [<a href=3D"mailto:salvato=
re.loreto@ericsson.com">salvatore.loreto@ericsson.com</a>]<br>
&gt; =A0 =A0 =A0 &gt; Sent: Thursday, April 01, 2010 3:36 AM<br>
&gt; =A0 =A0 =A0 &gt;<br>
&gt; =A0 =A0 =A0 &gt; A good mechanism to discovery UA capabilities is esse=
ntial to<br>
&gt; =A0 =A0 =A0 &gt; provide meaningful service:<br>
&gt; =A0 =A0 =A0 &gt; So I am in favor of pursuing the work,<br>
&gt; =A0 =A0 =A0 &gt; in particular the possibility to use a single OPTIONS=
 request<br>
&gt; =A0 =A0 =A0 &gt; to retrieve the capabilities from ALL UAS(s) behind a=
<br>
&gt; forking proxy.<br>
&gt; =A0 =A0 =A0 &gt; _______________________________________<br>
&gt; =A0 =A0 =A0 &gt;<br>
&gt; =A0 =A0 =A0 &gt; I think it is a misconception to use UA discovery to =
provide<br>
&gt; =A0 =A0 =A0 &gt; services. =A0The difficulty is that the set of availa=
ble UAs<br>
&gt; =A0 =A0 =A0 &gt; for a given destination URI changes unpredictably -- =
elements<br>
&gt; =A0 =A0 =A0 &gt; become inacessible (mobile phone out of range!), call=
 routing<br>
&gt; =A0 =A0 =A0 &gt; changes due to time of day, etc. =A0Instead, SIP prov=
ides a<br>
&gt; =A0 =A0 =A0 &gt; remarkably effective system for selecting the best ta=
rget of<br>
&gt; =A0 =A0 =A0 &gt; a call *at the moment the call is initiated* -- the c=
aller<br>
&gt; =A0 =A0 =A0 &gt; declares his preferences, and the various intermediat=
e<br>
&gt; =A0 =A0 =A0 &gt; elements combine the caller&#39;s desires with the ro=
uting<br>
&gt; =A0 =A0 =A0 &gt; information that they possess to forward the call clo=
ser to<br>
&gt; =A0 =A0 =A0 &gt; the preferred destination.<br>
&gt; =A0 =A0 =A0 &gt;<br>
&gt; =A0 =A0 =A0 &gt; As a developer, I&#39;ve had to wrestle with this. =
=A0In order to<br>
&gt; =A0 =A0 =A0 &gt; provide clean &quot;Is extension 1234 busy?&quot; inf=
ormation, our &quot;BLF<br>
&gt; =A0 =A0 =A0 &gt; agent&quot; maintains subscriptions to the &quot;reg&=
quot; event package for<br>
&gt; =A0 =A0 =A0 &gt; the AOR &quot;<a href=3D"mailto:sip%3A1234@example.co=
m">sip:1234@example.com</a><br>
</div></div>&gt; &lt;mailto:<a href=3D"mailto:sip%253A1234@example.com">sip=
%3A1234@example.com</a>&gt; &quot;. =A0For each reported contact<br>
<div><div></div><div class=3D"h5">&gt; =A0 =A0 =A0 &gt; for the AOR, it mai=
ntains a dialog subscription to the<br>
&gt; =A0 =A0 =A0 &gt; contact. =A0Only by tracking the &quot;reg&quot; even=
ts can it keep track<br>
&gt; =A0 =A0 =A0 &gt; of the UAs available for that AOR. =A0And even then, =
it is at<br>
&gt; =A0 =A0 =A0 &gt; the mercy of network failures.<br>
&gt; =A0 =A0 =A0 &gt;<br>
&gt; =A0 =A0 =A0 &gt; Dale<br>
&gt; =A0 =A0 =A0 &gt; _______________________________________________<br>
&gt; =A0 =A0 =A0 &gt; sipcore mailing list<br>
&gt; =A0 =A0 =A0 &gt; <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org<=
/a><br>
&gt; =A0 =A0 =A0 &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipc=
ore" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><br=
>
&gt; =A0 =A0 =A0 &gt;<br>
&gt; =A0 =A0 =A0 _______________________________________________<br>
&gt; =A0 =A0 =A0 sipcore mailing list<br>
&gt; =A0 =A0 =A0 <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><b=
r>
&gt; =A0 =A0 =A0 <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; </div></div></blockquote></div><br>

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From christer.holmberg@ericsson.com  Wed Apr  7 03:26:48 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hans Erik van Elburg <ietf.hanserik@gmail.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>
Date: Wed, 7 Apr 2010 12:26:25 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

The forking proxy can also have policies what it returns in the 3xx, depend=
ing on what the end-users actually represent.

And, also with other mechanisms the returned data per end-user may depend o=
n what type of end-point it represents. I don't think that is bound to a sp=
ecific mechanism as such.

Regards,

Christer




________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Hans Erik van Elburg
Sent: 7. huhtikuuta 2010 12:18
To: Elwell, John
Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?

Of course you can implement all kind of policies, but that makes this AOR q=
uite of a special user. I would argue that this is some kind of service AOR=
 and not a "normal" user AOR. The service AOR representing a group of diffe=
rent and dynamically changing users...

Demonstrating that this is not a "normal" user.

I expect that in practice one would solve this differently... making the ca=
se rather artificial.

/Hans Erik van Elburg

On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John <john.elwell@siemens-enterpri=
se.com<mailto:john.elwell@siemens-enterprise.com>> wrote:


> -----Original Message-----
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com<mailto:ietf.ha=
nserik@gmail.com>]
> Sent: 07 April 2010 08:24
> To: Elwell, John
> Cc: WORLEY, DALE R (DALE); Salvatore Loreto; sipcore@ietf.org<mailto:sipc=
ore@ietf.org>
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>
> This is a special case. And the application you are talking
> about is a B2BUA, right?
[JRE] Not necessarily. A proxy can be scripted to cycle between multiple re=
gistered contacts. This doesn't make it a B2BUA.

John

>
> In this case the AOR addressed is an application address
> without registered contacts. One can argue that the OPTIONS
> request should return the general capabilities of the
> application. No OPTIONS forking is happening in this case, no
> registered contacts. No issue?
>
> In reality what the B2BUA will do with the OPTION request
> (forward or answer) depends on the fantasy of the B2BUA developer.
>
> I dont think this scenario is actually usefull in discussing
> whether or not to write a BCP on OPTIONS handling. Or maybe
> it is as implication of B2BUA behaviours can be taken into account.
>
> /Hans Erik van Elburg
>
>
>
> On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John
> <john.elwell@siemens-enterprise.com<mailto:john.elwell@siemens-enterprise=
.com>> wrote:
>
>
>       I agree with Dale. What happens to an OPTIONS request
> will not necessarily be reflected when an INVITE request is
> sent shortly afterwards. As a further examples, in contact
> centres, it is usual to distribute calls cyclically or
> randomly between qualified and available agents. Sending an
> INVITE request to a UA determined by responses to an OPTIONS
> request would break that model - the issuer of the OPTIONS
> and INVITE requests cannot be expected to obey the same
> rotation rules as the domain proxy that routes INVITE
> requests. Issuing an INVITE request to the AOR is unlikely to
> result in it being delivered to the UA that might be expected
> from analysis of responses to an OPTIONS request.
>
>       I don't think we should work on trying to make OPTIONS
> work for this purpose. If we really need to do anything with
> OPTIONS, we should just issue warnings about its
> unsuitability for purposes other than a simple ping.
>
>       John
>
>
>
>       > -----Original Message-----
>       > From: sipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org>
>
>       > [mailto:sipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org>=
] On Behalf Of
> WORLEY, DALE R (DALE)
>       > Sent: 06 April 2010 21:01
>       > To: Salvatore Loreto; sipcore@ietf.org<mailto:sipcore@ietf.org>
>       > Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>       >
>       > ________________________________________
>
>       > From: sipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org> [=
sipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org>] On
>       > Behalf Of Salvatore Loreto [salvatore.loreto@ericsson.com<mailto:=
salvatore.loreto@ericsson.com>]
>       > Sent: Thursday, April 01, 2010 3:36 AM
>       >
>       > A good mechanism to discovery UA capabilities is essential to
>       > provide meaningful service:
>       > So I am in favor of pursuing the work,
>       > in particular the possibility to use a single OPTIONS request
>       > to retrieve the capabilities from ALL UAS(s) behind a
> forking proxy.
>       > _______________________________________
>       >
>       > I think it is a misconception to use UA discovery to provide
>       > services.  The difficulty is that the set of available UAs
>       > for a given destination URI changes unpredictably -- elements
>       > become inacessible (mobile phone out of range!), call routing
>       > changes due to time of day, etc.  Instead, SIP provides a
>       > remarkably effective system for selecting the best target of
>       > a call *at the moment the call is initiated* -- the caller
>       > declares his preferences, and the various intermediate
>       > elements combine the caller's desires with the routing
>       > information that they possess to forward the call closer to
>       > the preferred destination.
>       >
>       > As a developer, I've had to wrestle with this.  In order to
>       > provide clean "Is extension 1234 busy?" information, our "BLF
>       > agent" maintains subscriptions to the "reg" event package for
>       > the AOR "sip:1234@example.com<mailto:sip%3A1234@example.com>
> <mailto:sip%3A1234@example.com<mailto:sip%253A1234@example.com>> ".  For =
each reported contact
>       > for the AOR, it maintains a dialog subscription to the
>       > contact.  Only by tracking the "reg" events can it keep track
>       > of the UAs available for that AOR.  And even then, it is at
>       > the mercy of network failures.
>       >
>       > Dale
>       > _______________________________________________
>       > sipcore mailing list
>       > sipcore@ietf.org<mailto:sipcore@ietf.org>
>       > https://www.ietf.org/mailman/listinfo/sipcore
>       >
>       _______________________________________________
>       sipcore mailing list
>       sipcore@ietf.org<mailto:sipcore@ietf.org>
>       https://www.ietf.org/mailman/listinfo/sipcore
>
>
>
>


--_000_FF84A09F50A6DC48ACB6714F4666CC745E21C7C978ESESSCMS0354e_
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<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D062102410-07042010>Hi,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D062102410-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D062102410-07042010>The forking proxy can also have policies what it=
=20
returns in the 3xx, depending on what the end-users actually=20
represent.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D062102410-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D062102410-07042010>And, also with other mechanisms the returned dat=
a per=20
end-user may depend on what type of end-point it represents. I don't think =
that=20
is bound to a specific mechanism as such.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D062102410-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D062102410-07042010>Regards,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D062102410-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D062102410-07042010>Christer</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sipcore-bounces@ietf.org=20
  [mailto:sipcore-bounces@ietf.org] <B>On Behalf Of </B>Hans Erik van=20
  Elburg<BR><B>Sent:</B> 7. huhtikuuta 2010 12:18<BR><B>To:</B> Elwell,=20
  John<BR><B>Cc:</B> WORLEY, DALE R (DALE); sipcore@ietf.org<BR><B>Subject:=
</B>=20
  Re: [sipcore] SIP OPTIONS: Shall we work on this?<BR></FONT><BR></DIV>
  <DIV></DIV>Of course you can implement all kind of policies, but that mak=
es=20
  this AOR quite of a special user. I would argue that this is some kind of=
=20
  service AOR and not a "normal" user AOR. The service AOR representing a g=
roup=20
  of different and dynamically changing users...<BR><BR>Demonstrating that =
this=20
  is not a "normal" user.<BR><BR>I expect that in practice one would solve =
this=20
  differently... making the case rather artificial.<BR><BR clear=3Dall>/Han=
s Erik=20
  van Elburg<BR><BR>
  <DIV class=3Dgmail_quote>On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John <S=
PAN=20
  dir=3Dltr>&lt;<A=20
  href=3D"mailto:john.elwell@siemens-enterprise.com">john.elwell@siemens-en=
terprise.com</A>&gt;</SPAN>=20
  wrote:<BR>
  <BLOCKQUOTE class=3Dgmail_quote=20
  style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(2=
04,204,204) 1px solid">
    <DIV class=3Dim><BR><BR>&gt; -----Original Message-----<BR>&gt; From: H=
ans=20
    Erik van Elburg [mailto:<A=20
    href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail.com</A>]<BR=
>&gt;=20
    Sent: 07 April 2010 08:24<BR>&gt; To: Elwell, John<BR>&gt; Cc: WORLEY, =
DALE=20
    R (DALE); Salvatore Loreto; <A=20
    href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</A><BR>&gt; Subject: =
Re:=20
    [sipcore] SIP OPTIONS: Shall we work on this?<BR>&gt;<BR></DIV>
    <DIV class=3Dim>&gt; This is a special case. And the application you ar=
e=20
    talking<BR>&gt; about is a B2BUA, right?<BR></DIV>[JRE] Not necessarily=
. A=20
    proxy can be scripted to cycle between multiple registered contacts. Th=
is=20
    doesn't make it a B2BUA.<BR><BR>John<BR>
    <DIV>
    <DIV></DIV>
    <DIV class=3Dh5><BR>&gt;<BR>&gt; In this case the AOR addressed is an=20
    application address<BR>&gt; without registered contacts. One can argue =
that=20
    the OPTIONS<BR>&gt; request should return the general capabilities of=20
    the<BR>&gt; application. No OPTIONS forking is happening in this case,=
=20
    no<BR>&gt; registered contacts. No issue?<BR>&gt;<BR>&gt; In reality wh=
at=20
    the B2BUA will do with the OPTION request<BR>&gt; (forward or answer)=20
    depends on the fantasy of the B2BUA developer.<BR>&gt;<BR>&gt; I dont t=
hink=20
    this scenario is actually usefull in discussing<BR>&gt; whether or not =
to=20
    write a BCP on OPTIONS handling. Or maybe<BR>&gt; it is as implication =
of=20
    B2BUA behaviours can be taken into account.<BR>&gt;<BR>&gt; /Hans Erik =
van=20
    Elburg<BR>&gt;<BR>&gt;<BR>&gt;<BR>&gt; On Wed, Apr 7, 2010 at 8:31 AM,=
=20
    Elwell, John<BR>&gt; &lt;<A=20
    href=3D"mailto:john.elwell@siemens-enterprise.com">john.elwell@siemens-=
enterprise.com</A>&gt;=20
    wrote:<BR>&gt;<BR>&gt;<BR>&gt; &nbsp; &nbsp; &nbsp; I agree with Dale. =
What=20
    happens to an OPTIONS request<BR>&gt; will not necessarily be reflected=
 when=20
    an INVITE request is<BR>&gt; sent shortly afterwards. As a further exam=
ples,=20
    in contact<BR>&gt; centres, it is usual to distribute calls cyclically=
=20
    or<BR>&gt; randomly between qualified and available agents. Sending=20
    an<BR>&gt; INVITE request to a UA determined by responses to an=20
    OPTIONS<BR>&gt; request would break that model - the issuer of the=20
    OPTIONS<BR>&gt; and INVITE requests cannot be expected to obey the=20
    same<BR>&gt; rotation rules as the domain proxy that routes INVITE<BR>&=
gt;=20
    requests. Issuing an INVITE request to the AOR is unlikely to<BR>&gt; r=
esult=20
    in it being delivered to the UA that might be expected<BR>&gt; from ana=
lysis=20
    of responses to an OPTIONS request.<BR>&gt;<BR>&gt; &nbsp; &nbsp; &nbsp=
; I=20
    don't think we should work on trying to make OPTIONS<BR>&gt; work for t=
his=20
    purpose. If we really need to do anything with<BR>&gt; OPTIONS, we shou=
ld=20
    just issue warnings about its<BR>&gt; unsuitability for purposes other =
than=20
    a simple ping.<BR>&gt;<BR>&gt; &nbsp; &nbsp; &nbsp;=20
    John<BR>&gt;<BR>&gt;<BR>&gt;<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; -----Ori=
ginal=20
    Message-----<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; From: <A=20
    href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</A><B=
R>&gt;<BR>&gt;=20
    &nbsp; &nbsp; &nbsp; &gt; [mailto:<A=20
    href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</A>] =
On=20
    Behalf Of<BR>&gt; WORLEY, DALE R (DALE)<BR>&gt; &nbsp; &nbsp; &nbsp; &g=
t;=20
    Sent: 06 April 2010 21:01<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; To: Salvato=
re=20
    Loreto; <A href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</A><BR>&gt=
;=20
    &nbsp; &nbsp; &nbsp; &gt; Subject: Re: [sipcore] SIP OPTIONS: Shall we =
work=20
    on this?<BR>&gt; &nbsp; &nbsp; &nbsp; &gt;<BR>&gt; &nbsp; &nbsp; &nbsp;=
 &gt;=20
    ________________________________________<BR>&gt;<BR>&gt; &nbsp; &nbsp;=
=20
    &nbsp; &gt; From: <A=20
    href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</A> [=
<A=20
    href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</A>]=
=20
    On<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; Behalf Of Salvatore Loreto [<A=20
    href=3D"mailto:salvatore.loreto@ericsson.com">salvatore.loreto@ericsson=
.com</A>]<BR>&gt;=20
    &nbsp; &nbsp; &nbsp; &gt; Sent: Thursday, April 01, 2010 3:36 AM<BR>&gt=
;=20
    &nbsp; &nbsp; &nbsp; &gt;<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; A good mech=
anism=20
    to discovery UA capabilities is essential to<BR>&gt; &nbsp; &nbsp; &nbs=
p;=20
    &gt; provide meaningful service:<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; So I=
 am=20
    in favor of pursuing the work,<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; in=20
    particular the possibility to use a single OPTIONS request<BR>&gt; &nbs=
p;=20
    &nbsp; &nbsp; &gt; to retrieve the capabilities from ALL UAS(s) behind=
=20
    a<BR>&gt; forking proxy.<BR>&gt; &nbsp; &nbsp; &nbsp; &gt;=20
    _______________________________________<BR>&gt; &nbsp; &nbsp; &nbsp;=20
    &gt;<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; I think it is a misconception to=
 use=20
    UA discovery to provide<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; services.=20
    &nbsp;The difficulty is that the set of available UAs<BR>&gt; &nbsp; &n=
bsp;=20
    &nbsp; &gt; for a given destination URI changes unpredictably --=20
    elements<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; become inacessible (mobile p=
hone=20
    out of range!), call routing<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; changes =
due=20
    to time of day, etc. &nbsp;Instead, SIP provides a<BR>&gt; &nbsp; &nbsp=
;=20
    &nbsp; &gt; remarkably effective system for selecting the best target=20
    of<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; a call *at the moment the call is=
=20
    initiated* -- the caller<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; declares his=
=20
    preferences, and the various intermediate<BR>&gt; &nbsp; &nbsp; &nbsp; =
&gt;=20
    elements combine the caller's desires with the routing<BR>&gt; &nbsp; &=
nbsp;=20
    &nbsp; &gt; information that they possess to forward the call closer=20
    to<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; the preferred destination.<BR>&gt;=
=20
    &nbsp; &nbsp; &nbsp; &gt;<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; As a develo=
per,=20
    I've had to wrestle with this. &nbsp;In order to<BR>&gt; &nbsp; &nbsp;=
=20
    &nbsp; &gt; provide clean "Is extension 1234 busy?" information, our=20
    "BLF<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; agent" maintains subscriptions t=
o the=20
    "reg" event package for<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; the AOR "<A=20
    href=3D"mailto:sip%3A1234@example.com">sip:1234@example.com</A><BR></DI=
V></DIV>&gt;=20
    &lt;mailto:<A=20
    href=3D"mailto:sip%253A1234@example.com">sip%3A1234@example.com</A>&gt;=
 ".=20
    &nbsp;For each reported contact<BR>
    <DIV>
    <DIV></DIV>
    <DIV class=3Dh5>&gt; &nbsp; &nbsp; &nbsp; &gt; for the AOR, it maintain=
s a=20
    dialog subscription to the<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; contact.=20
    &nbsp;Only by tracking the "reg" events can it keep track<BR>&gt; &nbsp=
;=20
    &nbsp; &nbsp; &gt; of the UAs available for that AOR. &nbsp;And even th=
en,=20
    it is at<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; the mercy of network=20
    failures.<BR>&gt; &nbsp; &nbsp; &nbsp; &gt;<BR>&gt; &nbsp; &nbsp; &nbsp=
;=20
    &gt; Dale<BR>&gt; &nbsp; &nbsp; &nbsp; &gt;=20
    _______________________________________________<BR>&gt; &nbsp; &nbsp; &=
nbsp;=20
    &gt; sipcore mailing list<BR>&gt; &nbsp; &nbsp; &nbsp; &gt; <A=20
    href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</A><BR>&gt; &nbsp; &n=
bsp;=20
    &nbsp; &gt; <A href=3D"https://www.ietf.org/mailman/listinfo/sipcore"=20
    target=3D_blank>https://www.ietf.org/mailman/listinfo/sipcore</A><BR>&g=
t;=20
    &nbsp; &nbsp; &nbsp; &gt;<BR>&gt; &nbsp; &nbsp; &nbsp;=20
    _______________________________________________<BR>&gt; &nbsp; &nbsp; &=
nbsp;=20
    sipcore mailing list<BR>&gt; &nbsp; &nbsp; &nbsp; <A=20
    href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</A><BR>&gt; &nbsp; &n=
bsp;=20
    &nbsp; <A href=3D"https://www.ietf.org/mailman/listinfo/sipcore"=20
    target=3D_blank>https://www.ietf.org/mailman/listinfo/sipcore</A><BR>&g=
t;<BR>&gt;<BR>&gt;<BR>&gt;=20
    </DIV></DIV></BLOCKQUOTE></DIV><BR></BLOCKQUOTE></BODY></HTML>

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To: Christer Holmberg <christer.holmberg@ericsson.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Wed, 7 Apr 2010 13:04:09 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 07 April 2010 11:26
> To: Hans Erik van Elburg; Elwell, John
> Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
> Hi,
> =20
> The forking proxy can also have policies what it returns in=20
> the 3xx, depending on what the end-users actually represent.
> =20
> And, also with other mechanisms the returned data per=20
> end-user may depend on what type of end-point it represents.=20
> I don't think that is bound to a specific mechanism as such.
[JRE] Quite true. But Dale's point still holds - you might as well just sen=
d the INVITE request and see what happens.

John

> =20
> Regards,
> =20
> Christer
> =20
> =20
> =20
>=20
>=20
> ________________________________
>=20
> 	From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Hans Erik van Elburg
> 	Sent: 7. huhtikuuta 2010 12:18
> 	To: Elwell, John
> 	Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> 	Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> =09
> =09
> 	Of course you can implement all kind of policies, but=20
> that makes this AOR quite of a special user. I would argue=20
> that this is some kind of service AOR and not a "normal" user=20
> AOR. The service AOR representing a group of different and=20
> dynamically changing users...
> =09
> 	Demonstrating that this is not a "normal" user.
> =09
> 	I expect that in practice one would solve this=20
> differently... making the case rather artificial.
> =09
> 	/Hans Erik van Elburg
> =09
> =09
> 	On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John=20
> <john.elwell@siemens-enterprise.com> wrote:
> =09
>=20
>=20
>=20
> 		> -----Original Message-----
> 		> From: Hans Erik van Elburg=20
> [mailto:ietf.hanserik@gmail.com]
> 		> Sent: 07 April 2010 08:24
> 		> To: Elwell, John
> 		> Cc: WORLEY, DALE R (DALE); Salvatore Loreto;=20
> sipcore@ietf.org
> 		> Subject: Re: [sipcore] SIP OPTIONS: Shall we=20
> work on this?
> 		>
> 	=09
> 		> This is a special case. And the application=20
> you are talking
> 		> about is a B2BUA, right?
> 	=09
> 		[JRE] Not necessarily. A proxy can be scripted=20
> to cycle between multiple registered contacts. This doesn't=20
> make it a B2BUA.
> 	=09
> 		John
> 	=09
>=20
> 		>
> 		> In this case the AOR addressed is an=20
> application address
> 		> without registered contacts. One can argue=20
> that the OPTIONS
> 		> request should return the general capabilities of the
> 		> application. No OPTIONS forking is happening=20
> in this case, no
> 		> registered contacts. No issue?
> 		>
> 		> In reality what the B2BUA will do with the=20
> OPTION request
> 		> (forward or answer) depends on the fantasy of=20
> the B2BUA developer.
> 		>
> 		> I dont think this scenario is actually=20
> usefull in discussing
> 		> whether or not to write a BCP on OPTIONS=20
> handling. Or maybe
> 		> it is as implication of B2BUA behaviours can=20
> be taken into account.
> 		>
> 		> /Hans Erik van Elburg
> 		>
> 		>
> 		>
> 		> On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John
> 		> <john.elwell@siemens-enterprise.com> wrote:
> 		>
> 		>
> 		>       I agree with Dale. What happens to an=20
> OPTIONS request
> 		> will not necessarily be reflected when an=20
> INVITE request is
> 		> sent shortly afterwards. As a further=20
> examples, in contact
> 		> centres, it is usual to distribute calls cyclically or
> 		> randomly between qualified and available=20
> agents. Sending an
> 		> INVITE request to a UA determined by=20
> responses to an OPTIONS
> 		> request would break that model - the issuer=20
> of the OPTIONS
> 		> and INVITE requests cannot be expected to=20
> obey the same
> 		> rotation rules as the domain proxy that routes INVITE
> 		> requests. Issuing an INVITE request to the=20
> AOR is unlikely to
> 		> result in it being delivered to the UA that=20
> might be expected
> 		> from analysis of responses to an OPTIONS request.
> 		>
> 		>       I don't think we should work on trying=20
> to make OPTIONS
> 		> work for this purpose. If we really need to=20
> do anything with
> 		> OPTIONS, we should just issue warnings about its
> 		> unsuitability for purposes other than a simple ping.
> 		>
> 		>       John
> 		>
> 		>
> 		>
> 		>       > -----Original Message-----
> 		>       > From: sipcore-bounces@ietf.org
> 		>
> 		>       > [mailto:sipcore-bounces@ietf.org] On Behalf Of
> 		> WORLEY, DALE R (DALE)
> 		>       > Sent: 06 April 2010 21:01
> 		>       > To: Salvatore Loreto; sipcore@ietf.org
> 		>       > Subject: Re: [sipcore] SIP OPTIONS:=20
> Shall we work on this?
> 		>       >
> 		>       > ________________________________________
> 		>
> 		>       > From: sipcore-bounces@ietf.org=20
> [sipcore-bounces@ietf.org] On
> 		>       > Behalf Of Salvatore Loreto=20
> [salvatore.loreto@ericsson.com]
> 		>       > Sent: Thursday, April 01, 2010 3:36 AM
> 		>       >
> 		>       > A good mechanism to discovery UA=20
> capabilities is essential to
> 		>       > provide meaningful service:
> 		>       > So I am in favor of pursuing the work,
> 		>       > in particular the possibility to use=20
> a single OPTIONS request
> 		>       > to retrieve the capabilities from ALL=20
> UAS(s) behind a
> 		> forking proxy.
> 		>       > _______________________________________
> 		>       >
> 		>       > I think it is a misconception to use=20
> UA discovery to provide
> 		>       > services.  The difficulty is that the=20
> set of available UAs
> 		>       > for a given destination URI changes=20
> unpredictably -- elements
> 		>       > become inacessible (mobile phone out=20
> of range!), call routing
> 		>       > changes due to time of day, etc. =20
> Instead, SIP provides a
> 		>       > remarkably effective system for=20
> selecting the best target of
> 		>       > a call *at the moment the call is=20
> initiated* -- the caller
> 		>       > declares his preferences, and the=20
> various intermediate
> 		>       > elements combine the caller's desires=20
> with the routing
> 		>       > information that they possess to=20
> forward the call closer to
> 		>       > the preferred destination.
> 		>       >
> 		>       > As a developer, I've had to wrestle=20
> with this.  In order to
> 		>       > provide clean "Is extension 1234=20
> busy?" information, our "BLF
> 		>       > agent" maintains subscriptions to the=20
> "reg" event package for
> 		>       > the AOR "sip:1234@example.com=20
> <mailto:sip%3A1234@example.com>=20
> 	=09
> 		> <mailto:sip%3A1234@example.com=20
> <mailto:sip%253A1234@example.com> > ".  For each reported contact
> 	=09
> 		>       > for the AOR, it maintains a dialog=20
> subscription to the
> 		>       > contact.  Only by tracking the "reg"=20
> events can it keep track
> 		>       > of the UAs available for that AOR. =20
> And even then, it is at
> 		>       > the mercy of network failures.
> 		>       >
> 		>       > Dale
> 		>       >=20
> _______________________________________________
> 		>       > 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
> 		>
> 		>
> 		>
> 		>=20
>=20
>=20
> =

From christer.holmberg@ericsson.com  Wed Apr  7 04:08:40 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Wed, 7 Apr 2010 13:08:33 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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X-List-Received-Date: Wed, 07 Apr 2010 11:08:40 -0000

Hi,=20

>>The forking proxy can also have policies what it returns in the 3xx,=20
>>depending on what the end-users actually represent.
>> =20
>>And, also with other mechanisms the returned data per end-user may=20
>>depend on what type of end-point it represents.
>>I don't think that is bound to a specific mechanism as such.
>[JRE] Quite true. But Dale's point still holds - you might as well just se=
nd the INVITE request and see what happens.

Of course. But, as I said, that is really bad protocol design, and it may t=
rigger unwanted resource reservation actions etc in the network.

And, there could even be cases where the INVITE doesn't fail, but you don't=
 get the optimized user experience since the request may have looked differ=
ent should you have known what the receiver supports.

Regards,

Christer







> > ________________________________
> >=20
> > 	From: sipcore-bounces@ietf.org=20
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Hans Erik van Elburg
> > 	Sent: 7. huhtikuuta 2010 12:18
> > 	To: Elwell, John
> > 	Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> > 	Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> > =09
> > =09
> > 	Of course you can implement all kind of policies, but=20
> > that makes this AOR quite of a special user. I would argue=20
> > that this is some kind of service AOR and not a "normal" user=20
> > AOR. The service AOR representing a group of different and=20
> > dynamically changing users...
> > =09
> > 	Demonstrating that this is not a "normal" user.
> > =09
> > 	I expect that in practice one would solve this=20
> > differently... making the case rather artificial.
> > =09
> > 	/Hans Erik van Elburg
> > =09
> > =09
> > 	On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John=20
> > <john.elwell@siemens-enterprise.com> wrote:
> > =09
> >=20
> >=20
> >=20
> > 		> -----Original Message-----
> > 		> From: Hans Erik van Elburg=20
> > [mailto:ietf.hanserik@gmail.com]
> > 		> Sent: 07 April 2010 08:24
> > 		> To: Elwell, John
> > 		> Cc: WORLEY, DALE R (DALE); Salvatore Loreto;=20
> > sipcore@ietf.org
> > 		> Subject: Re: [sipcore] SIP OPTIONS: Shall we=20
> > work on this?
> > 		>
> > 	=09
> > 		> This is a special case. And the application=20
> > you are talking
> > 		> about is a B2BUA, right?
> > 	=09
> > 		[JRE] Not necessarily. A proxy can be scripted=20
> > to cycle between multiple registered contacts. This doesn't=20
> > make it a B2BUA.
> > 	=09
> > 		John
> > 	=09
> >=20
> > 		>
> > 		> In this case the AOR addressed is an=20
> > application address
> > 		> without registered contacts. One can argue=20
> > that the OPTIONS
> > 		> request should return the general capabilities of the
> > 		> application. No OPTIONS forking is happening=20
> > in this case, no
> > 		> registered contacts. No issue?
> > 		>
> > 		> In reality what the B2BUA will do with the=20
> > OPTION request
> > 		> (forward or answer) depends on the fantasy of=20
> > the B2BUA developer.
> > 		>
> > 		> I dont think this scenario is actually=20
> > usefull in discussing
> > 		> whether or not to write a BCP on OPTIONS=20
> > handling. Or maybe
> > 		> it is as implication of B2BUA behaviours can=20
> > be taken into account.
> > 		>
> > 		> /Hans Erik van Elburg
> > 		>
> > 		>
> > 		>
> > 		> On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John
> > 		> <john.elwell@siemens-enterprise.com> wrote:
> > 		>
> > 		>
> > 		>       I agree with Dale. What happens to an=20
> > OPTIONS request
> > 		> will not necessarily be reflected when an=20
> > INVITE request is
> > 		> sent shortly afterwards. As a further=20
> > examples, in contact
> > 		> centres, it is usual to distribute calls cyclically or
> > 		> randomly between qualified and available=20
> > agents. Sending an
> > 		> INVITE request to a UA determined by=20
> > responses to an OPTIONS
> > 		> request would break that model - the issuer=20
> > of the OPTIONS
> > 		> and INVITE requests cannot be expected to=20
> > obey the same
> > 		> rotation rules as the domain proxy that routes INVITE
> > 		> requests. Issuing an INVITE request to the=20
> > AOR is unlikely to
> > 		> result in it being delivered to the UA that=20
> > might be expected
> > 		> from analysis of responses to an OPTIONS request.
> > 		>
> > 		>       I don't think we should work on trying=20
> > to make OPTIONS
> > 		> work for this purpose. If we really need to=20
> > do anything with
> > 		> OPTIONS, we should just issue warnings about its
> > 		> unsuitability for purposes other than a simple ping.
> > 		>
> > 		>       John
> > 		>
> > 		>
> > 		>
> > 		>       > -----Original Message-----
> > 		>       > From: sipcore-bounces@ietf.org
> > 		>
> > 		>       > [mailto:sipcore-bounces@ietf.org] On Behalf Of
> > 		> WORLEY, DALE R (DALE)
> > 		>       > Sent: 06 April 2010 21:01
> > 		>       > To: Salvatore Loreto; sipcore@ietf.org
> > 		>       > Subject: Re: [sipcore] SIP OPTIONS:=20
> > Shall we work on this?
> > 		>       >
> > 		>       > ________________________________________
> > 		>
> > 		>       > From: sipcore-bounces@ietf.org=20
> > [sipcore-bounces@ietf.org] On
> > 		>       > Behalf Of Salvatore Loreto=20
> > [salvatore.loreto@ericsson.com]
> > 		>       > Sent: Thursday, April 01, 2010 3:36 AM
> > 		>       >
> > 		>       > A good mechanism to discovery UA=20
> > capabilities is essential to
> > 		>       > provide meaningful service:
> > 		>       > So I am in favor of pursuing the work,
> > 		>       > in particular the possibility to use=20
> > a single OPTIONS request
> > 		>       > to retrieve the capabilities from ALL=20
> > UAS(s) behind a
> > 		> forking proxy.
> > 		>       > _______________________________________
> > 		>       >
> > 		>       > I think it is a misconception to use=20
> > UA discovery to provide
> > 		>       > services.  The difficulty is that the=20
> > set of available UAs
> > 		>       > for a given destination URI changes=20
> > unpredictably -- elements
> > 		>       > become inacessible (mobile phone out=20
> > of range!), call routing
> > 		>       > changes due to time of day, etc. =20
> > Instead, SIP provides a
> > 		>       > remarkably effective system for=20
> > selecting the best target of
> > 		>       > a call *at the moment the call is=20
> > initiated* -- the caller
> > 		>       > declares his preferences, and the=20
> > various intermediate
> > 		>       > elements combine the caller's desires=20
> > with the routing
> > 		>       > information that they possess to=20
> > forward the call closer to
> > 		>       > the preferred destination.
> > 		>       >
> > 		>       > As a developer, I've had to wrestle=20
> > with this.  In order to
> > 		>       > provide clean "Is extension 1234=20
> > busy?" information, our "BLF
> > 		>       > agent" maintains subscriptions to the=20
> > "reg" event package for
> > 		>       > the AOR "sip:1234@example.com=20
> > <mailto:sip%3A1234@example.com>=20
> > 	=09
> > 		> <mailto:sip%3A1234@example.com=20
> > <mailto:sip%253A1234@example.com> > ".  For each reported contact
> > 	=09
> > 		>       > for the AOR, it maintains a dialog=20
> > subscription to the
> > 		>       > contact.  Only by tracking the "reg"=20
> > events can it keep track
> > 		>       > of the UAs available for that AOR. =20
> > And even then, it is at
> > 		>       > the mercy of network failures.
> > 		>       >
> > 		>       > Dale
> > 		>       >=20
> > _______________________________________________
> > 		>       > 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
> > 		>
> > 		>
> > 		>
> > 		>=20
> >=20
> >=20
> > =

From peter.musgrave@magorcorp.com  Wed Apr  7 04:30:00 2010
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From: Peter Musgrave <peter.musgrave@magorcorp.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

So in the interests of good protocol design...

Should we recap the different use cases we want this behaviour for?

Some cases may fall into the 'just send an INVITE camp' and others
which may be probing for support of event packages or specific media
capabilities and sending an INVITE is not helpful.

I had at first liked the idea but of using reg-events for some of this
- but endpoints which disappear (and do not expire their registration)
will not be reported absent until the next refresh cycle - which makes
that approach not usable IMHO.

Use cases I can (personally) see wanting:
- basic fork to see IF there even are multiple UAs at the AOR (and get
their gruus so a specific UA can subsequently be called, subscribed to
etc.))
- gather event support and media capabilities of the UAs
- others?


Peter

On Wed, Apr 7, 2010 at 7:08 AM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
>
> Hi,
>
>>>The forking proxy can also have policies what it returns in the 3xx,
>>>depending on what the end-users actually represent.
>>>
>>>And, also with other mechanisms the returned data per end-user may
>>>depend on what type of end-point it represents.
>>>I don't think that is bound to a specific mechanism as such.
>>[JRE] Quite true. But Dale's point still holds - you might as well just s=
end the INVITE request and see what happens.
>
> Of course. But, as I said, that is really bad protocol design, and it may=
 trigger unwanted resource reservation actions etc in the network.
>
> And, there could even be cases where the INVITE doesn't fail, but you don=
't get the optimized user experience since the request may have looked diff=
erent should you have known what the receiver supports.
>
> Regards,
>
> Christer
>
>
>
>
>
>
>
>> > ________________________________
>> >
>> > =A0 =A0 From: sipcore-bounces@ietf.org
>> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Hans Erik van Elburg
>> > =A0 =A0 Sent: 7. huhtikuuta 2010 12:18
>> > =A0 =A0 To: Elwell, John
>> > =A0 =A0 Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
>> > =A0 =A0 Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>> >
>> >
>> > =A0 =A0 Of course you can implement all kind of policies, but
>> > that makes this AOR quite of a special user. I would argue
>> > that this is some kind of service AOR and not a "normal" user
>> > AOR. The service AOR representing a group of different and
>> > dynamically changing users...
>> >
>> > =A0 =A0 Demonstrating that this is not a "normal" user.
>> >
>> > =A0 =A0 I expect that in practice one would solve this
>> > differently... making the case rather artificial.
>> >
>> > =A0 =A0 /Hans Erik van Elburg
>> >
>> >
>> > =A0 =A0 On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John
>> > <john.elwell@siemens-enterprise.com> wrote:
>> >
>> >
>> >
>> >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > -----Original Message-----
>> > =A0 =A0 =A0 =A0 =A0 =A0 > From: Hans Erik van Elburg
>> > [mailto:ietf.hanserik@gmail.com]
>> > =A0 =A0 =A0 =A0 =A0 =A0 > Sent: 07 April 2010 08:24
>> > =A0 =A0 =A0 =A0 =A0 =A0 > To: Elwell, John
>> > =A0 =A0 =A0 =A0 =A0 =A0 > Cc: WORLEY, DALE R (DALE); Salvatore Loreto;
>> > sipcore@ietf.org
>> > =A0 =A0 =A0 =A0 =A0 =A0 > Subject: Re: [sipcore] SIP OPTIONS: Shall we
>> > work on this?
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > This is a special case. And the application
>> > you are talking
>> > =A0 =A0 =A0 =A0 =A0 =A0 > about is a B2BUA, right?
>> >
>> > =A0 =A0 =A0 =A0 =A0 =A0 [JRE] Not necessarily. A proxy can be scripted
>> > to cycle between multiple registered contacts. This doesn't
>> > make it a B2BUA.
>> >
>> > =A0 =A0 =A0 =A0 =A0 =A0 John
>> >
>> >
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > In this case the AOR addressed is an
>> > application address
>> > =A0 =A0 =A0 =A0 =A0 =A0 > without registered contacts. One can argue
>> > that the OPTIONS
>> > =A0 =A0 =A0 =A0 =A0 =A0 > request should return the general capabiliti=
es of the
>> > =A0 =A0 =A0 =A0 =A0 =A0 > application. No OPTIONS forking is happening
>> > in this case, no
>> > =A0 =A0 =A0 =A0 =A0 =A0 > registered contacts. No issue?
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > In reality what the B2BUA will do with the
>> > OPTION request
>> > =A0 =A0 =A0 =A0 =A0 =A0 > (forward or answer) depends on the fantasy o=
f
>> > the B2BUA developer.
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > I dont think this scenario is actually
>> > usefull in discussing
>> > =A0 =A0 =A0 =A0 =A0 =A0 > whether or not to write a BCP on OPTIONS
>> > handling. Or maybe
>> > =A0 =A0 =A0 =A0 =A0 =A0 > it is as implication of B2BUA behaviours can
>> > be taken into account.
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > /Hans Erik van Elburg
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John
>> > =A0 =A0 =A0 =A0 =A0 =A0 > <john.elwell@siemens-enterprise.com> wrote:
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 I agree with Dale. What happens =
to an
>> > OPTIONS request
>> > =A0 =A0 =A0 =A0 =A0 =A0 > will not necessarily be reflected when an
>> > INVITE request is
>> > =A0 =A0 =A0 =A0 =A0 =A0 > sent shortly afterwards. As a further
>> > examples, in contact
>> > =A0 =A0 =A0 =A0 =A0 =A0 > centres, it is usual to distribute calls cyc=
lically or
>> > =A0 =A0 =A0 =A0 =A0 =A0 > randomly between qualified and available
>> > agents. Sending an
>> > =A0 =A0 =A0 =A0 =A0 =A0 > INVITE request to a UA determined by
>> > responses to an OPTIONS
>> > =A0 =A0 =A0 =A0 =A0 =A0 > request would break that model - the issuer
>> > of the OPTIONS
>> > =A0 =A0 =A0 =A0 =A0 =A0 > and INVITE requests cannot be expected to
>> > obey the same
>> > =A0 =A0 =A0 =A0 =A0 =A0 > rotation rules as the domain proxy that rout=
es INVITE
>> > =A0 =A0 =A0 =A0 =A0 =A0 > requests. Issuing an INVITE request to the
>> > AOR is unlikely to
>> > =A0 =A0 =A0 =A0 =A0 =A0 > result in it being delivered to the UA that
>> > might be expected
>> > =A0 =A0 =A0 =A0 =A0 =A0 > from analysis of responses to an OPTIONS req=
uest.
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 I don't think we should work on =
trying
>> > to make OPTIONS
>> > =A0 =A0 =A0 =A0 =A0 =A0 > work for this purpose. If we really need to
>> > do anything with
>> > =A0 =A0 =A0 =A0 =A0 =A0 > OPTIONS, we should just issue warnings about=
 its
>> > =A0 =A0 =A0 =A0 =A0 =A0 > unsuitability for purposes other than a simp=
le ping.
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 John
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > -----Original Message-----
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > From: sipcore-bounces@ietf.org
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > [mailto:sipcore-bounces@ietf.o=
rg] On Behalf Of
>> > =A0 =A0 =A0 =A0 =A0 =A0 > WORLEY, DALE R (DALE)
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > Sent: 06 April 2010 21:01
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > To: Salvatore Loreto; sipcore@=
ietf.org
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > Subject: Re: [sipcore] SIP OPT=
IONS:
>> > Shall we work on this?
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > ______________________________=
__________
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > From: sipcore-bounces@ietf.org
>> > [sipcore-bounces@ietf.org] On
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > Behalf Of Salvatore Loreto
>> > [salvatore.loreto@ericsson.com]
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > Sent: Thursday, April 01, 2010=
 3:36 AM
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > A good mechanism to discovery =
UA
>> > capabilities is essential to
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > provide meaningful service:
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > So I am in favor of pursuing t=
he work,
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > in particular the possibility =
to use
>> > a single OPTIONS request
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > to retrieve the capabilities f=
rom ALL
>> > UAS(s) behind a
>> > =A0 =A0 =A0 =A0 =A0 =A0 > forking proxy.
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > ______________________________=
_________
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > I think it is a misconception =
to use
>> > UA discovery to provide
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > services. =A0The difficulty is=
 that the
>> > set of available UAs
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > for a given destination URI ch=
anges
>> > unpredictably -- elements
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > become inacessible (mobile pho=
ne out
>> > of range!), call routing
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > changes due to time of day, et=
c.
>> > Instead, SIP provides a
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > remarkably effective system fo=
r
>> > selecting the best target of
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > a call *at the moment the call=
 is
>> > initiated* -- the caller
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > declares his preferences, and =
the
>> > various intermediate
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > elements combine the caller's =
desires
>> > with the routing
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > information that they possess =
to
>> > forward the call closer to
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > the preferred destination.
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > As a developer, I've had to wr=
estle
>> > with this. =A0In order to
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > provide clean "Is extension 12=
34
>> > busy?" information, our "BLF
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > agent" maintains subscriptions=
 to the
>> > "reg" event package for
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > the AOR "sip:1234@example.com
>> > <mailto:sip%3A1234@example.com>
>> >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > <mailto:sip%3A1234@example.com
>> > <mailto:sip%253A1234@example.com> > ". =A0For each reported contact
>> >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > for the AOR, it maintains a di=
alog
>> > subscription to the
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > contact. =A0Only by tracking t=
he "reg"
>> > events can it keep track
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > of the UAs available for that =
AOR.
>> > And even then, it is at
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > the mercy of network failures.
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > Dale
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
>> > _______________________________________________
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > sipcore mailing list
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > sipcore@ietf.org
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > https://www.ietf.org/mailman/l=
istinfo/sipcore
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 ________________________________=
_______________
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 sipcore mailing list
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 sipcore@ietf.org
>> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 https://www.ietf.org/mailman/lis=
tinfo/sipcore
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> > =A0 =A0 =A0 =A0 =A0 =A0 >
>> >
>> >
>> >
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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Date: Wed, 7 Apr 2010 13:35:33 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

The use-case to gather capabilities of an UAS is already there - I assume t=
hat's the reason OPTIONS was added in the first place :)

What we are discussing is simply to get the capabilities in the forking cas=
e, when there are multiple UASs associated with the AOR.

So, I am simply trying to make an existing capability usable for the the fo=
rking case. Nothing more - nothing less :)

If we don't like OPTIONS, let's deprecate it... Having methods and function=
s, but at the same time saying that it doesn't work and we therefor shouldn=
't use it, is very confusing...

Regards,

Christer
=20

> -----Original Message-----
> From: Peter Musgrave [mailto:peter.musgrave@magorcorp.com]=20
> Sent: 7. huhtikuuta 2010 14:30
> To: Christer Holmberg
> Cc: Elwell, John; Hans Erik van Elburg; WORLEY, DALE R=20
> (DALE); sipcore@ietf.org
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
> Hi,
>=20
> So in the interests of good protocol design...
>=20
> Should we recap the different use cases we want this behaviour for?
>=20
> Some cases may fall into the 'just send an INVITE camp' and=20
> others which may be probing for support of event packages or=20
> specific media capabilities and sending an INVITE is not helpful.
>=20
> I had at first liked the idea but of using reg-events for some of this
> - but endpoints which disappear (and do not expire their=20
> registration) will not be reported absent until the next=20
> refresh cycle - which makes that approach not usable IMHO.
>=20
> Use cases I can (personally) see wanting:
> - basic fork to see IF there even are multiple UAs at the AOR=20
> (and get their gruus so a specific UA can subsequently be=20
> called, subscribed to
> etc.))
> - gather event support and media capabilities of the UAs
> - others?
>=20
>=20
> Peter
>=20
> On Wed, Apr 7, 2010 at 7:08 AM, Christer Holmberg=20
> <christer.holmberg@ericsson.com> wrote:
> >
> > Hi,
> >
> >>>The forking proxy can also have policies what it returns=20
> in the 3xx,=20
> >>>depending on what the end-users actually represent.
> >>>
> >>>And, also with other mechanisms the returned data per end-user may=20
> >>>depend on what type of end-point it represents.
> >>>I don't think that is bound to a specific mechanism as such.
> >>[JRE] Quite true. But Dale's point still holds - you might=20
> as well just send the INVITE request and see what happens.
> >
> > Of course. But, as I said, that is really bad protocol=20
> design, and it may trigger unwanted resource reservation=20
> actions etc in the network.
> >
> > And, there could even be cases where the INVITE doesn't=20
> fail, but you don't get the optimized user experience since=20
> the request may have looked different should you have known=20
> what the receiver supports.
> >
> > Regards,
> >
> > Christer
> >
> >
> >
> >
> >
> >
> >
> >> > ________________________________
> >> >
> >> > =A0 =A0 From: sipcore-bounces@ietf.org
> >> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Hans Erik=20
> van Elburg
> >> > =A0 =A0 Sent: 7. huhtikuuta 2010 12:18
> >> > =A0 =A0 To: Elwell, John
> >> > =A0 =A0 Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> >> > =A0 =A0 Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> >> >
> >> >
> >> > =A0 =A0 Of course you can implement all kind of policies, but that=20
> >> > makes this AOR quite of a special user. I would argue=20
> that this is=20
> >> > some kind of service AOR and not a "normal" user AOR.=20
> The service=20
> >> > AOR representing a group of different and dynamically changing=20
> >> > users...
> >> >
> >> > =A0 =A0 Demonstrating that this is not a "normal" user.
> >> >
> >> > =A0 =A0 I expect that in practice one would solve this=20
> differently...=20
> >> > making the case rather artificial.
> >> >
> >> > =A0 =A0 /Hans Erik van Elburg
> >> >
> >> >
> >> > =A0 =A0 On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John=20
> >> > <john.elwell@siemens-enterprise.com> wrote:
> >> >
> >> >
> >> >
> >> >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > -----Original Message-----
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > From: Hans Erik van Elburg=20
> >> > [mailto:ietf.hanserik@gmail.com]
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > Sent: 07 April 2010 08:24
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > To: Elwell, John
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > Cc: WORLEY, DALE R (DALE); Salvatore Loret=
o;=20
> >> > sipcore@ietf.org
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > Subject: Re: [sipcore] SIP OPTIONS: Shall=
=20
> we work on=20
> >> > this?
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > This is a special case. And the=20
> application you are=20
> >> > talking
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > about is a B2BUA, right?
> >> >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 [JRE] Not necessarily. A proxy can be=20
> scripted to cycle=20
> >> > between multiple registered contacts. This doesn't make=20
> it a B2BUA.
> >> >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 John
> >> >
> >> >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > In this case the AOR addressed is an appli=
cation=20
> >> > address
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > without registered contacts. One can argue=
=20
> that the=20
> >> > OPTIONS
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > request should return the general=20
> capabilities of the
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > application. No OPTIONS forking is=20
> happening in this=20
> >> > case, no
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > registered contacts. No issue?
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > In reality what the B2BUA will do with the=
 OPTION=20
> >> > request
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > (forward or answer) depends on the fantasy=
 of the=20
> >> > B2BUA developer.
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > I dont think this scenario is actually use=
full in=20
> >> > discussing
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > whether or not to write a BCP on OPTIONS=20
> handling. Or=20
> >> > maybe
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > it is as implication of B2BUA behaviours=20
> can be taken=20
> >> > into account.
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > /Hans Erik van Elburg
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > On Wed, Apr 7, 2010 at 8:31 AM, Elwell, Jo=
hn
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > <john.elwell@siemens-enterprise.com> wrote=
:
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 I agree with Dale. What happen=
s to=20
> an OPTIONS=20
> >> > request
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > will not necessarily be reflected when an =
INVITE=20
> >> > request is
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > sent shortly afterwards. As a further exam=
ples, in=20
> >> > contact
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > centres, it is usual to distribute calls=20
> cyclically=20
> >> > or
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > randomly between qualified and available a=
gents.=20
> >> > Sending an
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > INVITE request to a UA determined by=20
> responses to an=20
> >> > OPTIONS
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > request would break that model - the issue=
r of the=20
> >> > OPTIONS
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > and INVITE requests cannot be expected to =
obey the=20
> >> > same
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > rotation rules as the domain proxy that=20
> routes INVITE
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > requests. Issuing an INVITE request to the=
 AOR is=20
> >> > unlikely to
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > result in it being delivered to the UA=20
> that might be=20
> >> > expected
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > from analysis of responses to an OPTIONS r=
equest.
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 I don't think we should work o=
n=20
> trying to make=20
> >> > OPTIONS
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > work for this purpose. If we really need t=
o do=20
> >> > anything with
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > OPTIONS, we should just issue warnings abo=
ut its
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > unsuitability for purposes other than a=20
> simple ping.
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 John
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > -----Original Message-----
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > From: sipcore-bounces@ietf.o=
rg
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > [mailto:sipcore-bounces@ietf=
.org]=20
> On Behalf=20
> >> > Of
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > WORLEY, DALE R (DALE)
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > Sent: 06 April 2010 21:01
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > To: Salvatore Loreto; sipcor=
e@ietf.org
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > Subject: Re: [sipcore] SIP O=
PTIONS:
> >> > Shall we work on this?
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > ____________________________=
____________
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > From: sipcore-bounces@ietf.o=
rg=20
> >> > [sipcore-bounces@ietf.org] On
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > Behalf Of Salvatore Loreto=20
> >> > [salvatore.loreto@ericsson.com]
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > Sent: Thursday, April 01, 20=
10 3:36 AM
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > A good mechanism to discover=
y UA=20
> capabilities=20
> >> > is essential to
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > provide meaningful service:
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > So I am in favor of pursuing=
 the work,
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > in particular the possibilit=
y to=20
> use a single=20
> >> > OPTIONS request
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > to retrieve the capabilities=
 from ALL
> >> > UAS(s) behind a
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > forking proxy.
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > ____________________________=
___________
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > I think it is a misconceptio=
n to use UA=20
> >> > discovery to provide
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > services. =A0The difficulty =
is that=20
> the set of=20
> >> > available UAs
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > for a given destination URI =
changes=20
> >> > unpredictably -- elements
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > become inacessible (mobile p=
hone out of=20
> >> > range!), call routing
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > changes due to time of day, =
etc.
> >> > Instead, SIP provides a
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > remarkably effective system =
for=20
> selecting the=20
> >> > best target of
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > a call *at the moment the ca=
ll is
> >> > initiated* -- the caller
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > declares his preferences, an=
d the various=20
> >> > intermediate
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > elements combine the caller'=
s desires with=20
> >> > the routing
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > information that they posses=
s to=20
> forward the=20
> >> > call closer to
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > the preferred destination.
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > As a developer, I've had to =
wrestle with=20
> >> > this. =A0In order to
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > provide clean "Is extension =
1234 busy?"=20
> >> > information, our "BLF
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > agent" maintains subscriptio=
ns to=20
> the "reg"=20
> >> > event package for
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > the AOR "sip:1234@example.co=
m=20
> >> > <mailto:sip%3A1234@example.com>
> >> >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > <mailto:sip%3A1234@example.com=20
> >> > <mailto:sip%253A1234@example.com> > ". =A0For each reported contact
> >> >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > for the AOR, it maintains a =
dialog=20
> >> > subscription to the
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > contact. =A0Only by tracking=
 the "reg"
> >> > events can it keep track
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > of the UAs available for tha=
t AOR.
> >> > And even then, it is at
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > the mercy of network failure=
s.
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > Dale
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
> >> > _______________________________________________
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > sipcore mailing list
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 > sipcore@ietf.org
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >=20
> https://www.ietf.org/mailman/listinfo/sipcore
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0=20
> _______________________________________________
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 sipcore mailing list
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 sipcore@ietf.org
> >> > =A0 =A0 =A0 =A0 =A0 =A0 > =A0 =A0 =A0 https://www.ietf.org/mailman/l=
istinfo/sipcore
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> > =A0 =A0 =A0 =A0 =A0 =A0 >
> >> >
> >> >
> >> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
> =

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From: Brett Tate <brett@broadsoft.com>
To: Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 7 Apr 2010 05:08:16 -0700
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Tuesday, April 06, 2010 7:37 PM
> To: Brett Tate
> Cc: Paul Kyzivat; SIPCORE
> Subject: Re: [sipcore] rfc3265bis: SIP events redux [was Minutes
> Posted]
>=20
> On 4/6/10 17:46, Apr 6, Brett Tate wrote:
> >> Is there *any* problem here? This seems to be working about
> >> as well as anyone might hope.
> >>
> > It works okay excluding the non backward compatibility with RFC 3265
> (concerning SUBSCRIBE's 2xx response impacts)...
>=20
> I don't think it's accurate to characterize the change as
> non-backwards-compatible.

RFC 3265 section 3.3.4 indicates that 2xx "creates a new subscription and a=
 new dialog"; rfc3265bis does not allow such a thing.

3.3.4. Dialog creation and termination

   If an initial SUBSCRIBE request is not sent on a pre-existing dialog,
   the subscriber will wait for a response to the SUBSCRIBE request or a
   matching NOTIFY.

   Responses are matched to such SUBSCRIBE requests if they contain the
   same the same "Call-ID", the same "From" header "tag", and the same
   "CSeq".  Rules for the comparison of these headers are described in
   SIP [1].  If a 200-class response matches such a SUBSCRIBE request,
   it creates a new subscription and a new dialog (unless they have
   already been created by a matching NOTIFY request; see below).

<snip>

> > The draft's Timer N (and lack of related NOTIFY) text is potentially
> > worded in a way which does not require the tear down (and non
> > creation) of subscription.
>=20
> Tear-down of existing dialogs is still an open issue, sure. I don't get
> where you see any wiggle room in what happens if Timer N fires for the
> first SUBSCRIBE/NOTIFY exchange.

That is the issue... as currently worded, there is no "wiggle room".  Thus =
a vendor that wants to implement rfc3265bis can't also remain compatible wi=
th rfc3265 devices.  And if a vendor wants to use a value other than 64*T1 =
for Timer N, rfc3265bis doesn't appear to allow it.

Timer N has its use.  Depending upon the dialog package, some vendors imple=
ment Timer N to be extremely low (such as 2-5 seconds).  For other packages=
, they apply Timer N (with value similar to rfc3265bis) only to dialogs bei=
ng created by NOTIFY; they allow the SUBSCRIBE 2xx's Expires to be Timer N =
value for the associated dialog.

I actually don't have strong opinion concerning all of this.  I raised the =
issue mainly because the divergence from rfc3265 appears to work worse duri=
ng overload situations and because frequent subscription creation/removal (=
such as because Timer N expiration) can lead to or prolong overload.=20


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From: Brett Tate <brett@broadsoft.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 7 Apr 2010 05:26:39 -0700
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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> Then what would happen in a polling subscribe (Expires:0) if=20
> the 200 is received and the NOTIFY is not received? How long=20
> should you wait before giving up? ISTM you still need timer-N=20
> to tell you it failed, so you can give up waiting.

I agree that you need a Timer N (or something like it); however Timer N sho=
uld not be restricted to 64*T1.  The rfc3265bis should accommodate other Ti=
mer N values instead being restricted to 64*T1 for all event packages, depl=
oyments, situations, etcetera.

Concerning the polling/fetch (Expires: 0), it might actually be useful to c=
ommunicate within SUBSCRIBE how long you are willing to wait for the NOTIFY=
.  For instance, there are some situation where I'm willing to wait 1-2 sec=
onds; there are other situations where I'm will wait more than 64*T1 if rec=
eived 2xx response.


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To: Christer Holmberg <christer.holmberg@ericsson.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Wed, 7 Apr 2010 14:56:38 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 07 April 2010 12:09
> To: Elwell, John; Hans Erik van Elburg
> Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
>=20
> Hi,=20
>=20
> >>The forking proxy can also have policies what it returns in=20
> the 3xx,=20
> >>depending on what the end-users actually represent.
> >> =20
> >>And, also with other mechanisms the returned data per end-user may=20
> >>depend on what type of end-point it represents.
> >>I don't think that is bound to a specific mechanism as such.
> >[JRE] Quite true. But Dale's point still holds - you might=20
> as well just send the INVITE request and see what happens.
>=20
> Of course. But, as I said, that is really bad protocol=20
> design,
[JRE] Why is it bad protocol design?

> and it may trigger unwanted resource reservation=20
> actions etc in the network.
[JRE] Concerning resource reservation, there are two ways of looking at it.=
 By doing resource reservation, you find out straight away if the resources=
 needed to reach the UA are not available, so resource reservation could be=
 considered a good thing. An OPTIONS transaction is never going to provide =
this information. On the other hand, if you want to avoid resource reservat=
ion, just send an offerless INVITE request.

John


>=20
> And, there could even be cases where the INVITE doesn't fail,=20
> but you don't get the optimized user experience since the=20
> request may have looked different should you have known what=20
> the receiver supports.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> > > ________________________________
> > >=20
> > > 	From: sipcore-bounces@ietf.org=20
> > > [mailto:sipcore-bounces@ietf.org] On Behalf Of Hans Erik=20
> van Elburg
> > > 	Sent: 7. huhtikuuta 2010 12:18
> > > 	To: Elwell, John
> > > 	Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> > > 	Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> > > =09
> > > =09
> > > 	Of course you can implement all kind of policies, but=20
> > > that makes this AOR quite of a special user. I would argue=20
> > > that this is some kind of service AOR and not a "normal" user=20
> > > AOR. The service AOR representing a group of different and=20
> > > dynamically changing users...
> > > =09
> > > 	Demonstrating that this is not a "normal" user.
> > > =09
> > > 	I expect that in practice one would solve this=20
> > > differently... making the case rather artificial.
> > > =09
> > > 	/Hans Erik van Elburg
> > > =09
> > > =09
> > > 	On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John=20
> > > <john.elwell@siemens-enterprise.com> wrote:
> > > =09
> > >=20
> > >=20
> > >=20
> > > 		> -----Original Message-----
> > > 		> From: Hans Erik van Elburg=20
> > > [mailto:ietf.hanserik@gmail.com]
> > > 		> Sent: 07 April 2010 08:24
> > > 		> To: Elwell, John
> > > 		> Cc: WORLEY, DALE R (DALE); Salvatore Loreto;=20
> > > sipcore@ietf.org
> > > 		> Subject: Re: [sipcore] SIP OPTIONS: Shall we=20
> > > work on this?
> > > 		>
> > > 	=09
> > > 		> This is a special case. And the application=20
> > > you are talking
> > > 		> about is a B2BUA, right?
> > > 	=09
> > > 		[JRE] Not necessarily. A proxy can be scripted=20
> > > to cycle between multiple registered contacts. This doesn't=20
> > > make it a B2BUA.
> > > 	=09
> > > 		John
> > > 	=09
> > >=20
> > > 		>
> > > 		> In this case the AOR addressed is an=20
> > > application address
> > > 		> without registered contacts. One can argue=20
> > > that the OPTIONS
> > > 		> request should return the general capabilities of the
> > > 		> application. No OPTIONS forking is happening=20
> > > in this case, no
> > > 		> registered contacts. No issue?
> > > 		>
> > > 		> In reality what the B2BUA will do with the=20
> > > OPTION request
> > > 		> (forward or answer) depends on the fantasy of=20
> > > the B2BUA developer.
> > > 		>
> > > 		> I dont think this scenario is actually=20
> > > usefull in discussing
> > > 		> whether or not to write a BCP on OPTIONS=20
> > > handling. Or maybe
> > > 		> it is as implication of B2BUA behaviours can=20
> > > be taken into account.
> > > 		>
> > > 		> /Hans Erik van Elburg
> > > 		>
> > > 		>
> > > 		>
> > > 		> On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John
> > > 		> <john.elwell@siemens-enterprise.com> wrote:
> > > 		>
> > > 		>
> > > 		>       I agree with Dale. What happens to an=20
> > > OPTIONS request
> > > 		> will not necessarily be reflected when an=20
> > > INVITE request is
> > > 		> sent shortly afterwards. As a further=20
> > > examples, in contact
> > > 		> centres, it is usual to distribute calls cyclically or
> > > 		> randomly between qualified and available=20
> > > agents. Sending an
> > > 		> INVITE request to a UA determined by=20
> > > responses to an OPTIONS
> > > 		> request would break that model - the issuer=20
> > > of the OPTIONS
> > > 		> and INVITE requests cannot be expected to=20
> > > obey the same
> > > 		> rotation rules as the domain proxy that routes INVITE
> > > 		> requests. Issuing an INVITE request to the=20
> > > AOR is unlikely to
> > > 		> result in it being delivered to the UA that=20
> > > might be expected
> > > 		> from analysis of responses to an OPTIONS request.
> > > 		>
> > > 		>       I don't think we should work on trying=20
> > > to make OPTIONS
> > > 		> work for this purpose. If we really need to=20
> > > do anything with
> > > 		> OPTIONS, we should just issue warnings about its
> > > 		> unsuitability for purposes other than a simple ping.
> > > 		>
> > > 		>       John
> > > 		>
> > > 		>
> > > 		>
> > > 		>       > -----Original Message-----
> > > 		>       > From: sipcore-bounces@ietf.org
> > > 		>
> > > 		>       > [mailto:sipcore-bounces@ietf.org] On Behalf Of
> > > 		> WORLEY, DALE R (DALE)
> > > 		>       > Sent: 06 April 2010 21:01
> > > 		>       > To: Salvatore Loreto; sipcore@ietf.org
> > > 		>       > Subject: Re: [sipcore] SIP OPTIONS:=20
> > > Shall we work on this?
> > > 		>       >
> > > 		>       > ________________________________________
> > > 		>
> > > 		>       > From: sipcore-bounces@ietf.org=20
> > > [sipcore-bounces@ietf.org] On
> > > 		>       > Behalf Of Salvatore Loreto=20
> > > [salvatore.loreto@ericsson.com]
> > > 		>       > Sent: Thursday, April 01, 2010 3:36 AM
> > > 		>       >
> > > 		>       > A good mechanism to discovery UA=20
> > > capabilities is essential to
> > > 		>       > provide meaningful service:
> > > 		>       > So I am in favor of pursuing the work,
> > > 		>       > in particular the possibility to use=20
> > > a single OPTIONS request
> > > 		>       > to retrieve the capabilities from ALL=20
> > > UAS(s) behind a
> > > 		> forking proxy.
> > > 		>       > _______________________________________
> > > 		>       >
> > > 		>       > I think it is a misconception to use=20
> > > UA discovery to provide
> > > 		>       > services.  The difficulty is that the=20
> > > set of available UAs
> > > 		>       > for a given destination URI changes=20
> > > unpredictably -- elements
> > > 		>       > become inacessible (mobile phone out=20
> > > of range!), call routing
> > > 		>       > changes due to time of day, etc. =20
> > > Instead, SIP provides a
> > > 		>       > remarkably effective system for=20
> > > selecting the best target of
> > > 		>       > a call *at the moment the call is=20
> > > initiated* -- the caller
> > > 		>       > declares his preferences, and the=20
> > > various intermediate
> > > 		>       > elements combine the caller's desires=20
> > > with the routing
> > > 		>       > information that they possess to=20
> > > forward the call closer to
> > > 		>       > the preferred destination.
> > > 		>       >
> > > 		>       > As a developer, I've had to wrestle=20
> > > with this.  In order to
> > > 		>       > provide clean "Is extension 1234=20
> > > busy?" information, our "BLF
> > > 		>       > agent" maintains subscriptions to the=20
> > > "reg" event package for
> > > 		>       > the AOR "sip:1234@example.com=20
> > > <mailto:sip%3A1234@example.com>=20
> > > 	=09
> > > 		> <mailto:sip%3A1234@example.com=20
> > > <mailto:sip%253A1234@example.com> > ".  For each reported contact
> > > 	=09
> > > 		>       > for the AOR, it maintains a dialog=20
> > > subscription to the
> > > 		>       > contact.  Only by tracking the "reg"=20
> > > events can it keep track
> > > 		>       > of the UAs available for that AOR. =20
> > > And even then, it is at
> > > 		>       > the mercy of network failures.
> > > 		>       >
> > > 		>       > Dale
> > > 		>       >=20
> > > _______________________________________________
> > > 		>       > 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
> > > 		>
> > > 		>
> > > 		>
> > > 		>=20
> > >=20
> > >=20
> > > =

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Hi,

When looking deeper into draft-ietf-sipcore-subnot-etags, we also found the=
 following issue/question:

Figure1/Step 3 and Figure 3/Step 3 suggest that the subnot-etags compliant =
notifier includes the SIP-Etag in the NOT request even if the associated SU=
B request did not contain Suppress-If-Match. The overview text in section 3=
 also seems to indicate that SIP-Etag is inserted in every NOT, but that is=
 also informative text only.

However, Section 6 only contains normative text about inclusion of the SIP-=
Etag in the following situations:

1. The NOT request is generated upon SIP SUB with condition (section 6.2); =
and

2. The NOT request is generated with state differential (section 6.4)

So, the question is: is a subnot-etags compliant notifier required to inser=
t the SIP-Etag in *every* NOT request, or only in the cases explicitly desc=
ribed in Section 6?

Regards,

Christer



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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
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<font face=3D"Arial, sans-serif" size=3D"3">
<div>&nbsp;</div>
<div><font size=3D"2">Hi,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">When looking deeper into draft-ietf-sipcore-subnot-et=
ags, we also found the following issue/question:</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Figure1/Step 3 and Figure 3/Step 3 suggest that the s=
ubnot-etags compliant notifier includes the SIP-Etag in the NOT request eve=
n if the associated SUB request did not contain Suppress-If-Match. The over=
view text in section 3 also seems
to indicate that SIP-Etag is inserted in every NOT, but that is also inform=
ative text only. </font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">However, Section 6 only contains normative text about=
 inclusion of the SIP-Etag in the following situations:</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">1. The NOT request is generated upon SIP SUB with con=
dition (section 6.2); and</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">2. The NOT request is generated with state differenti=
al (section 6.4)</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">So, the question is: is a subnot-etags compliant noti=
fier required to insert the SIP-Etag in *every* NOT request, or only in the=
 cases explicitly described in Section 6?</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Regards,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Christer</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
</font>
</body>
</html>

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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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John,

You are illustrating the case that bad design leads to: You need to be aware
of the side effects of one method, so that you can circumvent these to use
it for other purposes then it was intended for.

/Hans Erik van Elburg


On Wed, Apr 7, 2010 at 2:56 PM, Elwell, John <
john.elwell@siemens-enterprise.com> wrote:

>
>
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: 07 April 2010 12:09
> > To: Elwell, John; Hans Erik van Elburg
> > Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> > Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?
> >
> >
> > Hi,
> >
> > >>The forking proxy can also have policies what it returns in
> > the 3xx,
> > >>depending on what the end-users actually represent.
> > >>
> > >>And, also with other mechanisms the returned data per end-user may
> > >>depend on what type of end-point it represents.
> > >>I don't think that is bound to a specific mechanism as such.
> > >[JRE] Quite true. But Dale's point still holds - you might
> > as well just send the INVITE request and see what happens.
> >
> > Of course. But, as I said, that is really bad protocol
> > design,
> [JRE] Why is it bad protocol design?
>
> > and it may trigger unwanted resource reservation
> > actions etc in the network.
> [JRE] Concerning resource reservation, there are two ways of looking at it.
> By doing resource reservation, you find out straight away if the resources
> needed to reach the UA are not available, so resource reservation could be
> considered a good thing. An OPTIONS transaction is never going to provide
> this information. On the other hand, if you want to avoid resource
> reservation, just send an offerless INVITE request.
>
> John
>
>
> >
> > And, there could even be cases where the INVITE doesn't fail,
> > but you don't get the optimized user experience since the
> > request may have looked different should you have known what
> > the receiver supports.
> >
> > Regards,
> >
> > Christer
> >
> >
> >
> >
> >
> >
> >
> > > > ________________________________
> > > >
> > > >   From: sipcore-bounces@ietf.org
> > > > [mailto:sipcore-bounces@ietf.org] On Behalf Of Hans Erik
> > van Elburg
> > > >   Sent: 7. huhtikuuta 2010 12:18
> > > >   To: Elwell, John
> > > >   Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> > > >   Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> > > >
> > > >
> > > >   Of course you can implement all kind of policies, but
> > > > that makes this AOR quite of a special user. I would argue
> > > > that this is some kind of service AOR and not a "normal" user
> > > > AOR. The service AOR representing a group of different and
> > > > dynamically changing users...
> > > >
> > > >   Demonstrating that this is not a "normal" user.
> > > >
> > > >   I expect that in practice one would solve this
> > > > differently... making the case rather artificial.
> > > >
> > > >   /Hans Erik van Elburg
> > > >
> > > >
> > > >   On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John
> > > > <john.elwell@siemens-enterprise.com> wrote:
> > > >
> > > >
> > > >
> > > >
> > > >           > -----Original Message-----
> > > >           > From: Hans Erik van Elburg
> > > > [mailto:ietf.hanserik@gmail.com]
> > > >           > Sent: 07 April 2010 08:24
> > > >           > To: Elwell, John
> > > >           > Cc: WORLEY, DALE R (DALE); Salvatore Loreto;
> > > > sipcore@ietf.org
> > > >           > Subject: Re: [sipcore] SIP OPTIONS: Shall we
> > > > work on this?
> > > >           >
> > > >
> > > >           > This is a special case. And the application
> > > > you are talking
> > > >           > about is a B2BUA, right?
> > > >
> > > >           [JRE] Not necessarily. A proxy can be scripted
> > > > to cycle between multiple registered contacts. This doesn't
> > > > make it a B2BUA.
> > > >
> > > >           John
> > > >
> > > >
> > > >           >
> > > >           > In this case the AOR addressed is an
> > > > application address
> > > >           > without registered contacts. One can argue
> > > > that the OPTIONS
> > > >           > request should return the general capabilities of the
> > > >           > application. No OPTIONS forking is happening
> > > > in this case, no
> > > >           > registered contacts. No issue?
> > > >           >
> > > >           > In reality what the B2BUA will do with the
> > > > OPTION request
> > > >           > (forward or answer) depends on the fantasy of
> > > > the B2BUA developer.
> > > >           >
> > > >           > I dont think this scenario is actually
> > > > usefull in discussing
> > > >           > whether or not to write a BCP on OPTIONS
> > > > handling. Or maybe
> > > >           > it is as implication of B2BUA behaviours can
> > > > be taken into account.
> > > >           >
> > > >           > /Hans Erik van Elburg
> > > >           >
> > > >           >
> > > >           >
> > > >           > On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John
> > > >           > <john.elwell@siemens-enterprise.com> wrote:
> > > >           >
> > > >           >
> > > >           >       I agree with Dale. What happens to an
> > > > OPTIONS request
> > > >           > will not necessarily be reflected when an
> > > > INVITE request is
> > > >           > sent shortly afterwards. As a further
> > > > examples, in contact
> > > >           > centres, it is usual to distribute calls cyclically or
> > > >           > randomly between qualified and available
> > > > agents. Sending an
> > > >           > INVITE request to a UA determined by
> > > > responses to an OPTIONS
> > > >           > request would break that model - the issuer
> > > > of the OPTIONS
> > > >           > and INVITE requests cannot be expected to
> > > > obey the same
> > > >           > rotation rules as the domain proxy that routes INVITE
> > > >           > requests. Issuing an INVITE request to the
> > > > AOR is unlikely to
> > > >           > result in it being delivered to the UA that
> > > > might be expected
> > > >           > from analysis of responses to an OPTIONS request.
> > > >           >
> > > >           >       I don't think we should work on trying
> > > > to make OPTIONS
> > > >           > work for this purpose. If we really need to
> > > > do anything with
> > > >           > OPTIONS, we should just issue warnings about its
> > > >           > unsuitability for purposes other than a simple ping.
> > > >           >
> > > >           >       John
> > > >           >
> > > >           >
> > > >           >
> > > >           >       > -----Original Message-----
> > > >           >       > From: sipcore-bounces@ietf.org
> > > >           >
> > > >           >       > [mailto:sipcore-bounces@ietf.org] On Behalf Of
> > > >           > WORLEY, DALE R (DALE)
> > > >           >       > Sent: 06 April 2010 21:01
> > > >           >       > To: Salvatore Loreto; sipcore@ietf.org
> > > >           >       > Subject: Re: [sipcore] SIP OPTIONS:
> > > > Shall we work on this?
> > > >           >       >
> > > >           >       > ________________________________________
> > > >           >
> > > >           >       > From: sipcore-bounces@ietf.org
> > > > [sipcore-bounces@ietf.org] On
> > > >           >       > Behalf Of Salvatore Loreto
> > > > [salvatore.loreto@ericsson.com]
> > > >           >       > Sent: Thursday, April 01, 2010 3:36 AM
> > > >           >       >
> > > >           >       > A good mechanism to discovery UA
> > > > capabilities is essential to
> > > >           >       > provide meaningful service:
> > > >           >       > So I am in favor of pursuing the work,
> > > >           >       > in particular the possibility to use
> > > > a single OPTIONS request
> > > >           >       > to retrieve the capabilities from ALL
> > > > UAS(s) behind a
> > > >           > forking proxy.
> > > >           >       > _______________________________________
> > > >           >       >
> > > >           >       > I think it is a misconception to use
> > > > UA discovery to provide
> > > >           >       > services.  The difficulty is that the
> > > > set of available UAs
> > > >           >       > for a given destination URI changes
> > > > unpredictably -- elements
> > > >           >       > become inacessible (mobile phone out
> > > > of range!), call routing
> > > >           >       > changes due to time of day, etc.
> > > > Instead, SIP provides a
> > > >           >       > remarkably effective system for
> > > > selecting the best target of
> > > >           >       > a call *at the moment the call is
> > > > initiated* -- the caller
> > > >           >       > declares his preferences, and the
> > > > various intermediate
> > > >           >       > elements combine the caller's desires
> > > > with the routing
> > > >           >       > information that they possess to
> > > > forward the call closer to
> > > >           >       > the preferred destination.
> > > >           >       >
> > > >           >       > As a developer, I've had to wrestle
> > > > with this.  In order to
> > > >           >       > provide clean "Is extension 1234
> > > > busy?" information, our "BLF
> > > >           >       > agent" maintains subscriptions to the
> > > > "reg" event package for
> > > >           >       > the AOR "sip:1234@example.com<sip%3A1234@example.com>
> > > > <mailto:sip%3A1234@example.com <sip%253A1234@example.com>>
> > > >
> > > >           > <mailto:sip%3A1234@example.com<sip%253A1234@example.com>
> > > > <mailto:sip%253A1234@example.com <sip%25253A1234@example.com>> > ".
>  For each reported contact
> > > >
> > > >           >       > for the AOR, it maintains a dialog
> > > > subscription to the
> > > >           >       > contact.  Only by tracking the "reg"
> > > > events can it keep track
> > > >           >       > of the UAs available for that AOR.
> > > > And even then, it is at
> > > >           >       > the mercy of network failures.
> > > >           >       >
> > > >           >       > Dale
> > > >           >       >
> > > > _______________________________________________
> > > >           >       > 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
> > > >           >
> > > >           >
> > > >           >
> > > >           >
> > > >
> > > >
> > > >
>

--001485326135607ee60483a53742
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

John,<br><br>You are illustrating the case that bad design leads to: You ne=
ed to be aware of the side effects of one method, so that you can circumven=
t these to use it for other purposes then it was intended for.<br><br clear=
=3D"all">
/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Wed, Apr 7, 2010 at 2:56 PM, 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"margin: 0pt 0pt 0pt 0.8ex; border-left: 1=
px solid rgb(204, 204, 204); padding-left: 1ex;">
<div class=3D"im"><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: Christer Holmberg [mailto:<a href=3D"mailto:christer.holmberg@er=
icsson.com">christer.holmberg@ericsson.com</a>]<br>
&gt; Sent: 07 April 2010 12:09<br>
&gt; To: Elwell, John; Hans Erik van Elburg<br>
&gt; Cc: WORLEY, DALE R (DALE); <a href=3D"mailto:sipcore@ietf.org">sipcore=
@ietf.org</a><br>
</div><div class=3D"im">&gt; Subject: RE: [sipcore] SIP OPTIONS: Shall we w=
ork on this?<br>
&gt;<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; &gt;&gt;The forking proxy can also have policies what it returns in<br=
>
&gt; the 3xx,<br>
&gt; &gt;&gt;depending on what the end-users actually represent.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;And, also with other mechanisms the returned data per end-user=
 may<br>
&gt; &gt;&gt;depend on what type of end-point it represents.<br>
&gt; &gt;&gt;I don&#39;t think that is bound to a specific mechanism as suc=
h.<br>
&gt; &gt;[JRE] Quite true. But Dale&#39;s point still holds - you might<br>
&gt; as well just send the INVITE request and see what happens.<br>
&gt;<br>
&gt; Of course. But, as I said, that is really bad protocol<br>
&gt; design,<br>
</div>[JRE] Why is it bad protocol design?<br>
<div class=3D"im"><br>
&gt; and it may trigger unwanted resource reservation<br>
&gt; actions etc in the network.<br>
</div>[JRE] Concerning resource reservation, there are two ways of looking =
at it. By doing resource reservation, you find out straight away if the res=
ources needed to reach the UA are not available, so resource reservation co=
uld be considered a good thing. An OPTIONS transaction is never going to pr=
ovide this information. On the other hand, if you want to avoid resource re=
servation, just send an offerless INVITE request.<br>

<font color=3D"#888888"><br>
John<br>
</font><div><div></div><div class=3D"h5"><br>
<br>
&gt;<br>
&gt; And, there could even be cases where the INVITE doesn&#39;t fail,<br>
&gt; but you don&#39;t get the optimized user experience since the<br>
&gt; request may have looked different should you have known what<br>
&gt; the receiver supports.<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; &gt; &gt; ________________________________<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 From: <a href=3D"mailto:sipcore-bounces@ietf.org">sipcor=
e-bounces@ietf.org</a><br>
&gt; &gt; &gt; [mailto:<a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-=
bounces@ietf.org</a>] On Behalf Of Hans Erik<br>
&gt; van Elburg<br>
&gt; &gt; &gt; =A0 Sent: 7. huhtikuuta 2010 12:18<br>
&gt; &gt; &gt; =A0 To: Elwell, John<br>
&gt; &gt; &gt; =A0 Cc: WORLEY, DALE R (DALE); <a href=3D"mailto:sipcore@iet=
f.org">sipcore@ietf.org</a><br>
&gt; &gt; &gt; =A0 Subject: Re: [sipcore] SIP OPTIONS: Shall we work on thi=
s?<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 Of course you can implement all kind of policies, but<br=
>
&gt; &gt; &gt; that makes this AOR quite of a special user. I would argue<b=
r>
&gt; &gt; &gt; that this is some kind of service AOR and not a &quot;normal=
&quot; user<br>
&gt; &gt; &gt; AOR. The service AOR representing a group of different and<b=
r>
&gt; &gt; &gt; dynamically changing users...<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 Demonstrating that this is not a &quot;normal&quot; user=
.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 I expect that in practice one would solve this<br>
&gt; &gt; &gt; differently... making the case rather artificial.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 /Hans Erik van Elburg<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John<br>
&gt; &gt; &gt; &lt;<a href=3D"mailto:john.elwell@siemens-enterprise.com">jo=
hn.elwell@siemens-enterprise.com</a>&gt; wrote:<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; -----Original Message-----<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; From: Hans Erik van Elburg<br>
&gt; &gt; &gt; [mailto:<a href=3D"mailto:ietf.hanserik@gmail.com">ietf.hans=
erik@gmail.com</a>]<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; Sent: 07 April 2010 08:24<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; To: Elwell, John<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; Cc: WORLEY, DALE R (DALE); Salvator=
e Loreto;<br>
&gt; &gt; &gt; <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; Subject: Re: [sipcore] SIP OPTIONS:=
 Shall we<br>
&gt; &gt; &gt; work on this?<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; This is a special case. And the app=
lication<br>
&gt; &gt; &gt; you are talking<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; about is a B2BUA, right?<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 [JRE] Not necessarily. A proxy can be sc=
ripted<br>
&gt; &gt; &gt; to cycle between multiple registered contacts. This doesn&#3=
9;t<br>
&gt; &gt; &gt; make it a B2BUA.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 John<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; In this case the AOR addressed is a=
n<br>
&gt; &gt; &gt; application address<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; without registered contacts. One ca=
n argue<br>
&gt; &gt; &gt; that the OPTIONS<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; request should return the general c=
apabilities of the<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; application. No OPTIONS forking is =
happening<br>
&gt; &gt; &gt; in this case, no<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; registered contacts. No issue?<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; In reality what the B2BUA will do w=
ith the<br>
&gt; &gt; &gt; OPTION request<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; (forward or answer) depends on the =
fantasy of<br>
&gt; &gt; &gt; the B2BUA developer.<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; I dont think this scenario is actua=
lly<br>
&gt; &gt; &gt; usefull in discussing<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; whether or not to write a BCP on OP=
TIONS<br>
&gt; &gt; &gt; handling. Or maybe<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; it is as implication of B2BUA behav=
iours can<br>
&gt; &gt; &gt; be taken into account.<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; /Hans Erik van Elburg<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; On Wed, Apr 7, 2010 at 8:31 AM, Elw=
ell, John<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; &lt;<a href=3D"mailto:john.elwell@s=
iemens-enterprise.com">john.elwell@siemens-enterprise.com</a>&gt; wrote:<br=
>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 I agree with Dale. What=
 happens to an<br>
&gt; &gt; &gt; OPTIONS request<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; will not necessarily be reflected w=
hen an<br>
&gt; &gt; &gt; INVITE request is<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; sent shortly afterwards. As a furth=
er<br>
&gt; &gt; &gt; examples, in contact<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; centres, it is usual to distribute =
calls cyclically or<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; randomly between qualified and avai=
lable<br>
&gt; &gt; &gt; agents. Sending an<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; INVITE request to a UA determined b=
y<br>
&gt; &gt; &gt; responses to an OPTIONS<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; request would break that model - th=
e issuer<br>
&gt; &gt; &gt; of the OPTIONS<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; and INVITE requests cannot be expec=
ted to<br>
&gt; &gt; &gt; obey the same<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; rotation rules as the domain proxy =
that routes INVITE<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; requests. Issuing an INVITE request=
 to the<br>
&gt; &gt; &gt; AOR is unlikely to<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; result in it being delivered to the=
 UA that<br>
&gt; &gt; &gt; might be expected<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; from analysis of responses to an OP=
TIONS request.<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 I don&#39;t think we sh=
ould work on trying<br>
&gt; &gt; &gt; to make OPTIONS<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; work for this purpose. If we really=
 need to<br>
&gt; &gt; &gt; do anything with<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; OPTIONS, we should just issue warni=
ngs about its<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; unsuitability for purposes other th=
an a simple ping.<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 John<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; -----Original Mess=
age-----<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; From: <a href=3D"m=
ailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a><br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; [mailto:<a href=3D=
"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a>] On Behalf O=
f<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; WORLEY, DALE R (DALE)<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; Sent: 06 April 201=
0 21:01<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; To: Salvatore Lore=
to; <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; Subject: Re: [sipc=
ore] SIP OPTIONS:<br>
&gt; &gt; &gt; Shall we work on this?<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; __________________=
______________________<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; From: <a href=3D"m=
ailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a><br>
&gt; &gt; &gt; [<a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces=
@ietf.org</a>] On<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; Behalf Of Salvator=
e Loreto<br>
&gt; &gt; &gt; [<a href=3D"mailto:salvatore.loreto@ericsson.com">salvatore.=
loreto@ericsson.com</a>]<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; Sent: Thursday, Ap=
ril 01, 2010 3:36 AM<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; A good mechanism t=
o discovery UA<br>
&gt; &gt; &gt; capabilities is essential to<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; provide meaningful=
 service:<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; So I am in favor o=
f pursuing the work,<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; in particular the =
possibility to use<br>
&gt; &gt; &gt; a single OPTIONS request<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; to retrieve the ca=
pabilities from ALL<br>
&gt; &gt; &gt; UAS(s) behind a<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; forking proxy.<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; __________________=
_____________________<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; I think it is a mi=
sconception to use<br>
&gt; &gt; &gt; UA discovery to provide<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; services. =A0The d=
ifficulty is that the<br>
&gt; &gt; &gt; set of available UAs<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; for a given destin=
ation URI changes<br>
&gt; &gt; &gt; unpredictably -- elements<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; become inacessible=
 (mobile phone out<br>
&gt; &gt; &gt; of range!), call routing<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; changes due to tim=
e of day, etc.<br>
&gt; &gt; &gt; Instead, SIP provides a<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; remarkably effecti=
ve system for<br>
&gt; &gt; &gt; selecting the best target of<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; a call *at the mom=
ent the call is<br>
&gt; &gt; &gt; initiated* -- the caller<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; declares his prefe=
rences, and the<br>
&gt; &gt; &gt; various intermediate<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; elements combine t=
he caller&#39;s desires<br>
&gt; &gt; &gt; with the routing<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; information that t=
hey possess to<br>
&gt; &gt; &gt; forward the call closer to<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; the preferred dest=
ination.<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; As a developer, I&=
#39;ve had to wrestle<br>
&gt; &gt; &gt; with this. =A0In order to<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; provide clean &quo=
t;Is extension 1234<br>
&gt; &gt; &gt; busy?&quot; information, our &quot;BLF<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; agent&quot; mainta=
ins subscriptions to the<br>
&gt; &gt; &gt; &quot;reg&quot; event package for<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; the AOR &quot;<a h=
ref=3D"mailto:sip%3A1234@example.com">sip:1234@example.com</a><br>
&gt; &gt; &gt; &lt;mailto:<a href=3D"mailto:sip%253A1234@example.com">sip%3=
A1234@example.com</a>&gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; &lt;mailto:<a href=3D"mailto:sip%25=
3A1234@example.com">sip%3A1234@example.com</a><br>
&gt; &gt; &gt; &lt;mailto:<a href=3D"mailto:sip%25253A1234@example.com">sip=
%253A1234@example.com</a>&gt; &gt; &quot;. =A0For each reported contact<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; for the AOR, it ma=
intains a dialog<br>
&gt; &gt; &gt; subscription to the<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; contact. =A0Only b=
y tracking the &quot;reg&quot;<br>
&gt; &gt; &gt; events can it keep track<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; of the UAs availab=
le for that AOR.<br>
&gt; &gt; &gt; And even then, it is at<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; the mercy of netwo=
rk failures.<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; Dale<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; _______________________________________________<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; sipcore mailing li=
st<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; <a href=3D"mailto:=
sipcore@ietf.org">sipcore@ietf.org</a><br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt; <a href=3D"https:/=
/www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank">https://www.ietf.=
org/mailman/listinfo/sipcore</a><br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 _______________________=
________________________<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 sipcore mailing list<br=
>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 <a href=3D"mailto:sipco=
re@ietf.org">sipcore@ietf.org</a><br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt; =A0 =A0 =A0 <a href=3D"https://www.=
ietf.org/mailman/listinfo/sipcore" target=3D"_blank">https://www.ietf.org/m=
ailman/listinfo/sipcore</a><br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; </div></div></blockquote></div><br>

--001485326135607ee60483a53742--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Wed, 7 Apr 2010 15:05:49 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>>>>The forking proxy can also have policies what it returns in
>>>>the 3xx, depending on what the end-users actually represent.
>>>> =20
>>>>And, also with other mechanisms the returned data per end-user may=20
>>>>depend on what type of end-point it represents.
>>>>I don't think that is bound to a specific mechanism as such.
>>>[JRE] Quite true. But Dale's point still holds - you might
>>as well just send the INVITE request and see what happens.
>>=20
>>Of course. But, as I said, that is really bad protocol design,
>[JRE] Why is it bad protocol design?

Since we have a mechanism where we can query users, I think is nicer to use=
.

Now, I am not saying that you need to send OPTIONS prior to every call. My =
main use-case is where the INVITE itself may look different depending on th=
e capabilities. So, again, an INVITE may not even fail, but you may not get=
 exactly what you would have wanted.

>>and it may trigger unwanted resource reservation actions etc in the=20
>>network.
>[JRE] Concerning resource reservation, there are two ways of=20
>looking at it. By doing resource reservation, you find out=20
>straight away if the resources needed to reach the UA are not=20
>available, so resource reservation could be considered a good=20
>thing.

You will most likely not figure out whether the resources are available, in=
 case there is no provisional response etc.

>An OPTIONS transaction is never going to provide this information. On the =
other hand, if you want to avoid resource=20
>reservation, just send an offerless INVITE request.

That is going to bring other issues, if the SDP is part of the service info=
rmation etc.

In addition, the resources do not necessarily be related to the SDP and the=
 media plane.

Regards,

Christer





> > And, there could even be cases where the INVITE doesn't=20
> fail, but you=20
> > don't get the optimized user experience since the request may have=20
> > looked different should you have known what the receiver supports.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> > > > ________________________________
> > > >=20
> > > > 	From: sipcore-bounces@ietf.org
> > > > [mailto:sipcore-bounces@ietf.org] On Behalf Of Hans Erik
> > van Elburg
> > > > 	Sent: 7. huhtikuuta 2010 12:18
> > > > 	To: Elwell, John
> > > > 	Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> > > > 	Subject: Re: [sipcore] SIP OPTIONS: Shall we=20
> work on this?
> > > > =09
> > > > =09
> > > > 	Of course you can implement all kind of=20
> policies, but that makes=20
> > > > this AOR quite of a special user. I would argue that=20
> this is some=20
> > > > kind of service AOR and not a "normal" user AOR. The=20
> service AOR=20
> > > > representing a group of different and dynamically changing=20
> > > > users...
> > > > =09
> > > > 	Demonstrating that this is not a "normal" user.
> > > > =09
> > > > 	I expect that in practice one would solve this=20
> differently...=20
> > > > making the case rather artificial.
> > > > =09
> > > > 	/Hans Erik van Elburg
> > > > =09
> > > > =09
> > > > 	On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John=20
> > > > <john.elwell@siemens-enterprise.com> wrote:
> > > > =09
> > > >=20
> > > >=20
> > > >=20
> > > > 		> -----Original Message-----
> > > > 		> From: Hans Erik van Elburg
> > > > [mailto:ietf.hanserik@gmail.com]
> > > > 		> Sent: 07 April 2010 08:24
> > > > 		> To: Elwell, John
> > > > 		> Cc: WORLEY, DALE R (DALE); Salvatore=20
> Loreto; sipcore@ietf.org
> > > > 		> Subject: Re: [sipcore] SIP OPTIONS:=20
> Shall we work on this?
> > > > 		>
> > > > 	=09
> > > > 		> This is a special case. And the=20
> application you are talking
> > > > 		> about is a B2BUA, right?
> > > > 	=09
> > > > 		[JRE] Not necessarily. A proxy can be=20
> scripted to cycle between=20
> > > > multiple registered contacts. This doesn't make it a B2BUA.
> > > > 	=09
> > > > 		John
> > > > 	=09
> > > >=20
> > > > 		>
> > > > 		> In this case the AOR addressed is an=20
> application address
> > > > 		> without registered contacts. One can=20
> argue that the OPTIONS
> > > > 		> request should return the general=20
> capabilities of the
> > > > 		> application. No OPTIONS forking is=20
> happening in this case, no
> > > > 		> registered contacts. No issue?
> > > > 		>
> > > > 		> In reality what the B2BUA will do=20
> with the OPTION request
> > > > 		> (forward or answer) depends on the=20
> fantasy of the B2BUA=20
> > > > developer.
> > > > 		>
> > > > 		> I dont think this scenario is=20
> actually usefull in discussing
> > > > 		> whether or not to write a BCP on=20
> OPTIONS handling. Or maybe
> > > > 		> it is as implication of B2BUA=20
> behaviours can be taken into=20
> > > > account.
> > > > 		>
> > > > 		> /Hans Erik van Elburg
> > > > 		>
> > > > 		>
> > > > 		>
> > > > 		> On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John
> > > > 		> <john.elwell@siemens-enterprise.com> wrote:
> > > > 		>
> > > > 		>
> > > > 		>       I agree with Dale. What happens to an=20
> > > > OPTIONS request
> > > > 		> will not necessarily be reflected=20
> when an INVITE request is
> > > > 		> sent shortly afterwards. As a further=20
> examples, in contact
> > > > 		> centres, it is usual to distribute=20
> calls cyclically or
> > > > 		> randomly between qualified and=20
> available agents. Sending an
> > > > 		> INVITE request to a UA determined by=20
> responses to an OPTIONS
> > > > 		> request would break that model - the=20
> issuer of the OPTIONS
> > > > 		> and INVITE requests cannot be=20
> expected to obey the same
> > > > 		> rotation rules as the domain proxy=20
> that routes INVITE
> > > > 		> requests. Issuing an INVITE request=20
> to the AOR is unlikely to
> > > > 		> result in it being delivered to the=20
> UA that might be expected
> > > > 		> from analysis of responses to an=20
> OPTIONS request.
> > > > 		>
> > > > 		>       I don't think we should work on trying=20
> > > > to make OPTIONS
> > > > 		> work for this purpose. If we really=20
> need to do anything with
> > > > 		> OPTIONS, we should just issue=20
> warnings about its
> > > > 		> unsuitability for purposes other than=20
> a simple ping.
> > > > 		>
> > > > 		>       John
> > > > 		>
> > > > 		>
> > > > 		>
> > > > 		>       > -----Original Message-----
> > > > 		>       > From: sipcore-bounces@ietf.org
> > > > 		>
> > > > 		>       >=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of
> > > > 		> WORLEY, DALE R (DALE)
> > > > 		>       > Sent: 06 April 2010 21:01
> > > > 		>       > To: Salvatore Loreto; sipcore@ietf.org
> > > > 		>       > Subject: Re: [sipcore] SIP OPTIONS:=20
> > > > Shall we work on this?
> > > > 		>       >
> > > > 		>       >=20
> ________________________________________
> > > > 		>
> > > > 		>       > From: sipcore-bounces@ietf.org=20
> > > > [sipcore-bounces@ietf.org] On
> > > > 		>       > Behalf Of Salvatore Loreto=20
> > > > [salvatore.loreto@ericsson.com]
> > > > 		>       > Sent: Thursday, April 01, 2010 3:36 AM
> > > > 		>       >
> > > > 		>       > A good mechanism to discovery UA=20
> > > > capabilities is essential to
> > > > 		>       > provide meaningful service:
> > > > 		>       > So I am in favor of pursuing the work,
> > > > 		>       > in particular the possibility to use=20
> > > > a single OPTIONS request
> > > > 		>       > to retrieve the capabilities from ALL=20
> > > > UAS(s) behind a
> > > > 		> forking proxy.
> > > > 		>       >=20
> _______________________________________
> > > > 		>       >
> > > > 		>       > I think it is a misconception to use=20
> > > > UA discovery to provide
> > > > 		>       > services.  The difficulty is that the=20
> > > > set of available UAs
> > > > 		>       > for a given destination URI changes=20
> > > > unpredictably -- elements
> > > > 		>       > become inacessible (mobile phone out=20
> > > > of range!), call routing
> > > > 		>       > changes due to time of day, etc. =20
> > > > Instead, SIP provides a
> > > > 		>       > remarkably effective system for=20
> > > > selecting the best target of
> > > > 		>       > a call *at the moment the call is=20
> > > > initiated* -- the caller
> > > > 		>       > declares his preferences, and the=20
> > > > various intermediate
> > > > 		>       > elements combine the caller's desires=20
> > > > with the routing
> > > > 		>       > information that they possess to=20
> > > > forward the call closer to
> > > > 		>       > the preferred destination.
> > > > 		>       >
> > > > 		>       > As a developer, I've had to wrestle=20
> > > > with this.  In order to
> > > > 		>       > provide clean "Is extension 1234=20
> > > > busy?" information, our "BLF
> > > > 		>       > agent" maintains subscriptions to the=20
> > > > "reg" event package for
> > > > 		>       > the AOR "sip:1234@example.com=20
> > > > <mailto:sip%3A1234@example.com>
> > > > 	=09
> > > > 		> <mailto:sip%3A1234@example.com=20
> > > > <mailto:sip%253A1234@example.com> > ".  For each=20
> reported contact
> > > > 	=09
> > > > 		>       > for the AOR, it maintains a dialog=20
> > > > subscription to the
> > > > 		>       > contact.  Only by tracking the "reg"=20
> > > > events can it keep track
> > > > 		>       > of the UAs available for that AOR. =20
> > > > And even then, it is at
> > > > 		>       > the mercy of network failures.
> > > > 		>       >
> > > > 		>       > Dale
> > > > 		>       >=20
> > > > _______________________________________________
> > > > 		>       > sipcore mailing list
> > > > 		>       > sipcore@ietf.org
> > > > 		>       >=20
> https://www.ietf.org/mailman/listinfo/sipcore
> > > > 		>       >
> > > > 		>      =20
> _______________________________________________
> > > > 		>       sipcore mailing list
> > > > 		>       sipcore@ietf.org
> > > > 		>      =20
> https://www.ietf.org/mailman/listinfo/sipcore
> > > > 		>
> > > > 		>
> > > > 		>
> > > > 		>
> > > >=20
> > > >=20
> > > > =

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To: Christer Holmberg <christer.holmberg@ericsson.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Wed, 7 Apr 2010 15:30:03 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 07 April 2010 14:06
> To: Elwell, John; Hans Erik van Elburg
> Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
>=20
> Hi,=20
>=20
> >>>>The forking proxy can also have policies what it returns in
> >>>>the 3xx, depending on what the end-users actually represent.
> >>>> =20
> >>>>And, also with other mechanisms the returned data per=20
> end-user may=20
> >>>>depend on what type of end-point it represents.
> >>>>I don't think that is bound to a specific mechanism as such.
> >>>[JRE] Quite true. But Dale's point still holds - you might
> >>as well just send the INVITE request and see what happens.
> >>=20
> >>Of course. But, as I said, that is really bad protocol design,
> >[JRE] Why is it bad protocol design?
>=20
> Since we have a mechanism where we can query users, I think=20
> is nicer to use.
[JRE] But even if you then target the INVITE request at the GRUU of the pre=
ferred user, there is no guarantee it will succeed (e.g., no media resource=
s, UA has become busy in the meantime), and because it is targeted directly=
, you lose the opportunity to take alternative action, e.g., forwarding to =
voicemail or to another UA with similar capabilities. It is a case of the c=
alling user trying to override policies associated with the called user, wh=
ich I think is a bad thing. The result of the OPTIONS request can only ever=
 act as a guideline - it is effectively no better than presence.

It would be helpful to have a concrete example of something that could go i=
nto an INVITE request targeted at a particular UA known to have a particula=
r capability, that could not go into an INVITE request targeted at the AOR =
and with caller preferences RECOMMENDING selection of a UA with that partic=
ular capability. Apologies if such an example has already been given earlie=
r in the discussion.

John


>=20
> Now, I am not saying that you need to send OPTIONS prior to=20
> every call. My main use-case is where the INVITE itself may=20
> look different depending on the capabilities. So, again, an=20
> INVITE may not even fail, but you may not get exactly what=20
> you would have wanted.
>=20
> >>and it may trigger unwanted resource reservation actions etc in the=20
> >>network.
> >[JRE] Concerning resource reservation, there are two ways of=20
> >looking at it. By doing resource reservation, you find out=20
> >straight away if the resources needed to reach the UA are not=20
> >available, so resource reservation could be considered a good=20
> >thing.
>=20
> You will most likely not figure out whether the resources are=20
> available, in case there is no provisional response etc.
>=20
> >An OPTIONS transaction is never going to provide this=20
> information. On the other hand, if you want to avoid resource=20
> >reservation, just send an offerless INVITE request.
>=20
> That is going to bring other issues, if the SDP is part of=20
> the service information etc.
>=20
> In addition, the resources do not necessarily be related to=20
> the SDP and the media plane.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
> > > And, there could even be cases where the INVITE doesn't=20
> > fail, but you=20
> > > don't get the optimized user experience since the request=20
> may have=20
> > > looked different should you have known what the receiver supports.
> > >=20
> > > Regards,
> > >=20
> > > Christer
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > > > > ________________________________
> > > > >=20
> > > > > 	From: sipcore-bounces@ietf.org
> > > > > [mailto:sipcore-bounces@ietf.org] On Behalf Of Hans Erik
> > > van Elburg
> > > > > 	Sent: 7. huhtikuuta 2010 12:18
> > > > > 	To: Elwell, John
> > > > > 	Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> > > > > 	Subject: Re: [sipcore] SIP OPTIONS: Shall we=20
> > work on this?
> > > > > =09
> > > > > =09
> > > > > 	Of course you can implement all kind of=20
> > policies, but that makes=20
> > > > > this AOR quite of a special user. I would argue that=20
> > this is some=20
> > > > > kind of service AOR and not a "normal" user AOR. The=20
> > service AOR=20
> > > > > representing a group of different and dynamically changing=20
> > > > > users...
> > > > > =09
> > > > > 	Demonstrating that this is not a "normal" user.
> > > > > =09
> > > > > 	I expect that in practice one would solve this=20
> > differently...=20
> > > > > making the case rather artificial.
> > > > > =09
> > > > > 	/Hans Erik van Elburg
> > > > > =09
> > > > > =09
> > > > > 	On Wed, Apr 7, 2010 at 10:41 AM, Elwell, John=20
> > > > > <john.elwell@siemens-enterprise.com> wrote:
> > > > > =09
> > > > >=20
> > > > >=20
> > > > >=20
> > > > > 		> -----Original Message-----
> > > > > 		> From: Hans Erik van Elburg
> > > > > [mailto:ietf.hanserik@gmail.com]
> > > > > 		> Sent: 07 April 2010 08:24
> > > > > 		> To: Elwell, John
> > > > > 		> Cc: WORLEY, DALE R (DALE); Salvatore=20
> > Loreto; sipcore@ietf.org
> > > > > 		> Subject: Re: [sipcore] SIP OPTIONS:=20
> > Shall we work on this?
> > > > > 		>
> > > > > 	=09
> > > > > 		> This is a special case. And the=20
> > application you are talking
> > > > > 		> about is a B2BUA, right?
> > > > > 	=09
> > > > > 		[JRE] Not necessarily. A proxy can be=20
> > scripted to cycle between=20
> > > > > multiple registered contacts. This doesn't make it a B2BUA.
> > > > > 	=09
> > > > > 		John
> > > > > 	=09
> > > > >=20
> > > > > 		>
> > > > > 		> In this case the AOR addressed is an=20
> > application address
> > > > > 		> without registered contacts. One can=20
> > argue that the OPTIONS
> > > > > 		> request should return the general=20
> > capabilities of the
> > > > > 		> application. No OPTIONS forking is=20
> > happening in this case, no
> > > > > 		> registered contacts. No issue?
> > > > > 		>
> > > > > 		> In reality what the B2BUA will do=20
> > with the OPTION request
> > > > > 		> (forward or answer) depends on the=20
> > fantasy of the B2BUA=20
> > > > > developer.
> > > > > 		>
> > > > > 		> I dont think this scenario is=20
> > actually usefull in discussing
> > > > > 		> whether or not to write a BCP on=20
> > OPTIONS handling. Or maybe
> > > > > 		> it is as implication of B2BUA=20
> > behaviours can be taken into=20
> > > > > account.
> > > > > 		>
> > > > > 		> /Hans Erik van Elburg
> > > > > 		>
> > > > > 		>
> > > > > 		>
> > > > > 		> On Wed, Apr 7, 2010 at 8:31 AM, Elwell, John
> > > > > 		> <john.elwell@siemens-enterprise.com> wrote:
> > > > > 		>
> > > > > 		>
> > > > > 		>       I agree with Dale. What happens to an=20
> > > > > OPTIONS request
> > > > > 		> will not necessarily be reflected=20
> > when an INVITE request is
> > > > > 		> sent shortly afterwards. As a further=20
> > examples, in contact
> > > > > 		> centres, it is usual to distribute=20
> > calls cyclically or
> > > > > 		> randomly between qualified and=20
> > available agents. Sending an
> > > > > 		> INVITE request to a UA determined by=20
> > responses to an OPTIONS
> > > > > 		> request would break that model - the=20
> > issuer of the OPTIONS
> > > > > 		> and INVITE requests cannot be=20
> > expected to obey the same
> > > > > 		> rotation rules as the domain proxy=20
> > that routes INVITE
> > > > > 		> requests. Issuing an INVITE request=20
> > to the AOR is unlikely to
> > > > > 		> result in it being delivered to the=20
> > UA that might be expected
> > > > > 		> from analysis of responses to an=20
> > OPTIONS request.
> > > > > 		>
> > > > > 		>       I don't think we should work on trying=20
> > > > > to make OPTIONS
> > > > > 		> work for this purpose. If we really=20
> > need to do anything with
> > > > > 		> OPTIONS, we should just issue=20
> > warnings about its
> > > > > 		> unsuitability for purposes other than=20
> > a simple ping.
> > > > > 		>
> > > > > 		>       John
> > > > > 		>
> > > > > 		>
> > > > > 		>
> > > > > 		>       > -----Original Message-----
> > > > > 		>       > From: sipcore-bounces@ietf.org
> > > > > 		>
> > > > > 		>       >=20
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of
> > > > > 		> WORLEY, DALE R (DALE)
> > > > > 		>       > Sent: 06 April 2010 21:01
> > > > > 		>       > To: Salvatore Loreto; sipcore@ietf.org
> > > > > 		>       > Subject: Re: [sipcore] SIP OPTIONS:=20
> > > > > Shall we work on this?
> > > > > 		>       >
> > > > > 		>       >=20
> > ________________________________________
> > > > > 		>
> > > > > 		>       > From: sipcore-bounces@ietf.org=20
> > > > > [sipcore-bounces@ietf.org] On
> > > > > 		>       > Behalf Of Salvatore Loreto=20
> > > > > [salvatore.loreto@ericsson.com]
> > > > > 		>       > Sent: Thursday, April 01, 2010 3:36 AM
> > > > > 		>       >
> > > > > 		>       > A good mechanism to discovery UA=20
> > > > > capabilities is essential to
> > > > > 		>       > provide meaningful service:
> > > > > 		>       > So I am in favor of pursuing the work,
> > > > > 		>       > in particular the possibility to use=20
> > > > > a single OPTIONS request
> > > > > 		>       > to retrieve the capabilities from ALL=20
> > > > > UAS(s) behind a
> > > > > 		> forking proxy.
> > > > > 		>       >=20
> > _______________________________________
> > > > > 		>       >
> > > > > 		>       > I think it is a misconception to use=20
> > > > > UA discovery to provide
> > > > > 		>       > services.  The difficulty is that the=20
> > > > > set of available UAs
> > > > > 		>       > for a given destination URI changes=20
> > > > > unpredictably -- elements
> > > > > 		>       > become inacessible (mobile phone out=20
> > > > > of range!), call routing
> > > > > 		>       > changes due to time of day, etc. =20
> > > > > Instead, SIP provides a
> > > > > 		>       > remarkably effective system for=20
> > > > > selecting the best target of
> > > > > 		>       > a call *at the moment the call is=20
> > > > > initiated* -- the caller
> > > > > 		>       > declares his preferences, and the=20
> > > > > various intermediate
> > > > > 		>       > elements combine the caller's desires=20
> > > > > with the routing
> > > > > 		>       > information that they possess to=20
> > > > > forward the call closer to
> > > > > 		>       > the preferred destination.
> > > > > 		>       >
> > > > > 		>       > As a developer, I've had to wrestle=20
> > > > > with this.  In order to
> > > > > 		>       > provide clean "Is extension 1234=20
> > > > > busy?" information, our "BLF
> > > > > 		>       > agent" maintains subscriptions to the=20
> > > > > "reg" event package for
> > > > > 		>       > the AOR "sip:1234@example.com=20
> > > > > <mailto:sip%3A1234@example.com>
> > > > > 	=09
> > > > > 		> <mailto:sip%3A1234@example.com=20
> > > > > <mailto:sip%253A1234@example.com> > ".  For each=20
> > reported contact
> > > > > 	=09
> > > > > 		>       > for the AOR, it maintains a dialog=20
> > > > > subscription to the
> > > > > 		>       > contact.  Only by tracking the "reg"=20
> > > > > events can it keep track
> > > > > 		>       > of the UAs available for that AOR. =20
> > > > > And even then, it is at
> > > > > 		>       > the mercy of network failures.
> > > > > 		>       >
> > > > > 		>       > Dale
> > > > > 		>       >=20
> > > > > _______________________________________________
> > > > > 		>       > sipcore mailing list
> > > > > 		>       > sipcore@ietf.org
> > > > > 		>       >=20
> > https://www.ietf.org/mailman/listinfo/sipcore
> > > > > 		>       >
> > > > > 		>      =20
> > _______________________________________________
> > > > > 		>       sipcore mailing list
> > > > > 		>       sipcore@ietf.org
> > > > > 		>      =20
> > https://www.ietf.org/mailman/listinfo/sipcore
> > > > > 		>
> > > > > 		>
> > > > > 		>
> > > > > 		>
> > > > >=20
> > > > >=20
> > > > > =

From HKaplan@acmepacket.com  Wed Apr  7 06:53:17 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 7 Apr 2010 09:53:12 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Christer Holmberg
> Sent: Tuesday, April 06, 2010 4:08 PM
> To: WORLEY, DALE R (DALE); Salvatore Loreto; sipcore@ietf.org
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
> Hi Dale,
>=20
> From my perspective, the main use-case is not to discover services, but t=
o
> discover capabilities in order to determine what services can be applied.
>=20
> The idea is also that the OPTIONS request would be sent prior to the
> INVITE, to try to make sure that the returned contacts are actually
> available.

But you know that would never work in practice/the-real-world.  Think about=
 all the routing rules people have for INVITEs - in some cases *only* for I=
NVITEs.  And even if they route OPTIONS identically in the SIP domain, for =
any given AoR it'll go to any of: a SIP UA, or a PSTN gateway, or a H.323-g=
ateway, or an MGCP call-agent, or an IP-PBX, or a foobar-protocol-gateway, =
or a vmail server, or an announcement server, or an app server, etc.  Just =
because the OPTIONS reaches the end of the SIP domain, means nothing about =
whether an INVITE to the higher-layer target (which in practice is a E.164 =
number, that crosses from SIP into many other protocol domains) is actually=
 available to receive a call.

-hadriel

From HKaplan@acmepacket.com  Wed Apr  7 07:01:14 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 7 Apr 2010 10:01:08 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Christer Holmberg
> Sent: Tuesday, April 06, 2010 4:55 PM
> >
> > Dale Worley said:
> > Why not just send the INVITE and see if it fails?  In the most common
> > case (success), that saves one round-trip.
>=20
> Because that is crappy protocol design. And, the INVITE might trigger
> different resources etc to be reserved, so you don't want to keep sending
> those.

What's so crappy about using an INVITE to say you want to start a session? =
 A failure of an INVITE request isn't a protocol failure or error condition=
 (unless the response code is an error condition type). =20

What's next, we need an OPTIONS before the OPTIONS to make sure the second =
OPTIONS won't fail??  ;)

Sure the INVITE might trigger resources reservation - that's GOOD, because =
it verifies the resources are available - if those resources aren't availab=
le the request will fail, whereas for an OPTIONS it wouldn't have.

-hadriel

From ietf.hanserik@gmail.com  Wed Apr  7 07:05:23 2010
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In such cases you will get the capabilities of the gateway where the SIP
domain is left. Why isn't that info useful?

/Hans Erik van Elburg

On Wed, Apr 7, 2010 at 3:53 PM, Hadriel Kaplan <HKaplan@acmepacket.com>wrote:

>
>
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf
> > Of Christer Holmberg
> > Sent: Tuesday, April 06, 2010 4:08 PM
> > To: WORLEY, DALE R (DALE); Salvatore Loreto; sipcore@ietf.org
> > Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> >
> > Hi Dale,
> >
> > From my perspective, the main use-case is not to discover services, but
> to
> > discover capabilities in order to determine what services can be applied.
> >
> > The idea is also that the OPTIONS request would be sent prior to the
> > INVITE, to try to make sure that the returned contacts are actually
> > available.
>
> But you know that would never work in practice/the-real-world.  Think about
> all the routing rules people have for INVITEs - in some cases *only* for
> INVITEs.  And even if they route OPTIONS identically in the SIP domain, for
> any given AoR it'll go to any of: a SIP UA, or a PSTN gateway, or a
> H.323-gateway, or an MGCP call-agent, or an IP-PBX, or a
> foobar-protocol-gateway, or a vmail server, or an announcement server, or an
> app server, etc.  Just because the OPTIONS reaches the end of the SIP
> domain, means nothing about whether an INVITE to the higher-layer target
> (which in practice is a E.164 number, that crosses from SIP into many other
> protocol domains) is actually available to receive a call.
>
> -hadriel
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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In such cases you will get the capabilities of the gateway where the SIP do=
main is left. Why isn&#39;t that info useful?<br><br clear=3D"all">/Hans Er=
ik van Elburg<br>
<br><div class=3D"gmail_quote">On Wed, Apr 7, 2010 at 3:53 PM, Hadriel Kapl=
an <span dir=3D"ltr">&lt;<a href=3D"mailto:HKaplan@acmepacket.com">HKaplan@=
acmepacket.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" s=
tyle=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204=
); padding-left: 1ex;">
<div class=3D"im"><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf=
.org</a> [mailto:<a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounce=
s@ietf.org</a>] On Behalf<br>
&gt; Of Christer Holmberg<br>
&gt; Sent: Tuesday, April 06, 2010 4:08 PM<br>
&gt; To: WORLEY, DALE R (DALE); Salvatore Loreto; <a href=3D"mailto:sipcore=
@ietf.org">sipcore@ietf.org</a><br>
&gt; Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?<br>
&gt;<br>
</div><div class=3D"im">&gt; Hi Dale,<br>
&gt;<br>
&gt; From my perspective, the main use-case is not to discover services, bu=
t to<br>
&gt; discover capabilities in order to determine what services can be appli=
ed.<br>
&gt;<br>
&gt; The idea is also that the OPTIONS request would be sent prior to the<b=
r>
&gt; INVITE, to try to make sure that the returned contacts are actually<br=
>
&gt; available.<br>
<br>
</div>But you know that would never work in practice/the-real-world. =A0Thi=
nk about all the routing rules people have for INVITEs - in some cases *onl=
y* for INVITEs. =A0And even if they route OPTIONS identically in the SIP do=
main, for any given AoR it&#39;ll go to any of: a SIP UA, or a PSTN gateway=
, or a H.323-gateway, or an MGCP call-agent, or an IP-PBX, or a foobar-prot=
ocol-gateway, or a vmail server, or an announcement server, or an app serve=
r, etc. =A0Just because the OPTIONS reaches the end of the SIP domain, mean=
s nothing about whether an INVITE to the higher-layer target (which in prac=
tice is a E.164 number, that crosses from SIP into many other protocol doma=
ins) is actually available to receive a call.<br>

<font color=3D"#888888"><br>
-hadriel<br>
</font><div><div></div><div class=3D"h5">__________________________________=
_____________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</div></div></blockquote></div><br>

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From HKaplan@acmepacket.com  Wed Apr  7 07:32:12 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Wed, 7 Apr 2010 10:31:59 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Because it means nothing about the capabilities for the INVITE, because the=
 INVITE session's capabilities is dictated by the far-end beyond that gatew=
ay.

For example, I don't know what types of capabilities you guys are thinking =
of, but let's say it was for getting the codecs.  So you send an OPTIONS, a=
nd it gets to an H.323 gateway.  The gateway supports lots of codecs, so it=
 gives you some nice big SDP.  Then you send an INVITE, thinking it can do =
some new-fangled codec.  But that INVITE makes an H.323 call get setup, whe=
re the H.245 from the far-end only did G.711, so the call fails.

Then there are 3PCC call scenarios, and middlebox transcoders, and codec-ba=
sed routing, etc.  Different things happen with INVITES, at different times=
, due to different policies - none of which apply to OPTIONS.
The reality of the INVITE world is ugly. :(

-hadriel

________________________________
From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
Sent: Wednesday, April 07, 2010 10:05 AM
To: Hadriel Kaplan
Cc: Christer Holmberg; WORLEY, DALE R (DALE); Salvatore Loreto; sipcore@iet=
f.org
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?

In such cases you will get the capabilities of the gateway where the SIP do=
main is left. Why isn't that info useful?

/Hans Erik van Elburg
On Wed, Apr 7, 2010 at 3:53 PM, Hadriel Kaplan <HKaplan@acmepacket.com<mail=
to:HKaplan@acmepacket.com>> wrote:


> -----Original Message-----
> From: sipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org> [mailto:s=
ipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org>] On Behalf
> Of Christer Holmberg
> Sent: Tuesday, April 06, 2010 4:08 PM
> To: WORLEY, DALE R (DALE); Salvatore Loreto; sipcore@ietf.org<mailto:sipc=
ore@ietf.org>
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>
> Hi Dale,
>
> From my perspective, the main use-case is not to discover services, but t=
o
> discover capabilities in order to determine what services can be applied.
>
> The idea is also that the OPTIONS request would be sent prior to the
> INVITE, to try to make sure that the returned contacts are actually
> available.
But you know that would never work in practice/the-real-world.  Think about=
 all the routing rules people have for INVITEs - in some cases *only* for I=
NVITEs.  And even if they route OPTIONS identically in the SIP domain, for =
any given AoR it'll go to any of: a SIP UA, or a PSTN gateway, or a H.323-g=
ateway, or an MGCP call-agent, or an IP-PBX, or a foobar-protocol-gateway, =
or a vmail server, or an announcement server, or an app server, etc.  Just =
because the OPTIONS reaches the end of the SIP domain, means nothing about =
whether an INVITE to the higher-layer target (which in practice is a E.164 =
number, that crosses from SIP into many other protocol domains) is actually=
 available to receive a call.

-hadriel
_______________________________________________
sipcore mailing list
sipcore@ietf.org<mailto:sipcore@ietf.org>
https://www.ietf.org/mailman/listinfo/sipcore


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Because it means nothing about the
capabilities for the INVITE, because the INVITE session&#8217;s capabilitie=
s is
dictated by the far-end beyond that gateway. &nbsp;<o:p></o:p></span></font=
></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>For example, I don&#8217;t know what t=
ypes
of capabilities you guys are thinking of, but let&#8217;s say it was for
getting the codecs.&nbsp; So you send an OPTIONS, and it gets to an H.323
gateway. &nbsp;The gateway supports lots of codecs, so it gives you some ni=
ce
big SDP. &nbsp;Then you send an INVITE, thinking it can do some new-fangled
codec.&nbsp; But that INVITE makes an H.323 call get setup, where the H.245
from the far-end only did G.711, so the call fails.<o:p></o:p></span></font=
></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Then there are 3PCC call scenarios, an=
d
middlebox transcoders, and codec-based routing, etc.&nbsp; Different things
happen with INVITES, at different times, due to different policies &#8211; =
none
of which apply to OPTIONS.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>The reality of the INVITE world is ugl=
y. </span></font><font
size=3D2 color=3Dnavy face=3DWingdings><span style=3D'font-size:10.0pt;font=
-family:
Wingdings;color:navy'>L</span></font><font size=3D2 color=3Dnavy face=3DAri=
al><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p></o:p></span><=
/font></p>

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

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

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

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Hans Eri=
k van
Elburg [mailto:ietf.hanserik@gmail.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, April 07, 2=
010
10:05 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Hadriel Kaplan<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> Christer Holmberg; WORLE=
Y,
DALE R (DALE); Salvatore Loreto; sipcore@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [sipcore] SIP
OPTIONS: Shall we work on this?</span></font><o:p></o:p></p>

</div>

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

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>In such cases you=
 will
get the capabilities of the gateway where the SIP domain is left. Why isn't
that info useful?<br>
<br clear=3Dall>
/Hans Erik van Elburg<o:p></o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>On Wed, Apr 7, 2010 at 3:53 PM, Hadriel Kaplan &lt;<a
href=3D"mailto:HKaplan@acmepacket.com">HKaplan@acmepacket.com</a>&gt; wrote=
:<o:p></o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf=
.org</a>
[mailto:<a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.or=
g</a>]
On Behalf<br>
&gt; Of Christer Holmberg<br>
&gt; Sent: Tuesday, April 06, 2010 4:08 PM<br>
&gt; To: WORLEY, DALE R (DALE); Salvatore Loreto; <a
href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
&gt; Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?<br>
&gt;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&gt; Hi Dale,<br>
&gt;<br>
&gt; From my perspective, the main use-case is not to discover services, bu=
t to<br>
&gt; discover capabilities in order to determine what services can be appli=
ed.<br>
&gt;<br>
&gt; The idea is also that the OPTIONS request would be sent prior to the<b=
r>
&gt; INVITE, to try to make sure that the returned contacts are actually<br=
>
&gt; available.<o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>But you know that would never work in practice/the-real-world.
&nbsp;Think about all the routing rules people have for INVITEs - in some c=
ases
*only* for INVITEs. &nbsp;And even if they route OPTIONS identically in the=
 SIP
domain, for any given AoR it'll go to any of: a SIP UA, or a PSTN gateway, =
or a
H.323-gateway, or an MGCP call-agent, or an IP-PBX, or a
foobar-protocol-gateway, or a vmail server, or an announcement server, or a=
n
app server, etc. &nbsp;Just because the OPTIONS reaches the end of the SIP
domain, means nothing about whether an INVITE to the higher-layer target (w=
hich
in practice is a E.164 number, that crosses from SIP into many other protoc=
ol
domains) is actually available to receive a call.<br>
<font color=3D"#888888"><span style=3D'color:#888888'><br>
-hadriel</span></font><o:p></o:p></span></font></p>

<div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><o:p></o:p></span></font>=
</p>

</div>

</div>

</div>

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

</div>

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From keith.drage@alcatel-lucent.com  Wed Apr  7 07:44:22 2010
Return-Path: <keith.drage@alcatel-lucent.com>
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 7 Apr 2010 16:43:21 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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At the moment, my view is "Do nothing". The amount of resource needed for t=
his discussion does not justify the problem.

If someone wants to submit an author draft on a specific issue and fix, the=
n fine, I'll reconsider.

regards

Keith

________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Adam Roach
Sent: Tuesday, March 30, 2010 8:06 PM
To: SIPCORE
Subject: [sipcore] SIP OPTIONS: Shall we work on this?

[as chair]

One of the topics of discussion at the recent face-to-face meeting in Anahe=
im was the SIP OPTIONS method. We have had significant discussion of some s=
pecific technical topics on the SIPCORE mailing list, such that all partici=
pants should have a reasonable overview of the scope that any related work =
may entail. Participants may also benefit from reading through the minutes =
of the corresponding conversation in Anaheim:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTI=
ONS%20Method

Rather than continue discussion of the specific technical topics related to=
 OPTIONS, the chairs would encourage the working group to engage in a discu=
ssion around whether we should, as a group, work on addressing some of the =
issues that have been identified.

In particular, we would direct the group to consider the nature of such wor=
k:

 *   Are we proposing to introduce new features, or fix existing bugs?
 *   If we are adding new features: is OPTIONS the best way to introduce th=
ese features, or would other mechanisms (presence, callee-caps) produce sup=
erior results?
 *   If we are fixing bugs: are the bugs theoretical, or can we cite actual=
 interoperability issues in the field?
 *   If there are identified interoperability issues: is the level of effor=
t involved in fixing them commensurate with the impact of the problems?

Finally, keep in mind that work does not happen unless participants are wil=
ling to invest time. If you speak out in favor of pursuing this work, pleas=
e indicate the level of time commitment that you will personally invest in =
bringing such work to a successful conclusion (e.g., will author drafts; wi=
ll perform dedicated reviews; etc).

Thanks.

/a

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<HTML><HEAD>
<META http-equiv=3Dcontent-type content=3D"text/html; charset=3DISO-8859-1"=
>
<META content=3D"MSHTML 6.00.2900.3660" name=3DGENERATOR></HEAD>
<BODY text=3D#000000 bgColor=3D#ffffff>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D655004214-07042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>At the moment, my view is "Do nothing". The amount=
 of=20
resource needed for this discussion does not justify the=20
problem.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D655004214-07042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D655004214-07042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>If someone wants to submit an author draft on a sp=
ecific=20
issue and fix, then fine, I'll reconsider.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D655004214-07042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D655004214-07042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>regards</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D655004214-07042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D655004214-07042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Keith</FONT></SPAN></DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sipcore-bounces@ietf.org=20
  [mailto:sipcore-bounces@ietf.org] <B>On Behalf Of </B>Adam=20
  Roach<BR><B>Sent:</B> Tuesday, March 30, 2010 8:06 PM<BR><B>To:</B>=20
  SIPCORE<BR><B>Subject:</B> [sipcore] SIP OPTIONS: Shall we work on=20
  this?<BR></FONT><BR></DIV>
  <DIV></DIV>[as chair]<BR><BR>One of the topics of discussion at the recen=
t=20
  face-to-face meeting in Anaheim was the SIP OPTIONS method. We have had=20
  significant discussion of some specific technical topics on the SIPCORE=20
  mailing list, such that all participants should have a reasonable overvie=
w of=20
  the scope that any related work may entail. Participants may also benefit=
 from=20
  reading through the minutes of the corresponding conversation in=20
  Anaheim:<BR><BR><A class=3Dmoz-txt-link-freetext=20
  href=3D"http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20=
SIP%20OPTIONS%20Method">http://www.ietf.org/proceedings/10mar/minutes/sipco=
re.html#The%20SIP%20OPTIONS%20Method</A><BR><BR>Rather=20
  than continue discussion of the specific technical topics related to OPTI=
ONS,=20
  the chairs would encourage the working group to engage in a discussion ar=
ound=20
  whether we should, as a group, work on addressing some of the issues that=
 have=20
  been identified.<BR><BR>In particular, we would direct the group to consi=
der=20
  the nature of such work:<BR>
  <UL>
    <LI>Are we proposing to introduce new features, or fix existing bugs?=20
    <LI>If we are adding new features: is OPTIONS the best way to introduce=
=20
    these features, or would other mechanisms (presence, callee-caps) produ=
ce=20
    superior results?<BR>
    <LI>If we are fixing bugs: are the bugs theoretical, or can we cite act=
ual=20
    interoperability issues in the field?=20
    <LI>If there are identified interoperability issues: is the level of ef=
fort=20
    involved in fixing them commensurate with the impact of the problems?=20
  </LI></UL>Finally, keep in mind that work does not happen unless particip=
ants=20
  are willing to invest time. If you speak out in favor of pursuing this wo=
rk,=20
  please indicate the level of time commitment that you will personally inv=
est=20
  in bringing such work to a successful conclusion (e.g., will author draft=
s;=20
  will perform dedicated reviews;=20
etc).<BR><BR>Thanks.<BR><BR>/a<BR></BLOCKQUOTE></BODY></HTML>

--_000_EDC0A1AE77C57744B664A310A0B23AE211CCD821FRMRSSXCHMBSC3d_--

From adam@nostrum.com  Wed Apr  7 07:49:36 2010
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To: Hadriel Kaplan <HKaplan@acmepacket.com>
References: <4BB24B81.1050006@nostrum.com> <4BB44CE6.2070402@ericsson.com>	<CD5674C3CD99574EBA7432465FC13C1B21F6E96F97@DC-US1MBEX4.global.avaya.com>	<FF84A09F50A6DC48ACB6714F4666CC745E21B30A7F@ESESSCMS0354.eemea.ericsson.se>	<430FC6BDED356B4C8498F634416644A91A79F3CFC6@mail>	<v2z9ae56b1e1004070705x463059b3r298422c8cac82fa9@mail.gmail.com> <430FC6BDED356B4C8498F634416644A91A79F3CFE9@mail>
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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[as an individual]

I think Hadriel is right.

The more I read through this thread, the more it looks like we're 
running around with a hammer, desperately looking for nails to hit. 
Barring nails, we'll settle for screws. Or bolts. Or maybe kittens, if 
it comes to that.

If you're trying to accomplish something, I would strongly recommend 
starting from the other end. Instead of saying "I have a tool, let's 
make a problem for it to solve," let's concentrate on clearly defining 
existing problems. Once we agree on the problems, we can talk about what 
tools make sense to solve them.

/a

On 4/7/10 9:31 AM, Hadriel Kaplan wrote:
>
> Because it means nothing about the capabilities for the INVITE, 
> because the INVITE session's capabilities is dictated by the far-end 
> beyond that gateway.
>
> For example, I don't know what types of capabilities you guys are 
> thinking of, but let's say it was for getting the codecs.  So you send 
> an OPTIONS, and it gets to an H.323 gateway.  The gateway supports 
> lots of codecs, so it gives you some nice big SDP.  Then you send an 
> INVITE, thinking it can do some new-fangled codec.  But that INVITE 
> makes an H.323 call get setup, where the H.245 from the far-end only 
> did G.711, so the call fails.
>
> Then there are 3PCC call scenarios, and middlebox transcoders, and 
> codec-based routing, etc.  Different things happen with INVITES, at 
> different times, due to different policies -- none of which apply to 
> OPTIONS.
>
> The reality of the INVITE world is ugly. L
>
> -hadriel
>
> ------------------------------------------------------------------------
>
> *From:* Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> *Sent:* Wednesday, April 07, 2010 10:05 AM
> *To:* Hadriel Kaplan
> *Cc:* Christer Holmberg; WORLEY, DALE R (DALE); Salvatore Loreto; 
> sipcore@ietf.org
> *Subject:* Re: [sipcore] SIP OPTIONS: Shall we work on this?
>
> In such cases you will get the capabilities of the gateway where the 
> SIP domain is left. Why isn't that info useful?
>
> /Hans Erik van Elburg
>
> On Wed, Apr 7, 2010 at 3:53 PM, Hadriel Kaplan <HKaplan@acmepacket.com 
> <mailto:HKaplan@acmepacket.com>> wrote:
>
>
>
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org <mailto:sipcore-bounces@ietf.org> 
> [mailto:sipcore-bounces@ietf.org <mailto:sipcore-bounces@ietf.org>] On 
> Behalf
> > Of Christer Holmberg
> > Sent: Tuesday, April 06, 2010 4:08 PM
> > To: WORLEY, DALE R (DALE); Salvatore Loreto; sipcore@ietf.org 
> <mailto:sipcore@ietf.org>
> > Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> >
>
> > Hi Dale,
> >
> > From my perspective, the main use-case is not to discover services, 
> but to
> > discover capabilities in order to determine what services can be applied.
> >
> > The idea is also that the OPTIONS request would be sent prior to the
> > INVITE, to try to make sure that the returned contacts are actually
> > available.
>
> But you know that would never work in practice/the-real-world.  Think 
> about all the routing rules people have for INVITEs - in some cases 
> *only* for INVITEs.  And even if they route OPTIONS identically in the 
> SIP domain, for any given AoR it'll go to any of: a SIP UA, or a PSTN 
> gateway, or a H.323-gateway, or an MGCP call-agent, or an IP-PBX, or a 
> foobar-protocol-gateway, or a vmail server, or an announcement server, 
> or an app server, etc.  Just because the OPTIONS reaches the end of 
> the SIP domain, means nothing about whether an INVITE to the 
> higher-layer target (which in practice is a E.164 number, that crosses 
> from SIP into many other protocol domains) is actually available to 
> receive a call.
>
> -hadriel
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org <mailto:sipcore@ietf.org>
> https://www.ietf.org/mailman/listinfo/sipcore
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>    


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[as an individual]<br>
<br>
I think Hadriel is right.<br>
<br>
The more I read through this thread, the more it looks like we're
running around with a hammer, desperately looking for nails to hit.
Barring nails, we'll settle for screws. Or bolts. Or maybe kittens, if
it comes to that.<br>
<br>
If you're trying to accomplish something, I would strongly recommend
starting from the other end. Instead of saying "I have a tool, let's
make a problem for it to solve," let's concentrate on clearly defining
existing problems. Once we agree on the problems, we can talk about
what tools make sense to solve them.<br>
<br>
/a<br>
<br>
On 4/7/10 9:31 AM, Hadriel Kaplan wrote:
<blockquote cite="mid:430FC6BDED356B4C8498F634416644A91A79F3CFE9@mail"
 type="cite">
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  <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
 style="font-size: 10pt; font-family: Arial; color: navy;">Because it
means nothing about the
capabilities for the INVITE, because the INVITE session&#8217;s capabilities
is
dictated by the far-end beyond that gateway. &nbsp;<o:p></o:p></span></font></p>
  <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
 style="font-size: 10pt; font-family: Arial; color: navy;"><o:p>&nbsp;</o:p></span></font></p>
  <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
 style="font-size: 10pt; font-family: Arial; color: navy;">For example,
I don&#8217;t know what types
of capabilities you guys are thinking of, but let&#8217;s say it was for
getting the codecs.&nbsp; So you send an OPTIONS, and it gets to an H.323
gateway. &nbsp;The gateway supports lots of codecs, so it gives you some
nice
big SDP. &nbsp;Then you send an INVITE, thinking it can do some new-fangled
codec.&nbsp; But that INVITE makes an H.323 call get setup, where the H.245
from the far-end only did G.711, so the call fails.<o:p></o:p></span></font></p>
  <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
 style="font-size: 10pt; font-family: Arial; color: navy;"><o:p>&nbsp;</o:p></span></font></p>
  <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
 style="font-size: 10pt; font-family: Arial; color: navy;">Then there
are 3PCC call scenarios, and
middlebox transcoders, and codec-based routing, etc.&nbsp; Different things
happen with INVITES, at different times, due to different policies &#8211;
none
of which apply to OPTIONS.<o:p></o:p></span></font></p>
  <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
 style="font-size: 10pt; font-family: Arial; color: navy;">The reality
of the INVITE world is ugly. </span></font><font color="navy"
 face="Wingdings" size="2"><span
 style="font-size: 10pt; font-family: Wingdings; color: navy;">L</span></font><font
 color="navy" face="Arial" size="2"><span
 style="font-size: 10pt; font-family: Arial; color: navy;"><o:p></o:p></span></font></p>
  <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
 style="font-size: 10pt; font-family: Arial; color: navy;"><o:p>&nbsp;</o:p></span></font></p>
  <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
 style="font-size: 10pt; font-family: Arial; color: navy;">-hadriel<o:p></o:p></span></font></p>
  <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
 style="font-size: 10pt; font-family: Arial; color: navy;"><o:p>&nbsp;</o:p></span></font></p>
  <div
 style="border-style: none none none solid; border-color: -moz-use-text-color -moz-use-text-color -moz-use-text-color blue; border-width: medium medium medium 1.5pt; padding: 0in 0in 0in 4pt;">
  <div>
  <div class="MsoNormal" style="text-align: center;" align="center"><font
 face="Times New Roman" size="3"><span style="font-size: 12pt;">
  <hr tabindex="-1" align="center" size="2" width="100%"></span></font></div>
  <p class="MsoNormal"><b><font face="Tahoma" size="2"><span
 style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font
 face="Tahoma" size="2"><span
 style="font-size: 10pt; font-family: Tahoma;"> Hans Erik van
Elburg [<a class="moz-txt-link-freetext" href="mailto:ietf.hanserik@gmail.com">mailto:ietf.hanserik@gmail.com</a>] <br>
  <b><span style="font-weight: bold;">Sent:</span></b> Wednesday, April
07, 2010
10:05 AM<br>
  <b><span style="font-weight: bold;">To:</span></b> Hadriel Kaplan<br>
  <b><span style="font-weight: bold;">Cc:</span></b> Christer Holmberg;
WORLEY,
DALE R (DALE); Salvatore Loreto; <a class="moz-txt-link-abbreviated" href="mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
  <b><span style="font-weight: bold;">Subject:</span></b> Re: [sipcore]
SIP
OPTIONS: Shall we work on this?</span></font><o:p></o:p></p>
  </div>
  <p class="MsoNormal"><font face="Times New Roman" size="3"><span
 style="font-size: 12pt;"><o:p>&nbsp;</o:p></span></font></p>
  <p class="MsoNormal" style="margin-bottom: 12pt;"><font
 face="Times New Roman" size="3"><span style="font-size: 12pt;">In such
cases you will
get the capabilities of the gateway where the SIP domain is left. Why
isn't
that info useful?<br>
  <br clear="all">
/Hans Erik van Elburg<o:p></o:p></span></font></p>
  <div>
  <p class="MsoNormal"><font face="Times New Roman" size="3"><span
 style="font-size: 12pt;">On Wed, Apr 7, 2010 at 3:53 PM, Hadriel
Kaplan &lt;<a moz-do-not-send="true"
 href="mailto:HKaplan@acmepacket.com">HKaplan@acmepacket.com</a>&gt;
wrote:<o:p></o:p></span></font></p>
  <div>
  <p class="MsoNormal"><font face="Times New Roman" size="3"><span
 style="font-size: 12pt;"><br>
  <br>
&gt; -----Original Message-----<br>
&gt; From: <a moz-do-not-send="true"
 href="mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a>
[mailto:<a moz-do-not-send="true" href="mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a>]
On
Behalf<br>
&gt; Of Christer Holmberg<br>
&gt; Sent: Tuesday, April 06, 2010 4:08 PM<br>
&gt; To: WORLEY, DALE R (DALE); Salvatore Loreto; <a
 moz-do-not-send="true" href="mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
&gt; Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?<br>
&gt;<o:p></o:p></span></font></p>
  </div>
  <div>
  <p class="MsoNormal" style="margin-bottom: 12pt;"><font
 face="Times New Roman" size="3"><span style="font-size: 12pt;">&gt; Hi
Dale,<br>
&gt;<br>
&gt; From my perspective, the main use-case is not to discover
services, but to<br>
&gt; discover capabilities in order to determine what services can be
applied.<br>
&gt;<br>
&gt; The idea is also that the OPTIONS request would be sent prior to
the<br>
&gt; INVITE, to try to make sure that the returned contacts are actually<br>
&gt; available.<o:p></o:p></span></font></p>
  </div>
  <p class="MsoNormal"><font face="Times New Roman" size="3"><span
 style="font-size: 12pt;">But you know that would never work in
practice/the-real-world.
&nbsp;Think about all the routing rules people have for INVITEs - in some
cases
*only* for INVITEs. &nbsp;And even if they route OPTIONS identically in the
SIP
domain, for any given AoR it'll go to any of: a SIP UA, or a PSTN
gateway, or a
H.323-gateway, or an MGCP call-agent, or an IP-PBX, or a
foobar-protocol-gateway, or a vmail server, or an announcement server,
or an
app server, etc. &nbsp;Just because the OPTIONS reaches the end of the SIP
domain, means nothing about whether an INVITE to the higher-layer
target (which
in practice is a E.164 number, that crosses from SIP into many other
protocol
domains) is actually available to receive a call.<br>
  <font color="#888888"><span style="color: rgb(136, 136, 136);"><br>
-hadriel</span></font><o:p></o:p></span></font></p>
  <div>
  <div>
  <p class="MsoNormal"><font face="Times New Roman" size="3"><span
 style="font-size: 12pt;">_______________________________________________<br>
sipcore mailing list<br>
  <a moz-do-not-send="true" href="mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
  <a moz-do-not-send="true"
 href="https://www.ietf.org/mailman/listinfo/sipcore" target="_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><o:p></o:p></span></font></p>
  </div>
  </div>
  </div>
  <p class="MsoNormal"><font face="Times New Roman" size="3"><span
 style="font-size: 12pt;"><o:p>&nbsp;</o:p></span></font></p>
  </div>
  </div>
  <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
sipcore mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sipcore@ietf.org">sipcore@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/sipcore">https://www.ietf.org/mailman/listinfo/sipcore</a>
  </pre>
</blockquote>
<br>
</body>
</html>

--------------030909020002090300030707--

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From: "WORLEY, DALE R (DALE)" <dworley@avaya.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Wed, 7 Apr 2010 10:46:01 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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________________________________________
From: Christer Holmberg [christer.holmberg@ericsson.com]

Now, I am not saying that you need to send OPTIONS prior to every call. My =
main use-case is where the INVITE itself may look different depending on th=
e capabilities. So, again, an INVITE may not even fail, but you may not get=
 exactly what you would have wanted.
________________________________________

If the UAC would send different INVITEs based on its idea of what facilitie=
s are present in the potential UASs, that is a severe failure of the archit=
ecture of the SIP system.  In proper use of SIP, the UAC specifies what it =
prefers in the INVITE, and the intermediate and destination elements determ=
ine how those preferences are to be satisfied based on the destination reso=
urces available at the moment of the INVITE.

If you have a use-case where the UAC cannot specify its needs in a way that=
 conforms to the SIP architecture, there is a need for enhancement of how S=
IP allows the UAC to express its preferences.

Dale

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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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[as an individual]

On 4/7/10 9:43 AM, DRAGE, Keith (Keith) wrote:
> At the moment, my view is "Do nothing". The amount of resource needed 
> for this discussion does not justify the problem.

I'm inclined to agree.

> If someone wants to submit an author draft on a specific issue and 
> fix, then fine, I'll reconsider.

To be fair, someone has.

http://www.ietf.org/id/draft-jones-sip-options-ping-00.txt

I'm not saying I support this document (nor do I necessarily oppose it), 
but there *is* a concrete proposal on the table.

/a

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[as an individual]<br>
<br>
On 4/7/10 9:43 AM, DRAGE, Keith (Keith) wrote:
<blockquote
 cite="mid:EDC0A1AE77C57744B664A310A0B23AE211CCD821@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com"
 type="cite">
  <meta http-equiv="content-type"
 content="text/html; charset=ISO-8859-1">
  <meta content="MSHTML 6.00.2900.3660" name="GENERATOR">
  <div dir="ltr" align="left"><span class="655004214-07042010"><font
 color="#0000ff" face="Arial" size="2">At the moment, my view is "Do
nothing". The amount of resource needed for this discussion does not
justify the problem.</font></span></div>
</blockquote>
<br>
I'm inclined to agree.<br>
<br>
<blockquote
 cite="mid:EDC0A1AE77C57744B664A310A0B23AE211CCD821@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com"
 type="cite">
  <div dir="ltr" align="left"><span class="655004214-07042010"><font
 color="#0000ff" face="Arial" size="2">If someone wants to submit an
author draft on a specific issue and fix, then fine, I'll reconsider.</font></span></div>
</blockquote>
<br>
To be fair, someone has.<br>
<br>
<a class="moz-txt-link-freetext" href="http://www.ietf.org/id/draft-jones-sip-options-ping-00.txt">http://www.ietf.org/id/draft-jones-sip-options-ping-00.txt</a><br>
<br>
I'm not saying I support this document (nor do I necessarily oppose
it), but there *is* a concrete proposal on the table.<br>
<br>
/a<br>
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From: "WORLEY, DALE R (DALE)" <dworley@avaya.com>
To: Adam Roach <adam@nostrum.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Date: Wed, 7 Apr 2010 10:55:39 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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________________________________________
From: Adam Roach [adam@nostrum.com]
[as an individual]

If you're trying to accomplish something, I would strongly recommend starti=
ng from the other end. Instead of saying "I have a tool, let's make a probl=
em for it to solve," let's concentrate on clearly defining existing problem=
s. Once we agree on the problems, we can talk about what tools make sense t=
o solve them.
________________________________

I think that Christer has a definite use situation in mind but hash't been =
very clear about exactly what it is.

IMHO, we need to consider several questions:

1. Is the proposed mechanism well-built?
2. Is the proposed mechanism a good way to achieve the desired functionalit=
y?
3. Is the desired functionality an architecturally sound way to satisfy the=
 users' need?

Now, I think that clarifying how OPTIONS works is in general a good thing, =
even if the answer to questions 2 and 3 are No.  But to take this up as rea=
l WG work, we need to verify that questions 2 and 3 are Yes, before we star=
t worrying about the details of the proposal.

So the first task is to lay out the users' needs that we are trying to addr=
ess, and see if this sort of usage of OPTIONs is a reasonable solution to t=
hem.

Dale

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From: "WORLEY, DALE R (DALE)" <dworley@avaya.com>
To: Hans Erik van Elburg <ietf.hanserik@gmail.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>
Date: Wed, 7 Apr 2010 11:17:22 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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________________________________________
From: Hans Erik van Elburg [ietf.hanserik@gmail.com]

Of course you can implement all kind of policies, but that makes this AOR q=
uite of a special user. I would argue that this is some kind of service AOR=
 and not a "normal" user AOR. The service AOR representing a group of diffe=
rent and dynamically changing users...
________________________________________

But a lot of AORs actually are service AORs and not "normal" users (that re=
present a small set of UAs that is fixed over long periods of time).  Worse=
, "service" AORs account for a much larger fraction of the traffic than the=
ir numbers indicate.  Consider "my company's IT helpdesk number".  You don'=
t want to invent a mechanism that works for 90% of the AORs but only 10% of=
 the calls made.

But also consider any sort of "intelligent network services".  Even moderat=
ely sophisticated call forwarding for a private household can eradicate the=
 concept that an AOR is the front-end for a definite set of UAs.

If you want to get a call to the "right" destination, send the INVITE to th=
e AOR.  Processing incoming INVITEs in a sensible manner is the one SIP fun=
ctionality every intermediate element is likely to get right, because other=
wise users would notice and complain.  And the only alternative is for the =
UAC to attempt to second-guess all the routing processing of every intermed=
iate element, and every UAS, of every AOR.

Dale

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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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[as individual]

Christer Holmberg wrote:
> Hi, 
> 
>>>> Is there *any* problem here? This seems to be working about as well 
>>>> as anyone might hope.
>>>>      
>>> It works okay excluding the non backward compatibility with RFC 3265 (concerning SUBSCRIBE's 2xx response impacts)...
>> I don't think it's accurate to characterize the change as 
>> non-backwards-compatible. The only wire-visible behavior that 
>> one could conceivable observe is that a subscriber might make 
>> a different decision about which subscription to accept if 
>> the SUBSCRIBE forks and the subscriber wants only one subscription.
>>
>> What we *do* gain is a simplification for implementors. 
>> Because the behavior of establishing the dialog when a NOTIFY 
>> shows up (instead of "the first of a 2xx or a NOTIFY) is much 
>> easier to code, people will make fewer mistakes, and have 
>> fewer bugs. But you'd be hard pressed to see a difference on the wire.
> 
> I agree with Adam.
> 
> Of course, for resubscriptions we don't have the forking issue, but I still think we should have the same behavior, ie the NOTIFY acknowledges the resub.
> 
> ...and, again, that is why I have an issue with allowing the suppression of NOTIFYs. It's not a problem for endpoints, but for stateful intermediaries (proxies ARE allowed to Record-Route SUB requests, so)

I think you just explained why NOTIFY should *not* ack the resub.

The response to the reSUBSCRIBE is important, to the transaction, and to 
the subscribe usage status.

Timer-N *could* still apply to the resulting NOTIFY, if one is expected, 
and the subscriber would then need to consider the usage broken if it 
isn't received. But in that case I think it would need to send another 
reSUBSCRIBE with Expires:0 to terminate the subscription - which would 
satisfy any intermediaries that were otherwise confused.

Whether applying timer-N to resubscribes is a *good* thing is not 
immediately clear to me. If the resubscribe is just intended as a 
refresh, then having to time it is an inconvenience. OTOH, if the 
resubscribe is a poll, or a way to change the filter in effect, then the 
immediate notify is probably desired.

One think I noted when checking the bis on this is that the state 
diagram doesn't mention 408 responses. I think it should initiate a 
teardown of the usage.

	Thanks,
	Paul

>>> The draft's Timer N (and lack of related NOTIFY) text is potentially 
>>> worded in a way which does not require the tear down (and non
>>> creation) of subscription.
>> Tear-down of existing dialogs is still an open issue, sure. I 
>> don't get where you see any wiggle room in what happens if 
>> Timer N fires for the first SUBSCRIBE/NOTIFY exchange.
> 
> For the initial SUB/NOT, I think that tearing down the dialog is the only thing you can do.
> 
> Regards,
> 
> Christer
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Your real  world scenario breaks any SIP specific feature... ;-)
Not sure if that was your point.

Still unclear what use case we are discussing here... but OPTIONS does not
necessarily need be used in combination with INVITE.

Just speculating: It could be used by application servers to detect the
capabilities devices a user has registered and do something with that. In
those scenarios maybe a presence server is to much of a burden and there is
no dark interworking nightmare that needs to be taken into account. 3xx
could well be used in this scenario to get all the capabilities of all the
registered devices.

/Hans Erik van Elburg


On Wed, Apr 7, 2010 at 4:31 PM, Hadriel Kaplan <HKaplan@acmepacket.com>wrot=
e:

>  Because it means nothing about the capabilities for the INVITE, because
> the INVITE session=92s capabilities is dictated by the far-end beyond tha=
t
> gateway.
>
>
>
> For example, I don=92t know what types of capabilities you guys are think=
ing
> of, but let=92s say it was for getting the codecs.  So you send an OPTION=
S,
> and it gets to an H.323 gateway.  The gateway supports lots of codecs, so=
 it
> gives you some nice big SDP.  Then you send an INVITE, thinking it can do
> some new-fangled codec.  But that INVITE makes an H.323 call get setup,
> where the H.245 from the far-end only did G.711, so the call fails.
>
>
>
> Then there are 3PCC call scenarios, and middlebox transcoders, and
> codec-based routing, etc.  Different things happen with INVITES, at
> different times, due to different policies =96 none of which apply to OPT=
IONS.
>
> The reality of the INVITE world is ugly. L
>
>
>
> -hadriel
>
>
>   ------------------------------
>
> *From:* Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
> *Sent:* Wednesday, April 07, 2010 10:05 AM
> *To:* Hadriel Kaplan
> *Cc:* Christer Holmberg; WORLEY, DALE R (DALE); Salvatore Loreto;
> sipcore@ietf.org
>
> *Subject:* Re: [sipcore] SIP OPTIONS: Shall we work on this?
>
>
>
> In such cases you will get the capabilities of the gateway where the SIP
> domain is left. Why isn't that info useful?
>
> /Hans Erik van Elburg
>
> On Wed, Apr 7, 2010 at 3:53 PM, Hadriel Kaplan <HKaplan@acmepacket.com>
> wrote:
>
>
>
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf
> > Of Christer Holmberg
> > Sent: Tuesday, April 06, 2010 4:08 PM
> > To: WORLEY, DALE R (DALE); Salvatore Loreto; sipcore@ietf.org
> > Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
> >
>
> > Hi Dale,
> >
> > From my perspective, the main use-case is not to discover services, but
> to
> > discover capabilities in order to determine what services can be applie=
d.
> >
> > The idea is also that the OPTIONS request would be sent prior to the
> > INVITE, to try to make sure that the returned contacts are actually
> > available.
>
> But you know that would never work in practice/the-real-world.  Think abo=
ut
> all the routing rules people have for INVITEs - in some cases *only* for
> INVITEs.  And even if they route OPTIONS identically in the SIP domain, f=
or
> any given AoR it'll go to any of: a SIP UA, or a PSTN gateway, or a
> H.323-gateway, or an MGCP call-agent, or an IP-PBX, or a
> foobar-protocol-gateway, or a vmail server, or an announcement server, or=
 an
> app server, etc.  Just because the OPTIONS reaches the end of the SIP
> domain, means nothing about whether an INVITE to the higher-layer target
> (which in practice is a E.164 number, that crosses from SIP into many oth=
er
> protocol domains) is actually available to receive a call.
>
> -hadriel
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>
>
>

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Your real=A0 world scenario breaks any SIP specific feature... ;-)<br>Not s=
ure if that was your point.<br><br>Still unclear what use case we are discu=
ssing here... but OPTIONS does not necessarily need be used in combination =
with INVITE.<br>
<br>Just speculating: It could be used by application servers to detect the=
 capabilities devices a user has registered and do something with that. In =
those scenarios maybe a presence server is to much of a burden and there is=
 no dark interworking nightmare that needs to be taken into account. 3xx co=
uld well be used in this scenario to get all the capabilities of all the re=
gistered devices.<br>
<br clear=3D"all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Wed, Apr 7, 2010 at 4:31 PM, Hadriel =
Kaplan <span dir=3D"ltr">&lt;<a href=3D"mailto:HKaplan@acmepacket.com">HKap=
lan@acmepacket.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204,=
 204); padding-left: 1ex;">










<div link=3D"blue" vlink=3D"blue" lang=3D"EN-US">

<div>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">Because it mea=
ns nothing about the
capabilities for the INVITE, because the INVITE session=92s capabilities is
dictated by the far-end beyond that gateway. =A0</span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">=A0</span></fo=
nt></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">For example, I=
 don=92t know what types
of capabilities you guys are thinking of, but let=92s say it was for
getting the codecs.=A0 So you send an OPTIONS, and it gets to an H.323
gateway. =A0The gateway supports lots of codecs, so it gives you some nice
big SDP. =A0Then you send an INVITE, thinking it can do some new-fangled
codec.=A0 But that INVITE makes an H.323 call get setup, where the H.245
from the far-end only did G.711, so the call fails.</span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">=A0</span></fo=
nt></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">Then there are=
 3PCC call scenarios, and
middlebox transcoders, and codec-based routing, etc.=A0 Different things
happen with INVITES, at different times, due to different policies =96 none
of which apply to OPTIONS.</span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">The reality of=
 the INVITE world is ugly. </span></font><font color=3D"navy" face=3D"Wingd=
ings" size=3D"2"><span style=3D"font-size: 10pt; font-family: Wingdings; co=
lor: navy;">L</span></font><font color=3D"navy" face=3D"Arial" size=3D"2"><=
span style=3D"font-size: 10pt; font-family: Arial; color: navy;"></span></f=
ont></p>


<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">=A0</span></fo=
nt></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">-hadriel</span=
></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">=A0</span></fo=
nt></p>

<div style=3D"border-width: medium medium medium 1.5pt; border-style: none =
none none solid; border-color: -moz-use-text-color -moz-use-text-color -moz=
-use-text-color blue; padding: 0in 0in 0in 4pt;">

<div>

<div class=3D"MsoNormal" style=3D"text-align: center;" align=3D"center"><fo=
nt face=3D"Times New Roman" size=3D"3"><span style=3D"font-size: 12pt;">

<hr width=3D"100%" align=3D"center" size=3D"2">

</span></font></div>

<p class=3D"MsoNormal"><b><font face=3D"Tahoma" size=3D"2"><span style=3D"f=
ont-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font=
></b><font face=3D"Tahoma" size=3D"2"><span style=3D"font-size: 10pt; font-=
family: Tahoma;"> Hans Erik van
Elburg [mailto:<a href=3D"mailto:ietf.hanserik@gmail.com" target=3D"_blank"=
>ietf.hanserik@gmail.com</a>] <br>
<b><span style=3D"font-weight: bold;">Sent:</span></b> Wednesday, April 07,=
 2010
10:05 AM<br>
<b><span style=3D"font-weight: bold;">To:</span></b> Hadriel Kaplan<br>
<b><span style=3D"font-weight: bold;">Cc:</span></b> Christer Holmberg; WOR=
LEY,
DALE R (DALE); Salvatore Loreto; <a href=3D"mailto:sipcore@ietf.org" target=
=3D"_blank">sipcore@ietf.org</a><div class=3D"im"><br>
<b><span style=3D"font-weight: bold;">Subject:</span></b> Re: [sipcore] SIP
OPTIONS: Shall we work on this?</div></span></font></p>

</div><div class=3D"im">

<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">=A0</span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;"><font face=3D"Times N=
ew Roman" size=3D"3"><span style=3D"font-size: 12pt;">In such cases you wil=
l
get the capabilities of the gateway where the SIP domain is left. Why isn&#=
39;t
that info useful?<br>
<br clear=3D"all">
/Hans Erik van Elburg</span></font></p>

<div>

<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">On Wed, Apr 7, 2010 at 3:53 PM, Hadriel Kaplan &lt;<=
a href=3D"mailto:HKaplan@acmepacket.com" target=3D"_blank">HKaplan@acmepack=
et.com</a>&gt; wrote:</span></font></p>


<div>

<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;"><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sipcore-bounces@ietf.org" target=3D"_blank">si=
pcore-bounces@ietf.org</a>
[mailto:<a href=3D"mailto:sipcore-bounces@ietf.org" target=3D"_blank">sipco=
re-bounces@ietf.org</a>]
On Behalf<br>
&gt; Of Christer Holmberg<br>
&gt; Sent: Tuesday, April 06, 2010 4:08 PM<br>
&gt; To: WORLEY, DALE R (DALE); Salvatore Loreto; <a href=3D"mailto:sipcore=
@ietf.org" target=3D"_blank">sipcore@ietf.org</a><br>
&gt; Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?<br>
&gt;</span></font></p>

</div>

<div>

<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;"><font face=3D"Times N=
ew Roman" size=3D"3"><span style=3D"font-size: 12pt;">&gt; Hi Dale,<br>
&gt;<br>
&gt; From my perspective, the main use-case is not to discover services, bu=
t to<br>
&gt; discover capabilities in order to determine what services can be appli=
ed.<br>
&gt;<br>
&gt; The idea is also that the OPTIONS request would be sent prior to the<b=
r>
&gt; INVITE, to try to make sure that the returned contacts are actually<br=
>
&gt; available.</span></font></p>

</div>

<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">But you know that would never work in practice/the-r=
eal-world.
=A0Think about all the routing rules people have for INVITEs - in some case=
s
*only* for INVITEs. =A0And even if they route OPTIONS identically in the SI=
P
domain, for any given AoR it&#39;ll go to any of: a SIP UA, or a PSTN gatew=
ay, or a
H.323-gateway, or an MGCP call-agent, or an IP-PBX, or a
foobar-protocol-gateway, or a vmail server, or an announcement server, or a=
n
app server, etc. =A0Just because the OPTIONS reaches the end of the SIP
domain, means nothing about whether an INVITE to the higher-layer target (w=
hich
in practice is a E.164 number, that crosses from SIP into many other protoc=
ol
domains) is actually available to receive a call.<br>
<font color=3D"#888888"><span style=3D"color: rgb(136, 136, 136);"><br>
-hadriel</span></font></span></font></p>

<div>

<div>

<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a></span></font></p>

</div>

</div>

</div>

<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">=A0</span></font></p>

</div></div>

</div>

</div>


</blockquote></div><br>

--001485ea7dca3a4ec10483a73834--

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To: Christer Holmberg <christer.holmberg@ericsson.com>, Adam Roach <adam@nostrum.com>, Brett Tate <brett@broadsoft.com>
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________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Chri=
ster Holmberg [christer.holmberg@ericsson.com]

Of course, for resubscriptions we don't have the forking issue, but I still=
 think we should have the same behavior, ie the NOTIFY acknowledges the res=
ub.
_______________________________________________

rfc3265bis isn't particularly clear on the point, but it looks like the 200=
 response to the re-SUBSCRIBE extends the subscription expiration time know=
n by the subscriber. So an intermediate element that tracks the subscriptio=
n has to be watching for such responses, too.  Given that, the fact that su=
b-not suppresses the NOTIFY is not a problem.

Dale

From christer.holmberg@ericsson.com  Wed Apr  7 10:22:45 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 7 Apr 2010 19:21:44 +0200
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Hi,

>>>>> Is there *any* problem here? This seems to be working about as well
>>>>> as anyone might hope.
>>>>>
>>>> It works okay excluding the non backward compatibility with RFC 3265 (=
concerning SUBSCRIBE's 2xx response impacts)...
>>> I don't think it's accurate to characterize the change as
>>> non-backwards-compatible. The only wire-visible behavior that
>>> one could conceivable observe is that a subscriber might make
>>> a different decision about which subscription to accept if
>>> the SUBSCRIBE forks and the subscriber wants only one subscription.
>>>
>>> What we *do* gain is a simplification for implementors.
>>> Because the behavior of establishing the dialog when a NOTIFY
>>> shows up (instead of "the first of a 2xx or a NOTIFY) is much
>>> easier to code, people will make fewer mistakes, and have
>>> fewer bugs. But you'd be hard pressed to see a difference on the wire.
>>
>> I agree with Adam.
>>
>> Of course, for resubscriptions we don't have the forking issue, but I st=
ill think we should have the same behavior, ie the NOTIFY acknowledges the =
resub.
>>
>> ...and, again, that is why I have an issue with allowing the suppression=
 of NOTIFYs. It's not a problem for endpoints, but for stateful intermediar=
ies (proxies ARE allowed to Record-Route SUB requests, so)
>
>I think you just explained why NOTIFY should *not* ack the resub.

Sure, if the SUB 200 response acks the resub, there should be no problem.

I just assumed that the NOT request (or response) would ack also resubs, to=
 make the state machine consistant with initial subs. But, if that is not t=
he case, we need to make it VERY clear in 3265bis.

>The response to the reSUBSCRIBE is important, to the transaction, and to t=
he subscribe usage status.
>
>Timer-N *could* still apply to the resulting NOTIFY, if one is expected,
>and the subscriber would then need to consider the usage broken if it
>isn't received. But in that case I think it would need to send another
>reSUBSCRIBE with Expires:0 to terminate the subscription - which would
>satisfy any intermediaries that were otherwise confused.

That could work.

>Whether applying timer-N to resubscribes is a *good* thing is not
>immediately clear to me. If the resubscribe is just intended as a
>refresh, then having to time it is an inconvenience. OTOH, if the
>resubscribe is a poll, or a way to change the filter in effect, then the
>immediate notify is probably desired.

I guess it would be good to be able to indicate whether an immediate notify=
 is required, in such indicator could start timer-N. But, all entities woul=
d of course have to understand such indicator.

Regards,

Christer



One think I noted when checking the bis on this is that the state
diagram doesn't mention 408 responses. I think it should initiate a
teardown of the usage.

        Thanks,
        Paul

>>> The draft's Timer N (and lack of related NOTIFY) text is potentially
>>> worded in a way which does not require the tear down (and non
>>> creation) of subscription.
>> Tear-down of existing dialogs is still an open issue, sure. I
>> don't get where you see any wiggle room in what happens if
>> Timer N fires for the first SUBSCRIBE/NOTIFY exchange.
>
> For the initial SUB/NOT, I think that tearing down the dialog is the only=
 thing you can do.
>
> Regards,
>
> Christer
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=

From christer.holmberg@ericsson.com  Wed Apr  7 10:24:56 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Adam Roach <adam@nostrum.com>, Brett Tate <brett@broadsoft.com>
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Hi,

>>Of course, for resubscriptions we don't have the forking issue, but I sti=
ll think we should have the same >>behavior, ie the NOTIFY acknowledges the=
 resub.
>>
>rfc3265bis isn't particularly clear on the point, but it looks like the 20=
0 response to the re-SUBSCRIBE >extends the subscription expiration time kn=
own by the subscriber. So an intermediate element that tracks the >subscrip=
tion has to be watching for such responses, too.  Given that, the fact that=
 sub-not suppresses the >NOTIFY is not a problem.

Agree.

So, if we choose that way forward, we need to make sure it is explicitly in=
dicated in 3265bis, or I am sure we will sooner or later end up with implem=
entation issues.

Regards,

Christer=

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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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[as individual]

Paul Kyzivat wrote:
> One think I noted when checking the bis on this is that the state 
> diagram doesn't mention 408 responses. I think it should initiate a 
> teardown of the usage.

408 responses aren't allowed for non-invite transactions.

/a

From pkyzivat@cisco.com  Wed Apr  7 11:29:26 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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[as individual]

Adam Roach wrote:
> [as individual]
> 
> Paul Kyzivat wrote:
>> One think I noted when checking the bis on this is that the state 
>> diagram doesn't mention 408 responses. I think it should initiate a 
>> teardown of the usage.
> 
> 408 responses aren't allowed for non-invite transactions.

Huh???

Of course they are. Any request can time out.

	Thanks,
	Paul

From brett@broadsoft.com  Wed Apr  7 11:34:28 2010
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From: Brett Tate <brett@broadsoft.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 7 Apr 2010 11:32:58 -0700
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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I assume that Adam is referring to RFC 4320.  However as discussed within t=
he following thread, the 408's can still be received.

https://lists.cs.columbia.edu/pipermail/sip-implementors/2010-April/024774.=
html=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Wednesday, April 07, 2010 2:29 PM
> To: Adam Roach
> Cc: Christer Holmberg; Brett Tate; SIPCORE
> Subject: Re: [sipcore] rfc3265bis: SIP events redux [was Minutes
> Posted]
>=20
> [as individual]
>=20
> Adam Roach wrote:
> > [as individual]
> >
> > Paul Kyzivat wrote:
> >> One think I noted when checking the bis on this is that the state
> >> diagram doesn't mention 408 responses. I think it should initiate a
> >> teardown of the usage.
> >
> > 408 responses aren't allowed for non-invite transactions.
>=20
> Huh???
>=20
> Of course they are. Any request can time out.
>=20
> 	Thanks,
> 	Paul

From adam@nostrum.com  Wed Apr  7 11:37:00 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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On 4/7/10 1:29 PM, Paul Kyzivat wrote:
> [as individual]
>
> Adam Roach wrote:
>> [as individual]
>>
>> Paul Kyzivat wrote:
>>> One think I noted when checking the bis on this is that the state 
>>> diagram doesn't mention 408 responses. I think it should initiate a 
>>> teardown of the usage.
>>
>> 408 responses aren't allowed for non-invite transactions.
>
> Huh???
>
> Of course they are. Any request can time out.

RFC 4320, section 4.2: "A transaction-stateful SIP element MUST NOT send 
a response with Status-Code of 408 to a non-INVITE request." The 
rationale for this normative update to 3261 is given in section 1.4 of 
RFC 4321.

/a

From pkyzivat@cisco.com  Wed Apr  7 11:52:19 2010
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[as individual]

Sorry. Perhaps I was being imprecise. I'm used to talking about 
receiving a 408 as synonymous with "receiving a 408 or having the 
transaction time out". Some stacks will return a 408 as a result of 
timeout even though one was never received.

The point remains - if some transaction in the usage times out, and the 
presumption is that the usage is then dead, I think its expected that 
there will still be signaling to indicate its being torn down, isn't 
there? Assuming this happens to the subscriber, shouldn't it send a 
reSUBSCRIBE with Expires:0 to explicitly terminate it. (Just as a BYE 
would be sent if this happened within an INVITE usage.)

	Thanks,
	Paul

Adam Roach wrote:
> On 4/7/10 1:29 PM, Paul Kyzivat wrote:
>> [as individual]
>>
>> Adam Roach wrote:
>>> [as individual]
>>>
>>> Paul Kyzivat wrote:
>>>> One think I noted when checking the bis on this is that the state 
>>>> diagram doesn't mention 408 responses. I think it should initiate a 
>>>> teardown of the usage.
>>>
>>> 408 responses aren't allowed for non-invite transactions.
>>
>> Huh???
>>
>> Of course they are. Any request can time out.
> 
> RFC 4320, section 4.2: "A transaction-stateful SIP element MUST NOT send 
> a response with Status-Code of 408 to a non-INVITE request." The 
> rationale for this normative update to 3261 is given in section 1.4 of 
> RFC 4321.
> 
> /a
> 

From dean.willis@softarmor.com  Wed Apr  7 14:19:28 2010
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On Apr 7, 2010, at 6:35 AM, Christer Holmberg wrote:
>
> If we don't like OPTIONS, let's deprecate it... Having methods and  
> functions, but at the same time saying that it doesn't work and we  
> therefor shouldn't use it, is very confusing...
>

As I'm no longer the SIP chair, I think I can say that I feel the same  
way about forking. If we deprecate that, we fix the forking OPTIONS  
problem at the same time.


Seriously, OPTIONS+forking is a bad combination. If we don't formally  
deprecate forking, at least we can recommend it not be used for  
OPTIONS requests. Or we could amend 3261 to say "proxies MUST NOT fork  
OPTIONS request messages". Or we could add an options tag, and say  
"Proxies that support this extension MUST NOT fork OPTIONS".

Or we could go back to our comfortable head-in-sand positions. I'm OK  
with that; I just pull mine out occasionally to laugh at the other  
ostrich-bottoms on display.

--
Dean

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Subject: Re: [sipcore] draft-ietf-sipcore-subnot-etags: When to include SIP-Etag in NOTIFY?
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On Apr 7, 2010, at 7:59 AM, Christer Holmberg wrote:

>
>
> So, the question is: is a subnot-etags compliant notifier required  
> to insert the SIP-Etag in *every* NOT request, or only in the cases  
> explicitly described in Section 6?


I believe the answer is "always".

I also believe I just saw an Auth 48 request come out on this draft.  
Anybody else got a bug to fix?

--
Dean

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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On Apr 7, 2010, at 4:19 PM, Dean Willis wrote:

>
> Seriously, OPTIONS+forking is a bad combination. If we don't  
> formally deprecate forking, at least we can recommend it not be used  
> for OPTIONS requests. Or we could amend 3261 to say "proxies MUST  
> NOT fork OPTIONS request messages". Or we could add an options tag,  
> and say "Proxies that support this extension MUST NOT fork OPTIONS".
>

I left out OR we require that a proxy that would ordinarily fork the  
request instead act as a B2BUA and return an aggregate response.

--
Dean

From christer.holmberg@ericsson.com  Wed Apr  7 14:31:07 2010
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Date: Wed, 7 Apr 2010 23:26:11 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

>Seriously, OPTIONS+forking is a bad combination. If we don't
>formally deprecate forking, at least we can recommend it not be used
>for OPTIONS requests. Or we could amend 3261 to say "proxies MUST
>NOT fork OPTIONS request messages". Or we could add an options tag,
>and say "Proxies that support this extension MUST NOT fork OPTIONS".

Using the O-3xx (I'm getting tired of writing OPTIONS 3xx :) mechanism the =
OPTIONS request isn't forked.

>I left out OR we require that a proxy that would ordinarily fork the reque=
st instead act as a B2BUA and=20
>return an aggregate response.

That is of course one alternative, but the "beauty" of O-3xx is that you do=
n't need to standardize anything. The tools are available (yes, I know we c=
an't be 100% sure that it will always work), so the question is whether we =
somehow want to document how we can solve the forking problem by combining =
those tools.

Regards,

Christer=

From christer.holmberg@ericsson.com  Wed Apr  7 14:33:33 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Wed, 7 Apr 2010 23:31:46 +0200
Thread-Topic: [sipcore] draft-ietf-sipcore-subnot-etags: When to include SIP-Etag in NOTIFY?
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Subject: Re: [sipcore] draft-ietf-sipcore-subnot-etags: When to include SIP-Etag in NOTIFY?
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Hi,

>> So, the question is: is a subnot-etags compliant notifier required
>> to insert the SIP-Etag in *every* NOT request, or only in the cases
>> explicitly described in Section 6?
>
>
>I believe the answer is "always".
>
>I also believe I just saw an Auth 48 request come out on this draft.
>Anybody else got a bug to fix?

In 3265bis we are aiming at getting rid of the 202 response. The subnot dra=
ft defines another one, 204, but I guess it doesn't matter since only endpo=
ints supporting the draft are going to send and receive it.

Regards,

Christer=

From adam@nostrum.com  Wed Apr  7 14:40:34 2010
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-ietf-sipcore-subnot-etags: When to include SIP-Etag in NOTIFY?
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On 4/7/10 4:31 PM, Christer Holmberg wrote:
> Hi,
>
>    
>>> So, the question is: is a subnot-etags compliant notifier required
>>> to insert the SIP-Etag in *every* NOT request, or only in the cases
>>> explicitly described in Section 6?
>>>        
>>
>> I believe the answer is "always".
>>
>> I also believe I just saw an Auth 48 request come out on this draft.
>> Anybody else got a bug to fix?
>>      
> In 3265bis we are aiming at getting rid of the 202 response. The subnot draft defines another one, 204, but I guess it doesn't matter since only endpoints supporting the draft are going to send and receive it.
>    

We're getting rid of 202 because it is vestigial, meaningless, and often 
implemented incorrectly.

subnot-etag's use of 204 is critical for proper functioning of its 
mechanism.

/a

From HKaplan@acmepacket.com  Wed Apr  7 15:20:55 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Wed, 7 Apr 2010 18:20:39 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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________________________________
From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
Sent: Wednesday, April 07, 2010 11:29 AM
To: Hadriel Kaplan
Cc: Christer Holmberg; WORLEY, DALE R (DALE); Salvatore Loreto; sipcore@iet=
f.org
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?

Your real  world scenario breaks any SIP specific feature... ;-)
[hsk] Yup, and that's the real world of inter-domain routing right now. :(
Still unclear what use case we are discussing here... but OPTIONS does not =
necessarily need be used in combination with INVITE.

[hsk] we're only discussing that because it was one example Christer descri=
bed for using he OPTIONS fork/3xx need.

Just speculating: It could be used by application servers to detect the cap=
abilities devices a user has registered and do something with that. In thos=
e scenarios maybe a presence server is to much of a burden and there is no =
dark interworking nightmare that needs to be taken into account. 3xx could =
well be used in this scenario to get all the capabilities of all the regist=
ered devices.

[hsk] That was the exact question I asked previously on this topic - whethe=
r that App-server was the use-case in mind, because for that the problem is=
 much simpler, and putting a new options-tag in a proxy-require is not only=
 likely to work, but also reasonably easy to deploy.  As long as you constr=
ain the use-case to be within the local domain, then doing things like prox=
y-require or even manual configuration and such is all reasonable, imho.  I=
f you're talking about using it across domains instead, between any UAC to =
any UAS in the World/Internet, this thing is about as likely to fail as Tig=
er Woods' marriage.

/Hans Erik van Elburg



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=3D'font-size:
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4.0pt'>

<div>

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<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Hans Eri=
k van
Elburg [mailto:ietf.hanserik@gmail.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, April 07, 2=
010
11:29 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Hadriel Kaplan<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> Christer Holmberg; WORLE=
Y,
DALE R (DALE); Salvatore Loreto; sipcore@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [sipcore] SIP
OPTIONS: Shall we work on this?</span></font><o:p></o:p></p>

</div>

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

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>Your real&nbsp; w=
orld
scenario breaks any SIP specific feature... ;-)<font color=3Dnavy><span
style=3D'color:navy'><o:p></o:p></span></font></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 color=3D=
navy
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:navy'>=
[hsk] Yup,
and that&#8217;s the real world of inter-domain routing right now. </span><=
/font><font
size=3D2 color=3Dnavy face=3DWingdings><span style=3D'font-size:10.0pt;font=
-family:
Wingdings;color:navy'>L</span></font><font size=3D2 color=3Dnavy face=3DAri=
al><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p></o:p></span><=
/font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>Still unclear wha=
t use
case we are discussing here... but OPTIONS does not necessarily need be use=
d in
combination with INVITE.<br>
<br>
<font color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></sp=
an></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 color=3D=
navy
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:navy'>=
[hsk] we&#8217;re
only discussing that because it was one example Christer described for usin=
g he
OPTIONS fork/3xx need.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'><br>
Just speculating: It could be used by application servers to detect the
capabilities devices a user has registered and do something with that. In t=
hose
scenarios maybe a presence server is to much of a burden and there is no da=
rk
interworking nightmare that needs to be taken into account. 3xx could well =
be
used in this scenario to get all the capabilities of all the registered
devices.<br>
<br>
<font color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></sp=
an></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 color=3D=
navy
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial;color:navy'>=
[hsk]
That was the exact question I asked previously on this topic &#8211; whethe=
r that
App-server was the use-case in mind, because for that the problem is much
simpler, and putting a new options-tag in a proxy-require is not only likel=
y to
work, but also reasonably easy to deploy. &nbsp;As long as you constrain th=
e
use-case to be within the local domain, then doing things like proxy-requir=
e or
even manual configuration and such is all reasonable, imho.&nbsp; If you&#8=
217;re
talking about using it across domains instead, between any UAC to any UAS i=
n
the World/Internet, this thing is about as likely to fail as Tiger Woods&#8=
217;
marriage. <o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:0in;margin-right:0in;margi=
n-bottom:
12.0pt;margin-left:5.25pt'><font size=3D3 face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'><br clear=3Dall>
/Hans Erik van Elburg<br>
<br>
<o:p></o:p></span></font></p>

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

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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>, Peter Musgrave <peter.musgrave@magorcorp.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On Apr 7, 2010, at 4:26 PM, Christer Holmberg wrote:

> Hi,
>
>> Seriously, OPTIONS+forking is a bad combination. If we don't
>> formally deprecate forking, at least we can recommend it not be used
>> for OPTIONS requests. Or we could amend 3261 to say "proxies MUST
>> NOT fork OPTIONS request messages". Or we could add an options tag,
>> and say "Proxies that support this extension MUST NOT fork OPTIONS".
>
> Using the O-3xx (I'm getting tired of writing OPTIONS 3xx :)  
> mechanism the OPTIONS request isn't forked.

Right. Saying proxies MUST NOT fork OPTIONS does not preclude sending  
a 302 response. Or a new error code saying "Can't do that, too many  
targets", or anything else.

>
>> I left out OR we require that a proxy that would ordinarily fork  
>> the request instead act as a B2BUA and
>> return an aggregate response.
>
> That is of course one alternative, but the "beauty" of O-3xx is that  
> you don't need to standardize anything. The tools are available  
> (yes, I know we can't be 100% sure that it will always work), so the  
> question is whether we somehow want to document how we can solve the  
> forking problem by combining those tools.


Assuming your application doesn't have to be absolutely sure that the  
OPTIONS isn't forking, then you're right; this could be done entirely  
on an informational basis. Might even be reasonable as an individual  
informational: "An Alternative to Forking for OPTIONS Requests to  
Enable the Discovery of User-Agent-Server Capabilities in the Session  
Initiation Protocol." Sounds like a good thesis project for a telecom  
student or senior project for an undergrad. Probably doesn't need to  
be a working-group effort.

--
Dean

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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Adam Roach <adam@nostrum.com>
Date: Thu, 8 Apr 2010 03:16:16 +0200
Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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While I accept that RFC 4320 is mandatory for all current implementations c=
alling themselves SIP, we do need to deal with compatibility issues with de=
ployments made before it was thought of and became an RFC.

Thus if an RFC 3261/3265 sans RFC 4320 can send a 408 response, then any ot=
her implementation needs to be able to receive it.

Unless of course you can argue that any entity sending it in the first plac=
e had to be so broken that it is otherwise inoperable.=20

regards

Keith

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Wednesday, April 07, 2010 7:52 PM
> To: Adam Roach
> Cc: SIPCORE
> Subject: Re: [sipcore] rfc3265bis: SIP events redux [was=20
> Minutes Posted]
>=20
> [as individual]
>=20
> Sorry. Perhaps I was being imprecise. I'm used to talking=20
> about receiving a 408 as synonymous with "receiving a 408 or=20
> having the transaction time out". Some stacks will return a=20
> 408 as a result of timeout even though one was never received.
>=20
> The point remains - if some transaction in the usage times=20
> out, and the presumption is that the usage is then dead, I=20
> think its expected that there will still be signaling to=20
> indicate its being torn down, isn't there? Assuming this=20
> happens to the subscriber, shouldn't it send a reSUBSCRIBE=20
> with Expires:0 to explicitly terminate it. (Just as a BYE=20
> would be sent if this happened within an INVITE usage.)
>=20
> 	Thanks,
> 	Paul
>=20
> Adam Roach wrote:
> > On 4/7/10 1:29 PM, Paul Kyzivat wrote:
> >> [as individual]
> >>
> >> Adam Roach wrote:
> >>> [as individual]
> >>>
> >>> Paul Kyzivat wrote:
> >>>> One think I noted when checking the bis on this is that=20
> the state=20
> >>>> diagram doesn't mention 408 responses. I think it should=20
> initiate a=20
> >>>> teardown of the usage.
> >>>
> >>> 408 responses aren't allowed for non-invite transactions.
> >>
> >> Huh???
> >>
> >> Of course they are. Any request can time out.
> >=20
> > RFC 4320, section 4.2: "A transaction-stateful SIP element MUST NOT=20
> > send a response with Status-Code of 408 to a non-INVITE=20
> request." The=20
> > rationale for this normative update to 3261 is given in=20
> section 1.4 of=20
> > RFC 4321.
> >=20
> > /a
> >=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From adam@nostrum.com  Wed Apr  7 20:36:52 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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On 4/7/10 20:16, Apr 7, DRAGE, Keith (Keith) wrote:
> While I accept that RFC 4320 is mandatory for all current implementations calling themselves SIP, we do need to deal with compatibility issues with deployments made before it was thought of and became an RFC.
>
> Thus if an RFC 3261/3265 sans RFC 4320 can send a 408 response, then any other implementation needs to be able to receive it.
>    

Be able to receive it, yes. But, as RFC 4321 explains, the local 
transaction timer will have already expired. The 408 will be treated as 
a stray response. A well-designed SIP stack will discard it before it 
even reaches the TU. Even a poorly-designed but 3261-compliant 
application will ignore a 408 for a non-INVITE, because the transaction 
is already gone.

In other words, by the time the 408 arrives, it is thoroughly 
irrelevant. We don't need to talk about it in 3265bis, because something 
else will have gone wrong first.

And that situation is already covered:

    If the NOTIFY request fails due to expiration of SIP Timer F
    (transaction timeout), the notifier SHOULD remove the subscription.


/a

--------------030803050608080403080101
Content-Type: text/html; charset=ISO-8859-1
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
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<head>
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 http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
On 4/7/10 20:16, Apr 7, DRAGE, Keith (Keith) wrote:
<blockquote
 cite="mid:EDC0A1AE77C57744B664A310A0B23AE211CCD8B2@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com"
 type="cite">
  <pre wrap="">While I accept that RFC 4320 is mandatory for all current implementations calling themselves SIP, we do need to deal with compatibility issues with deployments made before it was thought of and became an RFC.

Thus if an RFC 3261/3265 sans RFC 4320 can send a 408 response, then any other implementation needs to be able to receive it.
  </pre>
</blockquote>
<br>
Be able to receive it, yes. But, as RFC 4321 explains, the local
transaction timer will have already expired. The 408 will be treated as
a stray response. A well-designed SIP stack will discard it before it
even reaches the TU. Even a poorly-designed but 3261-compliant
application will ignore a 408 for a non-INVITE, because the transaction
is already gone.<br>
<br>
In other words, by the time the 408 arrives, it is thoroughly
irrelevant. We don't need to talk about it in 3265bis, because
something else will have gone wrong first.<br>
<br>
And that situation is already covered:<br>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
<pre class="newpage">   If the NOTIFY request fails due to expiration of SIP Timer F
   (transaction timeout), the notifier SHOULD remove the subscription.
</pre>
<br>
/a<br>
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Hi,
=20
In Section 14.2 UAS Behaviour, we have requirement stating as below
=20
   A UAS that receives a second INVITE before it sends the final
   response to a first INVITE with a lower CSeq sequence number on the
   same dialog MUST return a 500 (Server Internal Error) response to the
   second INVITE and MUST include a Retry-After header field with a
   randomly chosen value of between 0 and 10 seconds.
=20
Above requirement is valid when we receive INVITE, before sending 2xx
response... we are responding with 500 in our implementation...
=20
=20
What's the UAS behaviour ?
=20
If we receive INVITE, after sending final response, before getting ACK.=20
=20
=20
=20
=20

=20
Regards,
Naveen
=20

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<META http-equiv=3DContent-Type content=3D"text/html; =
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<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010>Hi,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D725454211-07042010>In =
Section 14.2 UAS=20
Behaviour, we have requirement stating as below</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D725454211-07042010><STRONG>&nbsp;&nbsp;=20
A UAS that receives a second INVITE before it sends the =
final<BR>&nbsp;&nbsp;=20
response to a first INVITE with a lower CSeq sequence number on=20
the<BR>&nbsp;&nbsp; same dialog MUST return a 500 (Server Internal =
Error)=20
response to the<BR>&nbsp;&nbsp; second INVITE and MUST include a =
Retry-After=20
header field with a<BR>&nbsp;&nbsp; randomly chosen value of between 0 =
and 10=20
seconds.</STRONG></SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010><STRONG></STRONG></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D725454211-07042010>Above =
requirement is=20
valid when we receive INVITE, before sending 2xx response... we are =
responding=20
with 500 in our implementation...</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D725454211-07042010></SPAN></FONT><FONT=20
face=3DArial size=3D2><SPAN =
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D725454211-07042010>What's =
the UAS=20
behaviour ?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#ff0000 size=3D2><SPAN =
class=3D725454211-07042010>If we=20
receive INVITE, after sending final response, before getting ACK.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D725454211-07042010><IMG=20
src=3D"cid:725454211@07042010-3500"></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG>&nbsp;</DIV>
<DIV><BR></DIV></STRONG></SPAN></FONT>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV align=3Dleft><FONT face=3DCalibri>Regards,</FONT></DIV>
<DIV align=3Dleft><FONT face=3DCalibri>Naveen</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

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Thread-Topic: [sipcore] Need Clarification on UAS behaviour
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "NAVEEN S-RHVG46" <naveen.shivanna@motorola.com>, <sipcore@ietf.org>
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Hi Naveen
=20
The UAS should send 500 response to the INVITE - because the earlier
INVITE transaction is not yet complete. .
=20
Also I feel this question is more appropriate in sip-implementors list
rather than sipcore

Regards=20
Ranjit=20

=20

________________________________

From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
Behalf Of NAVEEN S-RHVG46
Sent: Thursday, April 08, 2010 10:17 AM
To: sipcore@ietf.org
Subject: [sipcore] Need Clarification on UAS behaviour


Hi,
=20
In Section 14.2 UAS Behaviour, we have requirement stating as below
=20
   A UAS that receives a second INVITE before it sends the final
   response to a first INVITE with a lower CSeq sequence number on the
   same dialog MUST return a 500 (Server Internal Error) response to the
   second INVITE and MUST include a Retry-After header field with a
   randomly chosen value of between 0 and 10 seconds.
=20
Above requirement is valid when we receive INVITE, before sending 2xx
response... we are responding with 500 in our implementation...
=20
=20
What's the UAS behaviour ?
=20
If we receive INVITE, after sending final response, before getting ACK.=20
=20
=20
=20
=20

=20
Regards,
Naveen
=20

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<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D017162905-08042010><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Hi Naveen</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D017162905-08042010><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D017162905-08042010><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>The UAS should send 500 response to the INVITE =
- because=20
the earlier INVITE transaction is not yet complete. =
.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D017162905-08042010><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2>Also I feel this =
question is more=20
appropriate in sip-implementors list rather than =
sipcore</FONT></DIV><!-- Converted from text/rtf format -->
<P><SPAN lang=3Den-us><FONT face=3DArial size=3D2>Regards</FONT></SPAN> =
<BR><SPAN=20
lang=3Den-us><FONT face=3DArial size=3D2>Ranjit</FONT></SPAN> </P>
<DIV>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sipcore-bounces@ietf.org=20
[mailto:sipcore-bounces@ietf.org] <B>On Behalf Of </B>NAVEEN=20
S-RHVG46<BR><B>Sent:</B> Thursday, April 08, 2010 10:17 AM<BR><B>To:</B> =

sipcore@ietf.org<BR><B>Subject:</B> [sipcore] Need Clarification on UAS=20
behaviour<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010>Hi,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D725454211-07042010>In =
Section 14.2 UAS=20
Behaviour, we have requirement stating as below</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D725454211-07042010><STRONG>&nbsp;&nbsp;=20
A UAS that receives a second INVITE before it sends the =
final<BR>&nbsp;&nbsp;=20
response to a first INVITE with a lower CSeq sequence number on=20
the<BR>&nbsp;&nbsp; same dialog MUST return a 500 (Server Internal =
Error)=20
response to the<BR>&nbsp;&nbsp; second INVITE and MUST include a =
Retry-After=20
header field with a<BR>&nbsp;&nbsp; randomly chosen value of between 0 =
and 10=20
seconds.</STRONG></SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010><STRONG></STRONG></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D725454211-07042010>Above =
requirement is=20
valid when we receive INVITE, before sending 2xx response... we are =
responding=20
with 500 in our implementation...</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D725454211-07042010></SPAN></FONT><FONT=20
face=3DArial size=3D2><SPAN =
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D725454211-07042010>What's =
the UAS=20
behaviour ?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#ff0000 size=3D2><SPAN =
class=3D725454211-07042010>If we=20
receive INVITE, after sending final response, before getting ACK.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D725454211-07042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D725454211-07042010><IMG=20
src=3D"cid:017162905@08042010-1536"></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG>&nbsp;</DIV>
<DIV><BR></DIV></STRONG></SPAN></FONT>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV align=3Dleft><FONT face=3DCalibri>Regards,</FONT></DIV>
<DIV align=3Dleft><FONT face=3DCalibri>Naveen</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

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------_=_NextPart_001_01CAD6DC.9EF8C280--

From christer.holmberg@ericsson.com  Wed Apr  7 23:46:27 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 8 Apr 2010 08:46:20 +0200
Thread-Topic: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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Hi,

In the case of DE-subscriptions (which is just one variant of re-subscripti=
on), I still think there are some issues regarding whether the SUB 200 or t=
he NOT req acknowledges the SUB req.

Assuming the sub dialog is terminated when an entity receives a SUB 200 tha=
t contains 'Expires: 0'.

Now, if there is then a NOT request (that contains 'Subscription-State: ter=
minated'), won't it be treated as an out-of-dialog request, and rejected?

(Again, IF one is supposed to wait for the NOT req before terminating the d=
ialog we again have a problem with the subnot draft.)

Regards,

Christer





=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: 7. huhtikuuta 2010 9:21
> To: Adam Roach; Brett Tate
> Cc: SIPCORE
> Subject: [sipcore] Rfc3265bis: NOTIFY request or response=20
> confirms (re-)subscription? [was rfc3265bis: SIP events redux=20
> [was Minutes Posted]]
>=20
>=20
> Hi,
>=20
> I haven't double checked whether it is said in 3265bis, but=20
> we also need to make clear whether it is the NOTIFY request,=20
> or the NOT 200 response, that "acknowledges" the (re-)subscription.
>=20
> Because, a NOTIFY request can also be rejected (e.g. due to=20
> overload), so in order to avoid sub state unsynch every=20
> entity in the path needs to know what a NOT non-200 response means.
>=20
> Regards,
>=20
> Christer
>=20
>=20
> =20
>=20
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> > Sent: 7. huhtikuuta 2010 2:37
> > To: Brett Tate
> > Cc: SIPCORE
> > Subject: Re: [sipcore] rfc3265bis: SIP events redux [was Minutes=20
> > Posted]
> >=20
> > On 4/6/10 17:46, Apr 6, Brett Tate wrote:
> > >> Is there *any* problem here? This seems to be working
> > about as well
> > >> as anyone might hope.
> > >>     =20
> > > It works okay excluding the non backward compatibility with
> > RFC 3265 (concerning SUBSCRIBE's 2xx response impacts)...
> >=20
> > I don't think it's accurate to characterize the change as=20
> > non-backwards-compatible. The only wire-visible behavior that one=20
> > could conceivable observe is that a subscriber might make a=20
> different=20
> > decision about which subscription to accept if the=20
> SUBSCRIBE forks and=20
> > the subscriber wants only one subscription.
> >=20
> > What we *do* gain is a simplification for implementors.=20
> > Because the behavior of establishing the dialog when a=20
> NOTIFY shows up=20
> > (instead of "the first of a 2xx or a NOTIFY) is much easier=20
> to code,=20
> > people will make fewer mistakes, and have fewer bugs. But you'd be=20
> > hard pressed to see a difference on the wire.
> >=20
> > > The draft's Timer N (and lack of related NOTIFY) text is
> > potentially
> > > worded in a way which does not require the tear down (and non
> > > creation) of subscription.
> >=20
> > Tear-down of existing dialogs is still an open issue, sure. I don't=20
> > get where you see any wiggle room in what happens if Timer=20
> N fires for=20
> > the first SUBSCRIBE/NOTIFY exchange.
> >=20
> > /a
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >=20
> _______________________________________________
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From christer.holmberg@ericsson.com  Wed Apr  7 23:48:21 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 8 Apr 2010 08:48:09 +0200
Thread-Topic: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
Thread-Index: AcrV4iet/Jl04c79TTyKI1KBY3+fSQAN9L+QADL0YJAAAGJCcA==
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Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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Hi,

Also, both 3265 and 3265bis contains the following text:

"A subscriber may send a SUBSCRIBE request with an "Expires" header
field of 0 in order to trigger the sending of such a NOTIFY
request; however, for the purposes of subscription and dialog
lifetime, the subscription is not considered terminated until the
NOTIFY transaction with a "Subscription-State" of "terminated"
completes."

Regards,

Christer


> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: 8. huhtikuuta 2010 9:46
> To: sipcore@ietf.org
> Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response=20
> confirms (re-)subscription? [was rfc3265bis: SIP events redux=20
> [was Minutes Posted]]
>=20
>=20
> Hi,
>=20
> In the case of DE-subscriptions (which is just one variant of=20
> re-subscription), I still think there are some issues=20
> regarding whether the SUB 200 or the NOT req acknowledges the SUB req.
>=20
> Assuming the sub dialog is terminated when an entity receives=20
> a SUB 200 that contains 'Expires: 0'.
>=20
> Now, if there is then a NOT request (that contains=20
> 'Subscription-State: terminated'), won't it be treated as an=20
> out-of-dialog request, and rejected?
>=20
> (Again, IF one is supposed to wait for the NOT req before=20
> terminating the dialog we again have a problem with the subnot draft.)
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
> =20
>=20
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> > Sent: 7. huhtikuuta 2010 9:21
> > To: Adam Roach; Brett Tate
> > Cc: SIPCORE
> > Subject: [sipcore] Rfc3265bis: NOTIFY request or response confirms=20
> > (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes=20
> > Posted]]
> >=20
> >=20
> > Hi,
> >=20
> > I haven't double checked whether it is said in 3265bis, but we also=20
> > need to make clear whether it is the NOTIFY request, or the NOT 200=20
> > response, that "acknowledges" the (re-)subscription.
> >=20
> > Because, a NOTIFY request can also be rejected (e.g. due to=20
> overload),=20
> > so in order to avoid sub state unsynch every entity in the=20
> path needs=20
> > to know what a NOT non-200 response means.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> > =20
> >=20
> > > -----Original Message-----
> > > From: sipcore-bounces@ietf.org
> > > [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> > > Sent: 7. huhtikuuta 2010 2:37
> > > To: Brett Tate
> > > Cc: SIPCORE
> > > Subject: Re: [sipcore] rfc3265bis: SIP events redux [was Minutes=20
> > > Posted]
> > >=20
> > > On 4/6/10 17:46, Apr 6, Brett Tate wrote:
> > > >> Is there *any* problem here? This seems to be working
> > > about as well
> > > >> as anyone might hope.
> > > >>     =20
> > > > It works okay excluding the non backward compatibility with
> > > RFC 3265 (concerning SUBSCRIBE's 2xx response impacts)...
> > >=20
> > > I don't think it's accurate to characterize the change as=20
> > > non-backwards-compatible. The only wire-visible behavior that one=20
> > > could conceivable observe is that a subscriber might make a
> > different
> > > decision about which subscription to accept if the
> > SUBSCRIBE forks and
> > > the subscriber wants only one subscription.
> > >=20
> > > What we *do* gain is a simplification for implementors.=20
> > > Because the behavior of establishing the dialog when a
> > NOTIFY shows up
> > > (instead of "the first of a 2xx or a NOTIFY) is much easier
> > to code,
> > > people will make fewer mistakes, and have fewer bugs. But=20
> you'd be=20
> > > hard pressed to see a difference on the wire.
> > >=20
> > > > The draft's Timer N (and lack of related NOTIFY) text is
> > > potentially
> > > > worded in a way which does not require the tear down (and non
> > > > creation) of subscription.
> > >=20
> > > Tear-down of existing dialogs is still an open issue,=20
> sure. I don't=20
> > > get where you see any wiggle room in what happens if Timer
> > N fires for
> > > the first SUBSCRIBE/NOTIFY exchange.
> > >=20
> > > /a
> > > _______________________________________________
> > > 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
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

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From naveen.shivanna@motorola.com  Thu Apr  8 01:35:50 2010
Return-Path: <naveen.shivanna@motorola.com>
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thread-topic: [sipcore] Need Clarification on UAS behaviour
thread-index: AcrW8iAZyoAixMHOTfa08kBzig3RaAAA6UXQ
From: "NAVEEN S-RHVG46" <naveen.shivanna@motorola.com>
To: <gao.yang2@zte.com.cn>, <sipcore@ietf.org>
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Subject: Re: [sipcore] Need Clarification on UAS behaviour
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Hi,
=20
If we handle the new request before the ACK.=20
=20
What is the behaviour if we don't receive the ACK for the first INVITE.=20
Should we close the call or should we proceed with the Re-Invite or not.
=20
Regards
Naveen.S

________________________________

From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn]=20
Sent: Thursday, April 08, 2010 1:28 PM
To: NAVEEN S-RHVG46
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Need Clarification on UAS behaviour



Hi,=20

It depends on implementation. I know these kinds of implementation:=20

1. 500 + retry after;=20
2. Buffer the new request, and handle it after ACK;=20
3. Handle the new request before ACK, if there is not violation of other =
rules(such as O/A rules).=20

And there's one variation of the second one: Just discarding the new =
request, and handle it's retransmitting copy after ACK.=20

Gao=20

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zip    : 210012
Tel    : 87211
Tel2   :(+86)-025-52877211
e_mail : gao.yang2@zte.com.cn
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=20



"NAVEEN S-RHVG46" <naveen.shivanna@motorola.com>=20
=B7=A2=BC=FE=C8=CB:  sipcore-bounces@ietf.org=20

2010-04-08 12:47=20

=CA=D5=BC=FE=C8=CB
<sipcore@ietf.org>=20
=B3=AD=CB=CD
=D6=F7=CC=E2
[sipcore] Need Clarification on UAS behaviour

=09




Hi,=20
 =20
In Section 14.2 UAS Behaviour, we have requirement stating as below=20
 =20
   A UAS that receives a second INVITE before it sends the final
  response to a first INVITE with a lower CSeq sequence number on the
  same dialog MUST return a 500 (Server Internal Error) response to the
  second INVITE and MUST include a Retry-After header field with a
  randomly chosen value of between 0 and 10 seconds.=20
 =20
Above requirement is valid when we receive INVITE, before sending 2xx =
response... we are responding with 500 in our implementation...=20
 =20
 =20
What's the UAS behaviour ?=20
 =20
If we receive INVITE, after sending final response, before getting ACK.=20
 =20
 =20
 =20
 =20

 =20
Regards,=20
Naveen=20
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dgb2312">
<META content=3D"MSHTML 6.00.2900.2180" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D1><SPAN=20
class=3D735393008-08042010>Hi,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D735393008-08042010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D1><SPAN=20
class=3D735393008-08042010>If we handle the new request before the ACK.=20
</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D1><SPAN=20
class=3D735393008-08042010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D1><SPAN=20
class=3D735393008-08042010>What is&nbsp;the behaviour if we don't =
receive the ACK=20
for the first INVITE. </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D1><SPAN=20
class=3D735393008-08042010>Should we close the call or should we proceed =
with the=20
Re-Invite or not.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D1><SPAN=20
class=3D735393008-08042010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D1><SPAN=20
class=3D735393008-08042010>Regards</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D1><SPAN=20
class=3D735393008-08042010>Naveen.S</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><BR></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> gao.yang2@zte.com.cn=20
[mailto:gao.yang2@zte.com.cn] <BR><B>Sent:</B> Thursday, April 08, 2010 =
1:28=20
PM<BR><B>To:</B> NAVEEN S-RHVG46<BR><B>Cc:</B>=20
sipcore@ietf.org<BR><B>Subject:</B> Re: [sipcore] Need Clarification on =
UAS=20
behaviour<BR></FONT><BR></DIV>
<DIV></DIV><BR><FONT face=3Dsans-serif size=3D2>Hi,</FONT> <BR><BR><FONT =

face=3Dsans-serif size=3D2>It depends on implementation. I know these =
kinds of=20
implementation:</FONT> <BR><BR><FONT face=3Dsans-serif size=3D2>1. 500 + =
retry=20
after;</FONT> <BR><FONT face=3Dsans-serif size=3D2>2. Buffer the new =
request, and=20
handle it after ACK;</FONT> <BR><FONT face=3Dsans-serif size=3D2>3. =
Handle the new=20
request before ACK, if there is not violation of other rules(such as O/A =

rules).</FONT> <BR><BR><FONT face=3Dsans-serif size=3D2>And there's one =
variation of=20
the second one: Just discarding the new request, and handle it's =
retransmitting=20
copy after ACK.</FONT> <BR><BR><FONT face=3Dsans-serif =
size=3D2>Gao</FONT>=20
<BR><BR><FONT face=3Dsans-serif =
size=3D2>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<BR>Zip=20
&nbsp; &nbsp;: 210012<BR>Tel &nbsp; &nbsp;: 87211<BR>Tel2 &nbsp;=20
:(+86)-025-52877211<BR>e_mail :=20
gao.yang2@zte.com.cn<BR>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</FONT> =
<BR><BR><BR>
<TABLE width=3D"100%">
  <TBODY>
  <TR vAlign=3Dtop>
    <TD width=3D"26%"><FONT face=3Dsans-serif size=3D1><B>"NAVEEN =
S-RHVG46"=20
      &lt;naveen.shivanna@motorola.com&gt;</B> </FONT><BR><FONT =
face=3Dsans-serif=20
      size=3D1>=B7=A2=BC=FE=C8=CB: &nbsp;sipcore-bounces@ietf.org</FONT> =

      <P><FONT face=3Dsans-serif size=3D1>2010-04-08 12:47</FONT> </P>
    <TD width=3D"73%">
      <TABLE width=3D"100%">
        <TBODY>
        <TR vAlign=3Dtop>
          <TD>
            <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=CA=D5=BC=FE=C8=CB</FONT></DIV>
          <TD><FONT face=3Dsans-serif =
size=3D1>&lt;sipcore@ietf.org&gt;</FONT>=20
        <TR vAlign=3Dtop>
          <TD>
            <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=B3=AD=CB=CD</FONT></DIV>
          <TD>
        <TR vAlign=3Dtop>
          <TD>
            <DIV align=3Dright><FONT face=3Dsans-serif =
size=3D1>=D6=F7=CC=E2</FONT></DIV>
          <TD><FONT face=3Dsans-serif size=3D1>[sipcore] Need =
Clarification on UAS=20
            behaviour</FONT></TR></TBODY></TABLE><BR>
      <TABLE>
        <TBODY>
        <TR vAlign=3Dtop>
          <TD>
          =
<TD></TR></TBODY></TABLE><BR></TR></TBODY></TABLE><BR><BR><BR><FONT =
face=3DArial=20
size=3D2>Hi,</FONT> <BR><FONT size=3D3>&nbsp;</FONT> <BR><FONT =
face=3DArial size=3D2>In=20
Section 14.2 UAS Behaviour, we have requirement stating as below</FONT>=20
<BR><FONT size=3D3>&nbsp;</FONT> <BR><FONT face=3DArial =
size=3D2><B>&nbsp; &nbsp;A UAS=20
that receives a second INVITE before it sends the final<BR>&nbsp; =
response to a=20
first INVITE with a lower CSeq sequence number on the<BR>&nbsp; same =
dialog MUST=20
return a 500 (Server Internal Error) response to the<BR>&nbsp; second =
INVITE and=20
MUST include a Retry-After header field with a<BR>&nbsp; randomly chosen =
value=20
of between 0 and 10 seconds.</B></FONT> <BR><FONT size=3D3>&nbsp;</FONT> =
<BR><FONT=20
face=3DArial size=3D2>Above requirement is valid when we receive INVITE, =
before=20
sending 2xx response... we are responding with 500 in our=20
implementation...</FONT> <BR><FONT size=3D3>&nbsp;</FONT> <BR><FONT=20
size=3D3>&nbsp;</FONT> <BR><FONT face=3DArial size=3D2>What's the UAS =
behaviour=20
?</FONT> <BR><FONT size=3D3>&nbsp;</FONT> <BR><FONT face=3DArial =
color=3Dred size=3D2>If=20
we receive INVITE, after sending final response, before getting ACK.=20
</FONT><BR><FONT size=3D3>&nbsp;</FONT> <BR><IMG=20
src=3D"cid:735393008@08042010-2024"> <BR><FONT face=3DArial =
size=3D2>&nbsp;</FONT>=20
<BR><FONT face=3DArial size=3D2><B>&nbsp;</B></FONT> <BR><BR><FONT=20
size=3D3>&nbsp;</FONT> <BR><FONT size=3D3>Regards,</FONT> <BR><FONT=20
size=3D3>Naveen</FONT> <BR><FONT size=3D3>&nbsp;</FONT><TT><FONT=20
size=3D2>_______________________________________________<BR>sipcore =
mailing=20
list<BR>sipcore@ietf.org<BR>https://www.ietf.org/mailman/listinfo/sipcore=
<BR></FONT></TT><BR><BR><PRE>--------------------------------------------=
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From christer.holmberg@ericsson.com  Thu Apr  8 02:47:04 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>,  "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 8 Apr 2010 11:46:56 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

>>>Why not just send the INVITE and see if it fails?  In the most=20
>>>common case (success), that saves one round-trip.
>>=20
>>Because that is crappy protocol design. And, the INVITE might trigger=20
>>different resources etc to be reserved, so you don't want to keep=20
>>sending those.
>=20
>What's so crappy about using an INVITE to say you want to=20
>start a session?  A failure of an INVITE request isn't a=20
>protocol failure or error condition (unless the response code=20
>is an error condition type). =20
>=20
>What's next, we need an OPTIONS before the OPTIONS to make=20
>sure the second OPTIONS won't fail??  ;)
>=20
>Sure the INVITE might trigger resources reservation - that's=20
>GOOD, because it verifies the resources are available - if=20
>those resources aren't available the request will fail,=20
>whereas for an OPTIONS it wouldn't have.

The resources will be released when the first INVITE is rejected, and a new=
 resource reservation procedure will start when the next INVITE comes.

Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Thu, 8 Apr 2010 12:12:34 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,
=20
>[JRE] But even if you then target the INVITE request at the=20
>GRUU of the preferred user, there is no guarantee it will=20
>succeed (e.g., no media resources, UA has become busy in the=20
>meantime), and because it is targeted directly, you lose the=20
>opportunity to take alternative action, e.g., forwarding to=20
>voicemail or to another UA with similar capabilities. It is a=20
>case of the calling user trying to override policies=20
>associated with the called user, which I think is a bad=20
>thing. The result of the OPTIONS request can only ever act as=20
>a guideline - it is effectively no better than presence.
>=20
>It would be helpful to have a concrete example of something=20
>that could go into an INVITE request targeted at a particular=20
>UA known to have a particular capability, that could not go=20
>into an INVITE request targeted at the AOR and with caller=20
>preferences RECOMMENDING selection of a UA with that=20
>particular capability. Apologies if such an example has=20
>already been given earlier in the discussion.

I don't think I have said that the INVITE would necessarily be targeted to =
a particular UA. There MAY of course be use-cases where you would target al=
so the INVITE to a specific UAS. For example, if an UAS supports a specific=
 capability, and you don't want the call to proceed unless that UAS accepts=
 the INVITE. Then, if the INVITE fails, it may still be possible for the ca=
ll to be forwarded to a voice mail etc, depending on policies and service d=
esign in the network.

But, the main point of the O-3xx mechanisms is that it allows you to target=
 *OPTIONS* request to particular UAs, in order to get the capabilities of t=
hose.

That is the only thing the mechanism is supposed to provide. What you then =
do with that information is a separate issue, and that seems to be what the=
 discussion is now about.

Now, we can discuss use-cases etc, but I just want to make it clear that th=
e O-3xx mechanism itself does not make any assumptions regarding use-cases.=
 It only allows the usage of OPTIONS in forking scenarios - nothing more, n=
othing less. Apart from providing a way to deal with the forking issue, it =
does not solve any other issues, bugs or shortcomings there may be with OPT=
IONS.

Regards,

Christer

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 8 Apr 2010 06:16:02 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Thursday, April 08, 2010 5:47 AM
>=20
> Hadriel said:
> >What's so crappy about using an INVITE to say you want to
> >start a session?  A failure of an INVITE request isn't a
> >protocol failure or error condition (unless the response code
> >is an error condition type).
> >
> >What's next, we need an OPTIONS before the OPTIONS to make
> >sure the second OPTIONS won't fail??  ;)
> >
> >Sure the INVITE might trigger resources reservation - that's
> >GOOD, because it verifies the resources are available - if
> >those resources aren't available the request will fail,
> >whereas for an OPTIONS it wouldn't have.
>=20
> The resources will be released when the first INVITE is rejected, and a
> new resource reservation procedure will start when the next INVITE comes.

Yup, but I don't see what that has to do with what we're talking about? (th=
e statement confuses me in this context)

And again it's GOOD that the resources are reserved again.  It may not be g=
ood from a performance perspective, but it guarantees accuracy.  Resources =
change constantly.  There are any number of resource availability changes i=
n the network between the OPTIONS and the INVITE, or between two INVITES.  =
Obviously you could do a resource reservation model that is separate from I=
NVITEs (or SIP altogether), but I don't see how that's relevant to this dis=
cussion. (I must be missing something?)

-hadriel

From christer.holmberg@ericsson.com  Thu Apr  8 03:30:06 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "WORLEY, DALE R (DALE)" <dworley@avaya.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>,  "sipcore@ietf.org" <sipcore@ietf.org>
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Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>>From my perspective, the main use-case is not to discover services,=20
>>but to discover capabilities in order to determine what services can be a=
pplied.
>>=20
>>The idea is also that the OPTIONS request would be sent prior to the=20
>>INVITE, to try to make sure that the returned contacts are actually=20
>>available.
>=20
>But you know that would never work in practice/the-real-world. Think about=
 all the routing rules=20
>people have for INVITEs - in some cases *only* for INVITEs. =20
>And even if they route OPTIONS identically in the SIP domain,=20
>for any given AoR it'll go to any of: a SIP UA, or a PSTN=20
>gateway, or a H.323-gateway, or an MGCP call-agent, or an=20
>IP-PBX, or a foobar-protocol-gateway, or a vmail server, or=20
>an announcement server, or an app server, etc.  Just because=20
>the OPTIONS reaches the end of the SIP domain, means nothing=20
>about whether an INVITE to the higher-layer target (which in=20
>practice is a E.164 number, that crosses from SIP into many=20
>other protocol domains) is actually available to receive a call.

The assumption is that the OPTIONS request reaches the registrar, that retu=
rns the registered contacts. It is never forwarded to any UAS.

Then, at some point an INVITE is sent, and it is addressed to the same AOR =
(the INVITE request might be routed differently than the OPTIONS request). =
Based on the UAS information, the INVITE may now contain feature tags etc t=
hat indicates a wish to reach an UAS with certain capabilities.

Now, unless the INVITE is addressed to a specific UAS, it can of course not=
 be guranteed that the INVITE will actually reach such UAS. But, that is a =
"feature" of RFC 3841, and it would be the case even if you had never sent =
any OPTIONS. What OPTIONS allows provides you is capability information abo=
ut the UAS(s) associated with the AOR, which might be useful when sending t=
he INVITE.

No, it does not guarantee that the user is available - it only provides you=
 with that same information as OPTIONS in a non-forking case does.

Regards,

Christer=

From christer.holmberg@ericsson.com  Thu Apr  8 03:31:54 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Thu, 8 Apr 2010 12:31:47 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20
=09
>At the moment, my view is "Do nothing". The amount of resource needed for =
this discussion does not justify the problem.
>	=20
>If someone wants to submit an author draft on a specific issue and fix, th=
en fine, I'll reconsider.

My intention has been to submit a draft that describes the OPTIONS 3xx mech=
anism.

Regards,

Christer

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Date: Thu, 8 Apr 2010 12:35:45 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>The more I read through this thread, the more it looks like we're running =
around with a hammer, desperately looking for nails to hit. Barring nails, =
we'll settle for screws. Or bolts. Or maybe=20
>kittens, if it comes to that.
>=09
>If you're trying to accomplish something, I would strongly recommend start=
ing from the other end. Instead of saying "I have a tool, let's make a prob=
lem for it to solve," let's concentrate on=20
>clearly defining existing problems. Once we agree on the problems, we can =
talk about what tools make sense to solve them.

That problem is that OPTIONS doesn't work in forking cases, which makes is =
quite useless in those cases.

Regards,

Christer=

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Thu, 8 Apr 2010 06:44:40 -0400
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi Christer,
I know you're frustrated we keep going back to a use-case, but you're askin=
g for consensus on whether people are interested in working on the solution=
.  Without an actual use-case, how can we do that?  Why would people agree =
to spend time on something they don't see any use for solving?

-hadriel

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Christer Holmberg
> Sent: Thursday, April 08, 2010 6:13 AM
> To: Elwell, John; Hans Erik van Elburg
> Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
>=20
> Hi,
>=20
> >[JRE] But even if you then target the INVITE request at the
> >GRUU of the preferred user, there is no guarantee it will
> >succeed (e.g., no media resources, UA has become busy in the
> >meantime), and because it is targeted directly, you lose the
> >opportunity to take alternative action, e.g., forwarding to
> >voicemail or to another UA with similar capabilities. It is a
> >case of the calling user trying to override policies
> >associated with the called user, which I think is a bad
> >thing. The result of the OPTIONS request can only ever act as
> >a guideline - it is effectively no better than presence.
> >
> >It would be helpful to have a concrete example of something
> >that could go into an INVITE request targeted at a particular
> >UA known to have a particular capability, that could not go
> >into an INVITE request targeted at the AOR and with caller
> >preferences RECOMMENDING selection of a UA with that
> >particular capability. Apologies if such an example has
> >already been given earlier in the discussion.
>=20
> I don't think I have said that the INVITE would necessarily be targeted t=
o
> a particular UA. There MAY of course be use-cases where you would target
> also the INVITE to a specific UAS. For example, if an UAS supports a
> specific capability, and you don't want the call to proceed unless that
> UAS accepts the INVITE. Then, if the INVITE fails, it may still be
> possible for the call to be forwarded to a voice mail etc, depending on
> policies and service design in the network.
>=20
> But, the main point of the O-3xx mechanisms is that it allows you to
> target *OPTIONS* request to particular UAs, in order to get the
> capabilities of those.
>=20
> That is the only thing the mechanism is supposed to provide. What you the=
n
> do with that information is a separate issue, and that seems to be what
> the discussion is now about.
>=20
> Now, we can discuss use-cases etc, but I just want to make it clear that
> the O-3xx mechanism itself does not make any assumptions regarding use-
> cases. It only allows the usage of OPTIONS in forking scenarios - nothing
> more, nothing less. Apart from providing a way to deal with the forking
> issue, it does not solve any other issues, bugs or shortcomings there may
> be with OPTIONS.
>=20
> Regards,
>=20
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From john.elwell@siemens-enterprise.com  Thu Apr  8 03:50:00 2010
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To: Christer Holmberg <christer.holmberg@ericsson.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Thu, 8 Apr 2010 12:49:51 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Christer,

Thanks for that clarification - I wasn't certain about my interpretation.

So we have the ability to target OPTIONS requests at individual UAs that ar=
e contacts for a given AOR. From that I discover that UA N supports feature=
 X. I want to submit an INVITE request that uses feature X. If you say we d=
on't target the INVITE request at UA N, then presumably I would target it a=
t the AOR, which may or may not result in the request being forwarded to UA=
 N. I could use caller-prefs to indicate a preference for a UA that support=
s feature X, and this will doubtless increase the chances that the INVITE r=
equest will be forwarded to UA N.

What exactly have I gained by compared with skipping OPTIONS and just sendi=
ng an INVITE request to the AOR with a caller-pref for feature X? For examp=
le, are you suggesting it would be harmful to send the INVITE request in a =
case where no UA supports feature X?

Note that in the typical SIP trapezoid, with 3 registered contacts, use of =
OPTIONS would result in a significant amount of additional signalling traff=
ic. Assuming the second proxy sends back 3xx with 3 contacts, resulting in =
3 further OPTIONS requests:
- 1st OPTIONS request over two hops, with response: 4 message-hops
- 2nd, 3rd and 4th OPTIONS requests, each over three hops, with responses: =
18 message-hops
In other words, 22 additional SIP message hops (assuming the eventual INVIT=
E request results in the same amount of signalling whether or not OPTIONS i=
s used). So optimisation can hardly be claimed as a benefit.

John=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 08 April 2010 11:13
> To: Elwell, John; Hans Erik van Elburg
> Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
>=20
> Hi,
> =20
> >[JRE] But even if you then target the INVITE request at the=20
> >GRUU of the preferred user, there is no guarantee it will=20
> >succeed (e.g., no media resources, UA has become busy in the=20
> >meantime), and because it is targeted directly, you lose the=20
> >opportunity to take alternative action, e.g., forwarding to=20
> >voicemail or to another UA with similar capabilities. It is a=20
> >case of the calling user trying to override policies=20
> >associated with the called user, which I think is a bad=20
> >thing. The result of the OPTIONS request can only ever act as=20
> >a guideline - it is effectively no better than presence.
> >=20
> >It would be helpful to have a concrete example of something=20
> >that could go into an INVITE request targeted at a particular=20
> >UA known to have a particular capability, that could not go=20
> >into an INVITE request targeted at the AOR and with caller=20
> >preferences RECOMMENDING selection of a UA with that=20
> >particular capability. Apologies if such an example has=20
> >already been given earlier in the discussion.
>=20
> I don't think I have said that the INVITE would necessarily=20
> be targeted to a particular UA. There MAY of course be=20
> use-cases where you would target also the INVITE to a=20
> specific UAS. For example, if an UAS supports a specific=20
> capability, and you don't want the call to proceed unless=20
> that UAS accepts the INVITE. Then, if the INVITE fails, it=20
> may still be possible for the call to be forwarded to a voice=20
> mail etc, depending on policies and service design in the network.
>=20
> But, the main point of the O-3xx mechanisms is that it allows=20
> you to target *OPTIONS* request to particular UAs, in order=20
> to get the capabilities of those.
>=20
> That is the only thing the mechanism is supposed to provide.=20
> What you then do with that information is a separate issue,=20
> and that seems to be what the discussion is now about.
>=20
> Now, we can discuss use-cases etc, but I just want to make it=20
> clear that the O-3xx mechanism itself does not make any=20
> assumptions regarding use-cases. It only allows the usage of=20
> OPTIONS in forking scenarios - nothing more, nothing less.=20
> Apart from providing a way to deal with the forking issue, it=20
> does not solve any other issues, bugs or shortcomings there=20
> may be with OPTIONS.
>=20
> Regards,
>=20
> Christer
> =

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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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--00c09f8e6090ded22b0483b8bc8b
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Hi John,

But if you only want to probe the devices behind the AOR for some reason
(knowing the limitations of the obtained information), then you do not want
to use INVITE do you?

/Hans Erik van Elburg


On Thu, Apr 8, 2010 at 12:49 PM, Elwell, John <
john.elwell@siemens-enterprise.com> wrote:

> Christer,
>
> Thanks for that clarification - I wasn't certain about my interpretation.
>
> So we have the ability to target OPTIONS requests at individual UAs that
> are contacts for a given AOR. From that I discover that UA N supports
> feature X. I want to submit an INVITE request that uses feature X. If you
> say we don't target the INVITE request at UA N, then presumably I would
> target it at the AOR, which may or may not result in the request being
> forwarded to UA N. I could use caller-prefs to indicate a preference for a
> UA that supports feature X, and this will doubtless increase the chances
> that the INVITE request will be forwarded to UA N.
>
> What exactly have I gained by compared with skipping OPTIONS and just
> sending an INVITE request to the AOR with a caller-pref for feature X? For
> example, are you suggesting it would be harmful to send the INVITE request
> in a case where no UA supports feature X?
>
> Note that in the typical SIP trapezoid, with 3 registered contacts, use of
> OPTIONS would result in a significant amount of additional signalling
> traffic. Assuming the second proxy sends back 3xx with 3 contacts, resulting
> in 3 further OPTIONS requests:
> - 1st OPTIONS request over two hops, with response: 4 message-hops
> - 2nd, 3rd and 4th OPTIONS requests, each over three hops, with responses:
> 18 message-hops
> In other words, 22 additional SIP message hops (assuming the eventual
> INVITE request results in the same amount of signalling whether or not
> OPTIONS is used). So optimisation can hardly be claimed as a benefit.
>
> John
>
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: 08 April 2010 11:13
> > To: Elwell, John; Hans Erik van Elburg
> > Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> > Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?
> >
> >
> > Hi,
> >
> > >[JRE] But even if you then target the INVITE request at the
> > >GRUU of the preferred user, there is no guarantee it will
> > >succeed (e.g., no media resources, UA has become busy in the
> > >meantime), and because it is targeted directly, you lose the
> > >opportunity to take alternative action, e.g., forwarding to
> > >voicemail or to another UA with similar capabilities. It is a
> > >case of the calling user trying to override policies
> > >associated with the called user, which I think is a bad
> > >thing. The result of the OPTIONS request can only ever act as
> > >a guideline - it is effectively no better than presence.
> > >
> > >It would be helpful to have a concrete example of something
> > >that could go into an INVITE request targeted at a particular
> > >UA known to have a particular capability, that could not go
> > >into an INVITE request targeted at the AOR and with caller
> > >preferences RECOMMENDING selection of a UA with that
> > >particular capability. Apologies if such an example has
> > >already been given earlier in the discussion.
> >
> > I don't think I have said that the INVITE would necessarily
> > be targeted to a particular UA. There MAY of course be
> > use-cases where you would target also the INVITE to a
> > specific UAS. For example, if an UAS supports a specific
> > capability, and you don't want the call to proceed unless
> > that UAS accepts the INVITE. Then, if the INVITE fails, it
> > may still be possible for the call to be forwarded to a voice
> > mail etc, depending on policies and service design in the network.
> >
> > But, the main point of the O-3xx mechanisms is that it allows
> > you to target *OPTIONS* request to particular UAs, in order
> > to get the capabilities of those.
> >
> > That is the only thing the mechanism is supposed to provide.
> > What you then do with that information is a separate issue,
> > and that seems to be what the discussion is now about.
> >
> > Now, we can discuss use-cases etc, but I just want to make it
> > clear that the O-3xx mechanism itself does not make any
> > assumptions regarding use-cases. It only allows the usage of
> > OPTIONS in forking scenarios - nothing more, nothing less.
> > Apart from providing a way to deal with the forking issue, it
> > does not solve any other issues, bugs or shortcomings there
> > may be with OPTIONS.
> >
> > Regards,
> >
> > Christer
> >
>

--00c09f8e6090ded22b0483b8bc8b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi John,<br><br>But if you only want to probe the devices behind the AOR fo=
r some reason (knowing the limitations of the obtained information), then y=
ou do not want to use INVITE do you?<br><br clear=3D"all">/Hans Erik van El=
burg<br>

<br><br><div class=3D"gmail_quote">On Thu, Apr 8, 2010 at 12:49 PM, Elwell,=
 John <span dir=3D"ltr">&lt;<a href=3D"mailto:john.elwell@siemens-enterpris=
e.com">john.elwell@siemens-enterprise.com</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: =
1px solid rgb(204, 204, 204); padding-left: 1ex;">
Christer,<br>
<br>
Thanks for that clarification - I wasn&#39;t certain about my interpretatio=
n.<br>
<br>
So we have the ability to target OPTIONS requests at individual UAs that ar=
e contacts for a given AOR. From that I discover that UA N supports feature=
 X. I want to submit an INVITE request that uses feature X. If you say we d=
on&#39;t target the INVITE request at UA N, then presumably I would target =
it at the AOR, which may or may not result in the request being forwarded t=
o UA N. I could use caller-prefs to indicate a preference for a UA that sup=
ports feature X, and this will doubtless increase the chances that the INVI=
TE request will be forwarded to UA N.<br>

<br>
What exactly have I gained by compared with skipping OPTIONS and just sendi=
ng an INVITE request to the AOR with a caller-pref for feature X? For examp=
le, are you suggesting it would be harmful to send the INVITE request in a =
case where no UA supports feature X?<br>

<br>
Note that in the typical SIP trapezoid, with 3 registered contacts, use of =
OPTIONS would result in a significant amount of additional signalling traff=
ic. Assuming the second proxy sends back 3xx with 3 contacts, resulting in =
3 further OPTIONS requests:<br>

- 1st OPTIONS request over two hops, with response: 4 message-hops<br>
- 2nd, 3rd and 4th OPTIONS requests, each over three hops, with responses: =
18 message-hops<br>
In other words, 22 additional SIP message hops (assuming the eventual INVIT=
E request results in the same amount of signalling whether or not OPTIONS i=
s used). So optimisation can hardly be claimed as a benefit.<br>
<font color=3D"#888888"><br>
John<br>
</font><div class=3D"im"><br>
&gt; -----Original Message-----<br>
&gt; From: Christer Holmberg [mailto:<a href=3D"mailto:christer.holmberg@er=
icsson.com">christer.holmberg@ericsson.com</a>]<br>
</div><div class=3D"im">&gt; Sent: 08 April 2010 11:13<br>
&gt; To: Elwell, John; Hans Erik van Elburg<br>
&gt; Cc: WORLEY, DALE R (DALE); <a href=3D"mailto:sipcore@ietf.org">sipcore=
@ietf.org</a><br>
&gt; Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?<br>
&gt;<br>
&gt;<br>
</div><div><div></div><div class=3D"h5">&gt; Hi,<br>
&gt;<br>
&gt; &gt;[JRE] But even if you then target the INVITE request at the<br>
&gt; &gt;GRUU of the preferred user, there is no guarantee it will<br>
&gt; &gt;succeed (e.g., no media resources, UA has become busy in the<br>
&gt; &gt;meantime), and because it is targeted directly, you lose the<br>
&gt; &gt;opportunity to take alternative action, e.g., forwarding to<br>
&gt; &gt;voicemail or to another UA with similar capabilities. It is a<br>
&gt; &gt;case of the calling user trying to override policies<br>
&gt; &gt;associated with the called user, which I think is a bad<br>
&gt; &gt;thing. The result of the OPTIONS request can only ever act as<br>
&gt; &gt;a guideline - it is effectively no better than presence.<br>
&gt; &gt;<br>
&gt; &gt;It would be helpful to have a concrete example of something<br>
&gt; &gt;that could go into an INVITE request targeted at a particular<br>
&gt; &gt;UA known to have a particular capability, that could not go<br>
&gt; &gt;into an INVITE request targeted at the AOR and with caller<br>
&gt; &gt;preferences RECOMMENDING selection of a UA with that<br>
&gt; &gt;particular capability. Apologies if such an example has<br>
&gt; &gt;already been given earlier in the discussion.<br>
&gt;<br>
&gt; I don&#39;t think I have said that the INVITE would necessarily<br>
&gt; be targeted to a particular UA. There MAY of course be<br>
&gt; use-cases where you would target also the INVITE to a<br>
&gt; specific UAS. For example, if an UAS supports a specific<br>
&gt; capability, and you don&#39;t want the call to proceed unless<br>
&gt; that UAS accepts the INVITE. Then, if the INVITE fails, it<br>
&gt; may still be possible for the call to be forwarded to a voice<br>
&gt; mail etc, depending on policies and service design in the network.<br>
&gt;<br>
&gt; But, the main point of the O-3xx mechanisms is that it allows<br>
&gt; you to target *OPTIONS* request to particular UAs, in order<br>
&gt; to get the capabilities of those.<br>
&gt;<br>
&gt; That is the only thing the mechanism is supposed to provide.<br>
&gt; What you then do with that information is a separate issue,<br>
&gt; and that seems to be what the discussion is now about.<br>
&gt;<br>
&gt; Now, we can discuss use-cases etc, but I just want to make it<br>
&gt; clear that the O-3xx mechanism itself does not make any<br>
&gt; assumptions regarding use-cases. It only allows the usage of<br>
&gt; OPTIONS in forking scenarios - nothing more, nothing less.<br>
&gt; Apart from providing a way to deal with the forking issue, it<br>
&gt; does not solve any other issues, bugs or shortcomings there<br>
&gt; may be with OPTIONS.<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt; </div></div></blockquote></div><br>

--00c09f8e6090ded22b0483b8bc8b--

From brett@broadsoft.com  Thu Apr  8 05:37:41 2010
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From: Brett Tate <brett@broadsoft.com>
To: SIPCORE <sipcore@ietf.org>
Date: Thu, 8 Apr 2010 05:36:42 -0700
Thread-Topic: RFC 4320: deprecation of non-INVITE 408 responses
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Subject: [sipcore] RFC 4320: deprecation of non-INVITE 408 responses
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Since it was raised within rfc3265bis discussion, I thought that I'd attemp=
t to get clarity concerning RFC 4320's deprecation of non-INVITE 408 respon=
ses in case errata should be opened against RFC 4320.

RFC 4320 intentionally or unintentionally appears to have deprecated all no=
n-INVITE 408 responses.  However I thought that I'd still ask, is the depre=
cation only applicable for Timer F expiration type situations (i.e. the rea=
son for the deprecation)?

If not limited to Timer F expiration type situations, such a deprecation ca=
uses some ambiguity concerning shorter timeouts on other interfaces.  For i=
nstance, a received SUBSCRIBE is transformed into another protocol's FOO me=
ssage.  The FOO transaction delivery expires after 1 second.  Can a 408 res=
ponse (although 504 might be better per rfc3261) be sent for the SUBSCRIBE?=
  What about a B2BUA which using a short expiration (like 1 second) on the =
send side; can it send a 408 response or must it really start sending 504 o=
r other response instead?=20

Thanks,
Brett



From john.elwell@siemens-enterprise.com  Thu Apr  8 06:39:58 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Thu, 8 Apr 2010 15:39:52 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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References: <4BB24B81.1050006@nostrum.com> <y2o9ae56b1e1004070218v35c4725ex3409702ae4eea61@mail.gmail.com> <FF84A09F50A6DC48ACB6714F4666CC745E21C7C978@ESESSCMS0354.eemea.ericsson.se> <A444A0F8084434499206E78C106220CADE204402@MCHP058A.global-ad.net> <FF84A09F50A6DC48ACB6714F4666CC745E21C7C9F2@ESESSCMS0354.eemea.ericsson.se> <A444A0F8084434499206E78C106220CADE2044BC@MCHP058A.global-ad.net> <FF84A09F50A6DC48ACB6714F4666CC745E21C7CB37@ESESSCMS0354.eemea.ericsson.se> <A444A0F8084434499206E78C106220CADE204504@MCHP058A.global-ad.net> <FF84A09F50A6DC48ACB6714F4666CC745E21C7D0F3@ESESSCMS0354.eemea.ericsson.se> <A444A0F8084434499206E78C106220CADE2049BD@MCHP058A.global-ad.net> <q2t9ae56b1e1004080522qdb07b2deqae08626113123175@mail.gmail.com>
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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I thought we were talking about doing OPTIONS as a prelude to doing INVITE =
(or SUBSCRIBE or MESSAGE or whatever), in which case you could just send th=
e request concerned and see what happens - no need to bother with OPTIONS. =
But you now seem to be suggesting the use of OPTIONS standalone.

In either case, nobody seems to have given a convincing use case where prob=
ing all the devices behind an AOR is of significant benefit, and if there i=
s a significant benefit, why you can't just obtain the device contact URIs =
using the reg event package and then send OPTIONS to each.

John

> -----Original Message-----
> From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]=20
> Sent: 08 April 2010 13:23
> To: Elwell, John
> Cc: Christer Holmberg; WORLEY, DALE R (DALE); sipcore@ietf.org
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
> Hi John,
>=20
> But if you only want to probe the devices behind the AOR for=20
> some reason (knowing the limitations of the obtained=20
> information), then you do not want to use INVITE do you?
>=20
> /Hans Erik van Elburg
>=20
>=20
>=20
> On Thu, Apr 8, 2010 at 12:49 PM, Elwell, John=20
> <john.elwell@siemens-enterprise.com> wrote:
>=20
>=20
> 	Christer,
> =09
> 	Thanks for that clarification - I wasn't certain about=20
> my interpretation.
> =09
> 	So we have the ability to target OPTIONS requests at=20
> individual UAs that are contacts for a given AOR. From that I=20
> discover that UA N supports feature X. I want to submit an=20
> INVITE request that uses feature X. If you say we don't=20
> target the INVITE request at UA N, then presumably I would=20
> target it at the AOR, which may or may not result in the=20
> request being forwarded to UA N. I could use caller-prefs to=20
> indicate a preference for a UA that supports feature X, and=20
> this will doubtless increase the chances that the INVITE=20
> request will be forwarded to UA N.
> =09
> 	What exactly have I gained by compared with skipping=20
> OPTIONS and just sending an INVITE request to the AOR with a=20
> caller-pref for feature X? For example, are you suggesting it=20
> would be harmful to send the INVITE request in a case where=20
> no UA supports feature X?
> =09
> 	Note that in the typical SIP trapezoid, with 3=20
> registered contacts, use of OPTIONS would result in a=20
> significant amount of additional signalling traffic. Assuming=20
> the second proxy sends back 3xx with 3 contacts, resulting in=20
> 3 further OPTIONS requests:
> 	- 1st OPTIONS request over two hops, with response: 4=20
> message-hops
> 	- 2nd, 3rd and 4th OPTIONS requests, each over three=20
> hops, with responses: 18 message-hops
> 	In other words, 22 additional SIP message hops=20
> (assuming the eventual INVITE request results in the same=20
> amount of signalling whether or not OPTIONS is used). So=20
> optimisation can hardly be claimed as a benefit.
> =09
> 	John
> =09
>=20
> 	> -----Original Message-----
> 	> From: Christer Holmberg=20
> [mailto:christer.holmberg@ericsson.com]
> =09
> 	> Sent: 08 April 2010 11:13
> 	> To: Elwell, John; Hans Erik van Elburg
> 	> Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> 	> Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?
> 	>
> 	>
> =09
> 	> Hi,
> 	>
> 	> >[JRE] But even if you then target the INVITE request at the
> 	> >GRUU of the preferred user, there is no guarantee it will
> 	> >succeed (e.g., no media resources, UA has become busy in the
> 	> >meantime), and because it is targeted directly, you lose the
> 	> >opportunity to take alternative action, e.g., forwarding to
> 	> >voicemail or to another UA with similar capabilities. It is a
> 	> >case of the calling user trying to override policies
> 	> >associated with the called user, which I think is a bad
> 	> >thing. The result of the OPTIONS request can only ever act as
> 	> >a guideline - it is effectively no better than presence.
> 	> >
> 	> >It would be helpful to have a concrete example of something
> 	> >that could go into an INVITE request targeted at a particular
> 	> >UA known to have a particular capability, that could not go
> 	> >into an INVITE request targeted at the AOR and with caller
> 	> >preferences RECOMMENDING selection of a UA with that
> 	> >particular capability. Apologies if such an example has
> 	> >already been given earlier in the discussion.
> 	>
> 	> I don't think I have said that the INVITE would necessarily
> 	> be targeted to a particular UA. There MAY of course be
> 	> use-cases where you would target also the INVITE to a
> 	> specific UAS. For example, if an UAS supports a specific
> 	> capability, and you don't want the call to proceed unless
> 	> that UAS accepts the INVITE. Then, if the INVITE fails, it
> 	> may still be possible for the call to be forwarded to a voice
> 	> mail etc, depending on policies and service design in=20
> the network.
> 	>
> 	> But, the main point of the O-3xx mechanisms is that it allows
> 	> you to target *OPTIONS* request to particular UAs, in order
> 	> to get the capabilities of those.
> 	>
> 	> That is the only thing the mechanism is supposed to provide.
> 	> What you then do with that information is a separate issue,
> 	> and that seems to be what the discussion is now about.
> 	>
> 	> Now, we can discuss use-cases etc, but I just want to make it
> 	> clear that the O-3xx mechanism itself does not make any
> 	> assumptions regarding use-cases. It only allows the usage of
> 	> OPTIONS in forking scenarios - nothing more, nothing less.
> 	> Apart from providing a way to deal with the forking issue, it
> 	> does not solve any other issues, bugs or shortcomings there
> 	> may be with OPTIONS.
> 	>
> 	> Regards,
> 	>
> 	> Christer
> 	>=20
>=20
>=20
> =

From pkyzivat@cisco.com  Thu Apr  8 07:01:22 2010
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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[as individual]

It seems that Christer has come close to offering a use case, even if 
that was not his intent. And its still a little incomplete. I'll try to 
fill it out a bit:

Alice wants to call Bob.
Her phone app first queries Bob's AOR with OPTIONS.
The response to that is a 3xx with Contact URIs for
Bob's home phone and his mobile phone. So Alice's phone
sends additional OPTIONS requests to each of those.
 From the responses it learns that Bob's mobile phone support
audio and video, while his home phone (behind a pstn gateway)
only supports audio.

Alice has video support on her phone, and since Bob may be able to 
support video too, decides to make an audio+video call. The
phone sends an INVITE with both, and caller preferences indicating
a preference for audio+video.

The INVITE reaches the server handling Bob's AOR.
It processes the callerprefs, and so decides to initially send the 
invite to Bob's mobile phone. But there is no answer there, so
the server forks the invite toward Bob's home phone. Bob answers that 
one, accepting the audio and rejecting the video.

The result of this is exactly the same as if no OPTIONS was ever sent, 
and the INVITE was sent indicating Alice's own preferences. That is the 
point that John already made.

The only "benefit" I can see from use of options here would be if it was 
discovered that Bob had no video support. In that case, the invite could 
have been sent without offering video. Is that advantageous? Does it 
cost more to initiate a call that offers video? Does it take more 
network resources to do so, even when ultimately the video is rejected?

The only real benefit I can see to this is if the calling user is given 
an enhanced user interface - akin to what one gets with buddy lists - 
where there is a display showing that Bob has two devices with differing 
capabilities, with Alice having the option to choose one or allowing the 
phone to make the choice.

To provide that feature, presence would be a much better choice.
OPTIONS is in a sense "poor man's presence". Christer has mentioned that 
he considers presence to be too "heavy weight" for this, but I remain to 
be convinced of that. Perhaps maintaining persistent subscriptions to 
presence status of everyone you might possibly call is too heavy weight, 
but doing a polling subscription prior to a call won't be any heavier 
than all this options stuff.

It may be that lots of devices don't publish presence state. But that 
can be solved in many ways. A registrar can synthesize basic presence 
state from registration data. And it could even use OPTIONS to refine that.

I suspect the real hidden issue here is about presence, and so I'd 
prefer to see it addressed from that direction. If that in turn leads to 
a desire to use OPTIONS to get some degree of presence state from 
devices that don't otherwise publish it, then lets confront it in that 
context. And if the issue is that presence is typically not authorized 
for as many requesters as OPTIONS is, then lets look at possible policy 
recommendations in that regard.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi, 
> 
>> >From my perspective, the main use-case is not to discover services, 
>>> but to discover capabilities in order to determine what services can be applied.
>>>
>>> The idea is also that the OPTIONS request would be sent prior to the 
>>> INVITE, to try to make sure that the returned contacts are actually 
>>> available.
>> But you know that would never work in practice/the-real-world. Think about all the routing rules 
>> people have for INVITEs - in some cases *only* for INVITEs.  
>> And even if they route OPTIONS identically in the SIP domain, 
>> for any given AoR it'll go to any of: a SIP UA, or a PSTN 
>> gateway, or a H.323-gateway, or an MGCP call-agent, or an 
>> IP-PBX, or a foobar-protocol-gateway, or a vmail server, or 
>> an announcement server, or an app server, etc.  Just because 
>> the OPTIONS reaches the end of the SIP domain, means nothing 
>> about whether an INVITE to the higher-layer target (which in 
>> practice is a E.164 number, that crosses from SIP into many 
>> other protocol domains) is actually available to receive a call.
> 
> The assumption is that the OPTIONS request reaches the registrar, that returns the registered contacts. It is never forwarded to any UAS.
> 
> Then, at some point an INVITE is sent, and it is addressed to the same AOR (the INVITE request might be routed differently than the OPTIONS request). Based on the UAS information, the INVITE may now contain feature tags etc that indicates a wish to reach an UAS with certain capabilities.
> 
> Now, unless the INVITE is addressed to a specific UAS, it can of course not be guranteed that the INVITE will actually reach such UAS. But, that is a "feature" of RFC 3841, and it would be the case even if you had never sent any OPTIONS. What OPTIONS allows provides you is capability information about the UAS(s) associated with the AOR, which might be useful when sending the INVITE.
> 
> No, it does not guarantee that the user is available - it only provides you with that same information as OPTIONS in a non-forking case does.
> 
> Regards,
> 
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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[as individual]

Christer Holmberg wrote:
> Hi,
> 
> In the case of DE-subscriptions (which is just one variant of re-subscription), I still think there are some issues regarding whether the SUB 200 or the NOT req acknowledges the SUB req.
> 
> Assuming the sub dialog is terminated when an entity receives a SUB 200 that contains 'Expires: 0'.
> 
> Now, if there is then a NOT request (that contains 'Subscription-State: terminated'), won't it be treated as an out-of-dialog request, and rejected?

That scenario has always bothered me. When my purpose is simply to 
request the termination of the existing dialog, and I have no further 
interest in the state, is there any reason for the NOTIFY? Can it be 
suppressed? Its especially a problem if I think the dialog is already 
broken, and I'm just trying to ensure its cleanly gone.

	Thanks,
	Paul

> (Again, IF one is supposed to wait for the NOT req before terminating the dialog we again have a problem with the subnot draft.)
> 
> Regards,
> 
> Christer
> 
> 
> 
> 
> 
>  
> 
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org 
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
>> Sent: 7. huhtikuuta 2010 9:21
>> To: Adam Roach; Brett Tate
>> Cc: SIPCORE
>> Subject: [sipcore] Rfc3265bis: NOTIFY request or response 
>> confirms (re-)subscription? [was rfc3265bis: SIP events redux 
>> [was Minutes Posted]]
>>
>>
>> Hi,
>>
>> I haven't double checked whether it is said in 3265bis, but 
>> we also need to make clear whether it is the NOTIFY request, 
>> or the NOT 200 response, that "acknowledges" the (re-)subscription.
>>
>> Because, a NOTIFY request can also be rejected (e.g. due to 
>> overload), so in order to avoid sub state unsynch every 
>> entity in the path needs to know what a NOT non-200 response means.
>>
>> Regards,
>>
>> Christer
>>
>>
>>  
>>
>>> -----Original Message-----
>>> From: sipcore-bounces@ietf.org
>>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
>>> Sent: 7. huhtikuuta 2010 2:37
>>> To: Brett Tate
>>> Cc: SIPCORE
>>> Subject: Re: [sipcore] rfc3265bis: SIP events redux [was Minutes 
>>> Posted]
>>>
>>> On 4/6/10 17:46, Apr 6, Brett Tate wrote:
>>>>> Is there *any* problem here? This seems to be working
>>> about as well
>>>>> as anyone might hope.
>>>>>      
>>>> It works okay excluding the non backward compatibility with
>>> RFC 3265 (concerning SUBSCRIBE's 2xx response impacts)...
>>>
>>> I don't think it's accurate to characterize the change as 
>>> non-backwards-compatible. The only wire-visible behavior that one 
>>> could conceivable observe is that a subscriber might make a 
>> different 
>>> decision about which subscription to accept if the 
>> SUBSCRIBE forks and 
>>> the subscriber wants only one subscription.
>>>
>>> What we *do* gain is a simplification for implementors. 
>>> Because the behavior of establishing the dialog when a 
>> NOTIFY shows up 
>>> (instead of "the first of a 2xx or a NOTIFY) is much easier 
>> to code, 
>>> people will make fewer mistakes, and have fewer bugs. But you'd be 
>>> hard pressed to see a difference on the wire.
>>>
>>>> The draft's Timer N (and lack of related NOTIFY) text is
>>> potentially
>>>> worded in a way which does not require the tear down (and non
>>>> creation) of subscription.
>>> Tear-down of existing dialogs is still an open issue, sure. I don't 
>>> get where you see any wiggle room in what happens if Timer 
>> N fires for 
>>> the first SUBSCRIBE/NOTIFY exchange.
>>>
>>> /a
>>> _______________________________________________
>>> 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
>>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From keith.drage@alcatel-lucent.com  Thu Apr  8 07:51:10 2010
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Thread-Topic: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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Your argument is accepted.

That is an appropriate compatibility analysis.

Keith

________________________________
From: Adam Roach [mailto:adam@nostrum.com]
Sent: Thursday, April 08, 2010 4:37 AM
To: DRAGE, Keith (Keith)
Cc: Paul Kyzivat; SIPCORE
Subject: Re: [sipcore] rfc3265bis: SIP events redux [was Minutes Posted]

On 4/7/10 20:16, Apr 7, DRAGE, Keith (Keith) wrote:

While I accept that RFC 4320 is mandatory for all current implementations c=
alling themselves SIP, we do need to deal with compatibility issues with de=
ployments made before it was thought of and became an RFC.

Thus if an RFC 3261/3265 sans RFC 4320 can send a 408 response, then any ot=
her implementation needs to be able to receive it.


Be able to receive it, yes. But, as RFC 4321 explains, the local transactio=
n timer will have already expired. The 408 will be treated as a stray respo=
nse. A well-designed SIP stack will discard it before it even reaches the T=
U. Even a poorly-designed but 3261-compliant application will ignore a 408 =
for a non-INVITE, because the transaction is already gone.

In other words, by the time the 408 arrives, it is thoroughly irrelevant. W=
e don't need to talk about it in 3265bis, because something else will have =
gone wrong first.

And that situation is already covered:

   If the NOTIFY request fails due to expiration of SIP Timer F
   (transaction timeout), the notifier SHOULD remove the subscription.


/a

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3DISO-8859-1"=
>
<META content=3D"MSHTML 6.00.2900.3660" name=3DGENERATOR></HEAD>
<BODY text=3D#000000 bgColor=3D#ffffff>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D516004214-08042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Your argument is accepted.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D516004214-08042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D516004214-08042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>That is an appropriate compatibility=20
analysis.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D516004214-08042010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2><BR>Keith</FONT></SPAN></DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Adam Roach [mailto:adam@nostrum=
.com]=20
  <BR><B>Sent:</B> Thursday, April 08, 2010 4:37 AM<BR><B>To:</B> DRAGE, Ke=
ith=20
  (Keith)<BR><B>Cc:</B> Paul Kyzivat; SIPCORE<BR><B>Subject:</B> Re: [sipco=
re]=20
  rfc3265bis: SIP events redux [was Minutes Posted]<BR></FONT><BR></DIV>
  <DIV></DIV>On 4/7/10 20:16, Apr 7, DRAGE, Keith (Keith) wrote:=20
  <BLOCKQUOTE=20
  cite=3Dmid:EDC0A1AE77C57744B664A310A0B23AE211CCD8B2@FRMRSSXCHMBSC3.dc-m.a=
lcatel-lucent.com=20
  type=3D"cite"><PRE wrap=3D"">While I accept that RFC 4320 is mandatory fo=
r all current implementations calling themselves SIP, we do need to deal wi=
th compatibility issues with deployments made before it was thought of and =
became an RFC.

Thus if an RFC 3261/3265 sans RFC 4320 can send a 408 response, then any ot=
her implementation needs to be able to receive it.
  </PRE></BLOCKQUOTE><BR>Be able to receive it, yes. But, as RFC 4321=20
  explains, the local transaction timer will have already expired. The 408 =
will=20
  be treated as a stray response. A well-designed SIP stack will discard it=
=20
  before it even reaches the TU. Even a poorly-designed but 3261-compliant=
=20
  application will ignore a 408 for a non-INVITE, because the transaction i=
s=20
  already gone.<BR><BR>In other words, by the time the 408 arrives, it is=20
  thoroughly irrelevant. We don't need to talk about it in 3265bis, because=
=20
  something else will have gone wrong first.<BR><BR>And that situation is=20
  already covered:<BR><PRE class=3Dnewpage>   If the NOTIFY request fails d=
ue to expiration of SIP Timer F
   (transaction timeout), the notifier SHOULD remove the subscription.
</PRE><BR>/a<BR></BLOCKQUOTE></BODY></HTML>

--_000_EDC0A1AE77C57744B664A310A0B23AE211CCDA0AFRMRSSXCHMBSC3d_--

From salvatore.loreto@ericsson.com  Thu Apr  8 08:09:46 2010
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi Paul,

the use case you have written down is exactly the one I have in mind and 
was going to send to this ml.

I want add one more benefit in knowing all the capabilities of the 
different Bob devices in advance and
not just of one device.
An OPTIONS response (or another mechanism!) that contains a level of 
granularity that let Alice (or rather Alices device)
know that Bob can take only audio on one device, but 
audio/video/text/picture on his other device (as describe in Paul use case),
would help Alice's device understand that starting a voice session with 
only voice Bob device, there is no way a re-INVITE to add video would work;
so if she thinks that a voice call could be then transformed in a 
voice+video, she knows that should contact directly the 
audio/video/text/picture device.

cheers
Sal


On 4/8/10 4:01 PM, Paul Kyzivat wrote:
> [as individual]
>
> It seems that Christer has come close to offering a use case, even if
> that was not his intent. And its still a little incomplete. I'll try to
> fill it out a bit:
>
> Alice wants to call Bob.
> Her phone app first queries Bob's AOR with OPTIONS.
> The response to that is a 3xx with Contact URIs for
> Bob's home phone and his mobile phone. So Alice's phone
> sends additional OPTIONS requests to each of those.
>   From the responses it learns that Bob's mobile phone support
> audio and video, while his home phone (behind a pstn gateway)
> only supports audio.
>
> Alice has video support on her phone, and since Bob may be able to
> support video too, decides to make an audio+video call. The
> phone sends an INVITE with both, and caller preferences indicating
> a preference for audio+video.
>
> The INVITE reaches the server handling Bob's AOR.
> It processes the callerprefs, and so decides to initially send the
> invite to Bob's mobile phone. But there is no answer there, so
> the server forks the invite toward Bob's home phone. Bob answers that
> one, accepting the audio and rejecting the video.
>
> The result of this is exactly the same as if no OPTIONS was ever sent,
> and the INVITE was sent indicating Alice's own preferences. That is the
> point that John already made.
>
> The only "benefit" I can see from use of options here would be if it was
> discovered that Bob had no video support. In that case, the invite could
> have been sent without offering video. Is that advantageous? Does it
> cost more to initiate a call that offers video? Does it take more
> network resources to do so, even when ultimately the video is rejected?
>
> The only real benefit I can see to this is if the calling user is given
> an enhanced user interface - akin to what one gets with buddy lists -
> where there is a display showing that Bob has two devices with differing
> capabilities, with Alice having the option to choose one or allowing the
> phone to make the choice.
>
> To provide that feature, presence would be a much better choice.
> OPTIONS is in a sense "poor man's presence". Christer has mentioned that
> he considers presence to be too "heavy weight" for this, but I remain to
> be convinced of that. Perhaps maintaining persistent subscriptions to
> presence status of everyone you might possibly call is too heavy weight,
> but doing a polling subscription prior to a call won't be any heavier
> than all this options stuff.
>
> It may be that lots of devices don't publish presence state. But that
> can be solved in many ways. A registrar can synthesize basic presence
> state from registration data. And it could even use OPTIONS to refine that.
>
> I suspect the real hidden issue here is about presence, and so I'd
> prefer to see it addressed from that direction. If that in turn leads to
> a desire to use OPTIONS to get some degree of presence state from
> devices that don't otherwise publish it, then lets confront it in that
> context. And if the issue is that presence is typically not authorized
> for as many requesters as OPTIONS is, then lets look at possible policy
> recommendations in that regard.
>
> 	Thanks,
> 	Paul
>
> Christer Holmberg wrote:
>    
>> Hi,
>>
>>      
>>> > From my perspective, the main use-case is not to discover services,
>>>        
>>>> but to discover capabilities in order to determine what services can be applied.
>>>>
>>>> The idea is also that the OPTIONS request would be sent prior to the
>>>> INVITE, to try to make sure that the returned contacts are actually
>>>> available.
>>>>          
>>> But you know that would never work in practice/the-real-world. Think about all the routing rules
>>> people have for INVITEs - in some cases *only* for INVITEs.
>>> And even if they route OPTIONS identically in the SIP domain,
>>> for any given AoR it'll go to any of: a SIP UA, or a PSTN
>>> gateway, or a H.323-gateway, or an MGCP call-agent, or an
>>> IP-PBX, or a foobar-protocol-gateway, or a vmail server, or
>>> an announcement server, or an app server, etc.  Just because
>>> the OPTIONS reaches the end of the SIP domain, means nothing
>>> about whether an INVITE to the higher-layer target (which in
>>> practice is a E.164 number, that crosses from SIP into many
>>> other protocol domains) is actually available to receive a call.
>>>        
>> The assumption is that the OPTIONS request reaches the registrar, that returns the registered contacts. It is never forwarded to any UAS.
>>
>> Then, at some point an INVITE is sent, and it is addressed to the same AOR (the INVITE request might be routed differently than the OPTIONS request). Based on the UAS information, the INVITE may now contain feature tags etc that indicates a wish to reach an UAS with certain capabilities.
>>
>> Now, unless the INVITE is addressed to a specific UAS, it can of course not be guranteed that the INVITE will actually reach such UAS. But, that is a "feature" of RFC 3841, and it would be the case even if you had never sent any OPTIONS. What OPTIONS allows provides you is capability information about the UAS(s) associated with the AOR, which might be useful when sending the INVITE.
>>
>> No, it does not guarantee that the user is available - it only provides you with that same information as OPTIONS in a non-forking case does.
>>
>> Regards,
>>
>> Christer
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>>      


-- 
Salvatore Loreto
www.sloreto.com


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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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FWIW, this is exactly what we do in XMPP.

On 4/8/10 9:08 AM, Salvatore Loreto wrote:
> Hi Paul,
>=20
> the use case you have written down is exactly the one I have in mind an=
d
> was going to send to this ml.
>=20
> I want add one more benefit in knowing all the capabilities of the
> different Bob devices in advance and
> not just of one device.
> An OPTIONS response (or another mechanism!) that contains a level of
> granularity that let Alice (or rather Alices device)
> know that Bob can take only audio on one device, but
> audio/video/text/picture on his other device (as describe in Paul use
> case),
> would help Alice's device understand that starting a voice session with=

> only voice Bob device, there is no way a re-INVITE to add video would w=
ork;
> so if she thinks that a voice call could be then transformed in a
> voice+video, she knows that should contact directly the
> audio/video/text/picture device.
>=20
> cheers
> Sal
>=20
>=20
> On 4/8/10 4:01 PM, Paul Kyzivat wrote:
>> [as individual]
>>
>> It seems that Christer has come close to offering a use case, even if
>> that was not his intent. And its still a little incomplete. I'll try t=
o
>> fill it out a bit:
>>
>> Alice wants to call Bob.
>> Her phone app first queries Bob's AOR with OPTIONS.
>> The response to that is a 3xx with Contact URIs for
>> Bob's home phone and his mobile phone. So Alice's phone
>> sends additional OPTIONS requests to each of those.
>>   From the responses it learns that Bob's mobile phone support
>> audio and video, while his home phone (behind a pstn gateway)
>> only supports audio.
>>
>> Alice has video support on her phone, and since Bob may be able to
>> support video too, decides to make an audio+video call. The
>> phone sends an INVITE with both, and caller preferences indicating
>> a preference for audio+video.
>>
>> The INVITE reaches the server handling Bob's AOR.
>> It processes the callerprefs, and so decides to initially send the
>> invite to Bob's mobile phone. But there is no answer there, so
>> the server forks the invite toward Bob's home phone. Bob answers that
>> one, accepting the audio and rejecting the video.
>>
>> The result of this is exactly the same as if no OPTIONS was ever sent,=

>> and the INVITE was sent indicating Alice's own preferences. That is th=
e
>> point that John already made.
>>
>> The only "benefit" I can see from use of options here would be if it w=
as
>> discovered that Bob had no video support. In that case, the invite cou=
ld
>> have been sent without offering video. Is that advantageous? Does it
>> cost more to initiate a call that offers video? Does it take more
>> network resources to do so, even when ultimately the video is rejected=
?
>>
>> The only real benefit I can see to this is if the calling user is give=
n
>> an enhanced user interface - akin to what one gets with buddy lists -
>> where there is a display showing that Bob has two devices with differi=
ng
>> capabilities, with Alice having the option to choose one or allowing t=
he
>> phone to make the choice.
>>
>> To provide that feature, presence would be a much better choice.
>> OPTIONS is in a sense "poor man's presence". Christer has mentioned th=
at
>> he considers presence to be too "heavy weight" for this, but I remain =
to
>> be convinced of that. Perhaps maintaining persistent subscriptions to
>> presence status of everyone you might possibly call is too heavy weigh=
t,
>> but doing a polling subscription prior to a call won't be any heavier
>> than all this options stuff.
>>
>> It may be that lots of devices don't publish presence state. But that
>> can be solved in many ways. A registrar can synthesize basic presence
>> state from registration data. And it could even use OPTIONS to refine
>> that.
>>
>> I suspect the real hidden issue here is about presence, and so I'd
>> prefer to see it addressed from that direction. If that in turn leads =
to
>> a desire to use OPTIONS to get some degree of presence state from
>> devices that don't otherwise publish it, then lets confront it in that=

>> context. And if the issue is that presence is typically not authorized=

>> for as many requesters as OPTIONS is, then lets look at possible polic=
y
>> recommendations in that regard.
>>
>>     Thanks,
>>     Paul
>>
>> Christer Holmberg wrote:
>>  =20
>>> Hi,
>>>
>>>    =20
>>>> > From my perspective, the main use-case is not to discover services=
,
>>>>      =20
>>>>> but to discover capabilities in order to determine what services
>>>>> can be applied.
>>>>>
>>>>> The idea is also that the OPTIONS request would be sent prior to th=
e
>>>>> INVITE, to try to make sure that the returned contacts are actually=

>>>>> available.
>>>>>         =20
>>>> But you know that would never work in practice/the-real-world. Think=

>>>> about all the routing rules
>>>> people have for INVITEs - in some cases *only* for INVITEs.
>>>> And even if they route OPTIONS identically in the SIP domain,
>>>> for any given AoR it'll go to any of: a SIP UA, or a PSTN
>>>> gateway, or a H.323-gateway, or an MGCP call-agent, or an
>>>> IP-PBX, or a foobar-protocol-gateway, or a vmail server, or
>>>> an announcement server, or an app server, etc.  Just because
>>>> the OPTIONS reaches the end of the SIP domain, means nothing
>>>> about whether an INVITE to the higher-layer target (which in
>>>> practice is a E.164 number, that crosses from SIP into many
>>>> other protocol domains) is actually available to receive a call.
>>>>       =20
>>> The assumption is that the OPTIONS request reaches the registrar,
>>> that returns the registered contacts. It is never forwarded to any UA=
S.
>>>
>>> Then, at some point an INVITE is sent, and it is addressed to the
>>> same AOR (the INVITE request might be routed differently than the
>>> OPTIONS request). Based on the UAS information, the INVITE may now
>>> contain feature tags etc that indicates a wish to reach an UAS with
>>> certain capabilities.
>>>
>>> Now, unless the INVITE is addressed to a specific UAS, it can of
>>> course not be guranteed that the INVITE will actually reach such UAS.=

>>> But, that is a "feature" of RFC 3841, and it would be the case even
>>> if you had never sent any OPTIONS. What OPTIONS allows provides you
>>> is capability information about the UAS(s) associated with the AOR,
>>> which might be useful when sending the INVITE.
>>>
>>> No, it does not guarantee that the user is available - it only
>>> provides you with that same information as OPTIONS in a non-forking
>>> case does.
>>>
>>> Regards,
>>>
>>> Christer
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>


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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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[as individual]

Salvatore Loreto wrote:
> Hi Paul,
> 
> the use case you have written down is exactly the one I have in mind and 
> was going to send to this ml.
> 
> I want add one more benefit in knowing all the capabilities of the 
> different Bob devices in advance and
> not just of one device.
> An OPTIONS response (or another mechanism!) that contains a level of 
> granularity that let Alice (or rather Alices device)
> know that Bob can take only audio on one device, but 
> audio/video/text/picture on his other device (as describe in Paul use 
> case),
> would help Alice's device understand that starting a voice session with 
> only voice Bob device, there is no way a re-INVITE to add video would work;
> so if she thinks that a voice call could be then transformed in a 
> voice+video, she knows that should contact directly the 
> audio/video/text/picture device.

If this kind of feature is desired, then lets spell that out and discuss 
alternative ways of achieving that result.

The original sip philosophy was that choice of device to respond to a 
request is a policy decision to be made by the server responsible for 
the AOR. It might take the wishes of the caller into account but that is 
still based on callee policy.

Presence turns that on its head. In the presence model, the "caller" is 
given a menu by the callee, and allowed to choose. Since that is what 
you seem to wish, I think presence is where to start in looking for a 
solution.

As an aside, if I have a video device that can be added into the current 
(audio) call, then *I* want to be the one that makes that decision. I 
find the idea of having the caller do it a bit creepy.

	Thanks,
	Paul

> cheers
> Sal
> 
> 
> On 4/8/10 4:01 PM, Paul Kyzivat wrote:
>> [as individual]
>>
>> It seems that Christer has come close to offering a use case, even if
>> that was not his intent. And its still a little incomplete. I'll try to
>> fill it out a bit:
>>
>> Alice wants to call Bob.
>> Her phone app first queries Bob's AOR with OPTIONS.
>> The response to that is a 3xx with Contact URIs for
>> Bob's home phone and his mobile phone. So Alice's phone
>> sends additional OPTIONS requests to each of those.
>>   From the responses it learns that Bob's mobile phone support
>> audio and video, while his home phone (behind a pstn gateway)
>> only supports audio.
>>
>> Alice has video support on her phone, and since Bob may be able to
>> support video too, decides to make an audio+video call. The
>> phone sends an INVITE with both, and caller preferences indicating
>> a preference for audio+video.
>>
>> The INVITE reaches the server handling Bob's AOR.
>> It processes the callerprefs, and so decides to initially send the
>> invite to Bob's mobile phone. But there is no answer there, so
>> the server forks the invite toward Bob's home phone. Bob answers that
>> one, accepting the audio and rejecting the video.
>>
>> The result of this is exactly the same as if no OPTIONS was ever sent,
>> and the INVITE was sent indicating Alice's own preferences. That is the
>> point that John already made.
>>
>> The only "benefit" I can see from use of options here would be if it was
>> discovered that Bob had no video support. In that case, the invite could
>> have been sent without offering video. Is that advantageous? Does it
>> cost more to initiate a call that offers video? Does it take more
>> network resources to do so, even when ultimately the video is rejected?
>>
>> The only real benefit I can see to this is if the calling user is given
>> an enhanced user interface - akin to what one gets with buddy lists -
>> where there is a display showing that Bob has two devices with differing
>> capabilities, with Alice having the option to choose one or allowing the
>> phone to make the choice.
>>
>> To provide that feature, presence would be a much better choice.
>> OPTIONS is in a sense "poor man's presence". Christer has mentioned that
>> he considers presence to be too "heavy weight" for this, but I remain to
>> be convinced of that. Perhaps maintaining persistent subscriptions to
>> presence status of everyone you might possibly call is too heavy weight,
>> but doing a polling subscription prior to a call won't be any heavier
>> than all this options stuff.
>>
>> It may be that lots of devices don't publish presence state. But that
>> can be solved in many ways. A registrar can synthesize basic presence
>> state from registration data. And it could even use OPTIONS to refine 
>> that.
>>
>> I suspect the real hidden issue here is about presence, and so I'd
>> prefer to see it addressed from that direction. If that in turn leads to
>> a desire to use OPTIONS to get some degree of presence state from
>> devices that don't otherwise publish it, then lets confront it in that
>> context. And if the issue is that presence is typically not authorized
>> for as many requesters as OPTIONS is, then lets look at possible policy
>> recommendations in that regard.
>>
>>     Thanks,
>>     Paul
>>
>> Christer Holmberg wrote:
>>   
>>> Hi,
>>>
>>>     
>>>> > From my perspective, the main use-case is not to discover services,
>>>>       
>>>>> but to discover capabilities in order to determine what services 
>>>>> can be applied.
>>>>>
>>>>> The idea is also that the OPTIONS request would be sent prior to the
>>>>> INVITE, to try to make sure that the returned contacts are actually
>>>>> available.
>>>>>          
>>>> But you know that would never work in practice/the-real-world. Think 
>>>> about all the routing rules
>>>> people have for INVITEs - in some cases *only* for INVITEs.
>>>> And even if they route OPTIONS identically in the SIP domain,
>>>> for any given AoR it'll go to any of: a SIP UA, or a PSTN
>>>> gateway, or a H.323-gateway, or an MGCP call-agent, or an
>>>> IP-PBX, or a foobar-protocol-gateway, or a vmail server, or
>>>> an announcement server, or an app server, etc.  Just because
>>>> the OPTIONS reaches the end of the SIP domain, means nothing
>>>> about whether an INVITE to the higher-layer target (which in
>>>> practice is a E.164 number, that crosses from SIP into many
>>>> other protocol domains) is actually available to receive a call.
>>>>        
>>> The assumption is that the OPTIONS request reaches the registrar, 
>>> that returns the registered contacts. It is never forwarded to any UAS.
>>>
>>> Then, at some point an INVITE is sent, and it is addressed to the 
>>> same AOR (the INVITE request might be routed differently than the 
>>> OPTIONS request). Based on the UAS information, the INVITE may now 
>>> contain feature tags etc that indicates a wish to reach an UAS with 
>>> certain capabilities.
>>>
>>> Now, unless the INVITE is addressed to a specific UAS, it can of 
>>> course not be guranteed that the INVITE will actually reach such UAS. 
>>> But, that is a "feature" of RFC 3841, and it would be the case even 
>>> if you had never sent any OPTIONS. What OPTIONS allows provides you 
>>> is capability information about the UAS(s) associated with the AOR, 
>>> which might be useful when sending the INVITE.
>>>
>>> No, it does not guarantee that the user is available - it only 
>>> provides you with that same information as OPTIONS in a non-forking 
>>> case does.
>>>
>>> Regards,
>>>
>>> Christer
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>
>>>      
> 
> 

From HKaplan@acmepacket.com  Thu Apr  8 10:33:32 2010
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To: Paul Kyzivat <pkyzivat@cisco.com>, Christer Holmberg <christer.holmberg@ericsson.com>
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Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Thursday, April 08, 2010 10:01 AM
>=20
> The only "benefit" I can see from use of options here would be if it was
> discovered that Bob had no video support. In that case, the invite could
> have been sent without offering video. Is that advantageous? Does it
> cost more to initiate a call that offers video? Does it take more
> network resources to do so, even when ultimately the video is rejected?

Well... when you put it that way, there is actually a small advantage.  In =
many resource reservation deployments the INVITE would be rejected if there=
 isn't sufficient bandwidth for the highest-bitrate codec or the aggregate =
of all offered medias.  Afaict this is due to either bad design, or the pro=
tocols being used to policy servers which can't really handle an if-else-el=
se type reservation model.

So in theory the INVITE would succeed had Alice only asked for audio.

But there are so many other policy decisions involved that it's a crap-shoo=
t anyway, no matter how much info you give the UAC.

-hadriel

From adam@nostrum.com  Thu Apr  8 11:07:01 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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On 4/7/10 7:08 AM, Brett Tate wrote:
>
>    
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: Tuesday, April 06, 2010 7:37 PM
>> To: Brett Tate
>> Cc: Paul Kyzivat; SIPCORE
>> Subject: Re: [sipcore] rfc3265bis: SIP events redux [was Minutes
>> Posted]
>>
>> On 4/6/10 17:46, Apr 6, Brett Tate wrote:
>>      
>>>> Is there *any* problem here? This seems to be working about
>>>> as well as anyone might hope.
>>>>
>>>>          
>>> It works okay excluding the non backward compatibility with RFC 3265
>>>        
>> (concerning SUBSCRIBE's 2xx response impacts)...
>>
>> I don't think it's accurate to characterize the change as
>> non-backwards-compatible.
>>      
> RFC 3265 section 3.3.4 indicates that 2xx "creates a new subscription and a new dialog"; rfc3265bis does not allow such a thing.
>    

Yes, it's a *change*, but it's a *backwards* *compatible* change.

/a

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Subject: Re: [sipcore] rfc3265bis: SIP events redux [was  Minutes Posted]
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On 4/7/10 7:26 AM, Brett Tate wrote:
>> Then what would happen in a polling subscribe (Expires:0) if
>> the 200 is received and the NOTIFY is not received? How long
>> should you wait before giving up? ISTM you still need timer-N
>> to tell you it failed, so you can give up waiting.
>>      
> I agree that you need a Timer N (or something like it); however Timer N should not be restricted to 64*T1.  The rfc3265bis should accommodate other Timer N values instead being restricted to 64*T1 for all event packages, deployments, situations, etcetera.
>    

Sure, we can enter into a discussion of how to set Timer N. But, to fix 
the problems that we had in 3265, any changes really need to be 
predicated on two qualities:

   1. Everyone involved in the transaction needs to know what value
      Timer N has -- otherwise, you can easily end up with differing
      ideas of dialog and subscription state.

   2. Variation in Timer N can't be based on a mechanism to withhold the
      initial NOTIFY message for an arbitrary (or even a relatively
      long) period of time. NOTIFY is an important part of the 3-way
      handshake used to set up a dialog, and extensions can reasonable
      make assumptions around this completing in a reasonable amount of
      time (3265 defines this as "immediately").

/a

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On 4/7/10 7:26 AM, Brett Tate wrote:
<blockquote
 cite="mid:747A6506A991724FB09B129B79D5FEB61480C55EC4@EXMBXCLUS01.citservers.local"
 type="cite">
  <blockquote type="cite">
    <pre wrap="">Then what would happen in a polling subscribe (Expires:0) if 
the 200 is received and the NOTIFY is not received? How long 
should you wait before giving up? ISTM you still need timer-N 
to tell you it failed, so you can give up waiting.
    </pre>
  </blockquote>
  <pre wrap="">
I agree that you need a Timer N (or something like it); however Timer N should not be restricted to 64*T1.  The rfc3265bis should accommodate other Timer N values instead being restricted to 64*T1 for all event packages, deployments, situations, etcetera.
  </pre>
</blockquote>
<br>
Sure, we can enter into a discussion of how to set Timer N. But, to fix
the problems that we had in 3265, any changes really need to be
predicated on two qualities:<br>
<ol>
  <li>Everyone involved in the transaction needs to know what value
Timer N has -- otherwise, you can easily end up with differing ideas of
dialog and subscription state.<br>
    <br>
  </li>
  <li>Variation in Timer N can't be based on a mechanism to withhold
the initial NOTIFY message for an arbitrary (or even a relatively long)
period of time. NOTIFY is an important part of the 3-way handshake used
to set up a dialog, and extensions can reasonable make assumptions
around this completing in a reasonable amount of time (3265 defines
this as "immediately").</li>
</ol>
/a<br>
</body>
</html>

--------------060405030809060004080001--

From adam@nostrum.com  Thu Apr  8 11:14:51 2010
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Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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On 4/8/10 1:48 AM, Christer Holmberg wrote:
> Hi,
>
> Also, both 3265 and 3265bis contains the following text:
>
> "A subscriber may send a SUBSCRIBE request with an "Expires" header
> field of 0 in order to trigger the sending of such a NOTIFY
> request; however, for the purposes of subscription and dialog
> lifetime, the subscription is not considered terminated until the
> NOTIFY transaction with a "Subscription-State" of "terminated"
> completes."
>    

Yes. The fixes to specifically accommodate subnot-etags are not in the 
current version of the document. This will all be addressed in a future 
version of the draft.

/a

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Subject: Re: [sipcore] RFC 4320: deprecation of non-INVITE 408 responses
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[as chair]

On 4/8/10 7:36 AM, Brett Tate wrote:
> Since it was raised within rfc3265bis discussion, I thought that I'd attempt to get clarity concerning RFC 4320's deprecation of non-INVITE 408 responses in case errata should be opened against RFC 4320.
>
> RFC 4320 intentionally or unintentionally appears to have deprecated all non-INVITE 408 responses.

It was intentional, AFAIK.

>    However I thought that I'd still ask, is the deprecation only applicable for Timer F expiration type situations (i.e. the reason for the deprecation)?
>    

That's a good question -- perhaps Robert can weigh in on this.



> If not limited to Timer F expiration type situations, such a deprecation causes some ambiguity concerning shorter timeouts on other interfaces.  For instance, a received SUBSCRIBE is transformed into another protocol's FOO message.  The FOO transaction delivery expires after 1 second.  Can a 408 response (although 504 might be better per rfc3261) be sent for the SUBSCRIBE?  What about a B2BUA which using a short expiration (like 1 second) on the send side; can it send a 408 response or must it really start sending 504 or other response instead?
>    


/a

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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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[as individual]

On 4/8/10 9:01 AM, Paul Kyzivat wrote:
> [as individual]
>
> It seems that Christer has come close to offering a use case, even if 
> that was not his intent. And its still a little incomplete. I'll try 
> to fill it out a bit:
>
> ...
>
> To provide that feature, presence would be a much better choice.

*ding* *ding* *ding*

We have an answer!

That's why I'm asking for use cases. Every use case *I* can come up with 
is better suited for something like presence, callerprefs, etc. Now, 
Christer clearly has some problem he's trying to solve. If he'd just 
describe the problem in terms of a use case (instead of "forking breaks 
OPTIONS"), then I think we could make some progress.

I'm not saying that the answer won't be OPTIONS, simply that a concise 
statement of the use case(s) that need to be supported is likely to make 
it clearer to everyone what kind of solution is actually called for.

/a

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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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[as individual]

On 4/8/10 9:17 AM, Paul Kyzivat wrote:
> [as individual]
>
> Christer Holmberg wrote:
>> Hi,
>>
>> In the case of DE-subscriptions (which is just one variant of 
>> re-subscription), I still think there are some issues regarding 
>> whether the SUB 200 or the NOT req acknowledges the SUB req.
>>
>> Assuming the sub dialog is terminated when an entity receives a SUB 
>> 200 that contains 'Expires: 0'.
>>
>> Now, if there is then a NOT request (that contains 
>> 'Subscription-State: terminated'), won't it be treated as an 
>> out-of-dialog request, and rejected?
>
> That scenario has always bothered me. When my purpose is simply to 
> request the termination of the existing dialog, and I have no further 
> interest in the state, is there any reason for the NOTIFY? Can it be 
> suppressed?

Yes. RFC 5839 (which will be published shortly) -- more widely known as 
subnot-etags -- will provide exactly the mechanism to perform this 
suppression.

/a

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Subject: Re: [sipcore] RFC 4320: deprecation of non-INVITE 408 responses
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On Apr 8, 2010, at 1:21 PM, Adam Roach wrote:

> [as chair]
>=20
> On 4/8/10 7:36 AM, Brett Tate wrote:
>> Since it was raised within rfc3265bis discussion, I thought that I'd =
attempt to get clarity concerning RFC 4320's deprecation of non-INVITE =
408 responses in case errata should be opened against RFC 4320.
>>=20
>> RFC 4320 intentionally or unintentionally appears to have deprecated =
all non-INVITE 408 responses.
>=20
> It was intentional, AFAIK.

It was quite intentional.

>=20
>>   However I thought that I'd still ask, is the deprecation only =
applicable for Timer F expiration type situations (i.e. the reason for =
the deprecation)?
>>  =20
>=20
> That's a good question -- perhaps Robert can weigh in on this.
408's semantics in 3261 are clearly bound to the transaction machinery. =
The other cases you are describing should use a different response code.

>=20
>=20
>=20
>> If not limited to Timer F expiration type situations, such a =
deprecation causes some ambiguity concerning shorter timeouts on other =
interfaces.  For instance, a received SUBSCRIBE is transformed into =
another protocol's FOO message.  The FOO transaction delivery expires =
after 1 second.  Can a 408 response (although 504 might be better per =
rfc3261) be sent for the SUBSCRIBE?  What about a B2BUA which using a =
short expiration (like 1 second) on the send side; can it send a 408 =
response or must it really start sending 504 or other response instead?
>>  =20
>=20
>=20
> /a


--Apple-Mail-26-387794322
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On Apr 8, 2010, at 1:21 PM, Adam Roach wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div>[as =
chair]<br><br>On 4/8/10 7:36 AM, Brett Tate wrote:<br><blockquote =
type=3D"cite">Since it was raised within rfc3265bis discussion, I =
thought that I'd attempt to get clarity concerning RFC 4320's =
deprecation of non-INVITE 408 responses in case errata should be opened =
against RFC 4320.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">RFC 4320 =
intentionally or unintentionally appears to have deprecated all =
non-INVITE 408 responses.<br></blockquote><br>It was intentional, =
AFAIK.<br></div></blockquote><div><br></div>It was quite =
intentional.</div><div><br><blockquote type=3D"cite"><div><br><blockquote =
type=3D"cite"> &nbsp;&nbsp;However I thought that I'd still ask, is the =
deprecation only applicable for Timer F expiration type situations (i.e. =
the reason for the deprecation)?<br></blockquote><blockquote =
type=3D"cite"> &nbsp;&nbsp;<br></blockquote><br>That's a good question =
-- perhaps Robert can weigh in on this.<br></div></blockquote>408's =
semantics in 3261 are clearly bound to the transaction machinery. The =
other cases you are describing should use a different response =
code.</div><div><br><blockquote type=3D"cite"><div><font =
class=3D"Apple-style-span" =
color=3D"#000000"><br></font><br><br><blockquote type=3D"cite">If not =
limited to Timer F expiration type situations, such a deprecation causes =
some ambiguity concerning shorter timeouts on other interfaces. =
&nbsp;For instance, a received SUBSCRIBE is transformed into another =
protocol's FOO message. &nbsp;The FOO transaction delivery expires after =
1 second. &nbsp;Can a 408 response (although 504 might be better per =
rfc3261) be sent for the SUBSCRIBE? &nbsp;What about a B2BUA which using =
a short expiration (like 1 second) on the send side; can it send a 408 =
response or must it really start sending 504 or other response =
instead?<br></blockquote><blockquote type=3D"cite"> =
&nbsp;&nbsp;<br></blockquote><br><br>/a<br></div></blockquote></div><br></=
body></html>=

--Apple-Mail-26-387794322--

From christer.holmberg@ericsson.com  Thu Apr  8 12:32:18 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Thu, 8 Apr 2010 21:31:54 +0200
Thread-Topic: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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Hi,

>>> In the case of DE-subscriptions (which is just one variant of
>>> re-subscription), I still think there are some issues regarding
>>> whether the SUB 200 or the NOT req acknowledges the SUB req.
>>>
>>> Assuming the sub dialog is terminated when an entity receives a SUB
>>> 200 that contains 'Expires: 0'.
>>>
>>> Now, if there is then a NOT request (that contains
>>> 'Subscription-State: terminated'), won't it be treated as an
>>> out-of-dialog request, and rejected?
>>
>> That scenario has always bothered me. When my purpose is simply to
>> request the termination of the existing dialog, and I have no further
>> interest in the state, is there any reason for the NOTIFY? Can it be
>> suppressed?
>
>Yes. RFC 5839 (which will be published shortly) -- more widely known as
>subnot-etags -- will provide exactly the mechanism to perform this
>suppression.

But again, stateful intermediaries that do not support subnot will wait for=
 the NOTIFY.

Regards,

Christer=

From christer.holmberg@ericsson.com  Thu Apr  8 12:34:40 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Thu, 8 Apr 2010 21:32:22 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi Hadriel,

I don't think documenting the O-3xx mechanism would require much work. And,=
 as some have indicated, it might even be done as an individual draft.

Then, if we don't think OPTIONS is good enough in the first place, and want=
 to define some other procedure, which actually requires some work, the sit=
uation is different. But, I have not suggested that.

Regards,

Christer

________________________________________
From: Hadriel Kaplan [HKaplan@acmepacket.com]
Sent: Thursday, April 08, 2010 1:44 PM
To: Christer Holmberg; Elwell, John; Hans Erik van Elburg
Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
Subject: RE: [sipcore] SIP OPTIONS: Shall we work on this?

Hi Christer,
I know you're frustrated we keep going back to a use-case, but you're askin=
g for consensus on whether people are interested in working on the solution=
.  Without an actual use-case, how can we do that?  Why would people agree =
to spend time on something they don't see any use for solving?

-hadriel

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Christer Holmberg
> Sent: Thursday, April 08, 2010 6:13 AM
> To: Elwell, John; Hans Erik van Elburg
> Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>
>
> Hi,
>
> >[JRE] But even if you then target the INVITE request at the
> >GRUU of the preferred user, there is no guarantee it will
> >succeed (e.g., no media resources, UA has become busy in the
> >meantime), and because it is targeted directly, you lose the
> >opportunity to take alternative action, e.g., forwarding to
> >voicemail or to another UA with similar capabilities. It is a
> >case of the calling user trying to override policies
> >associated with the called user, which I think is a bad
> >thing. The result of the OPTIONS request can only ever act as
> >a guideline - it is effectively no better than presence.
> >
> >It would be helpful to have a concrete example of something
> >that could go into an INVITE request targeted at a particular
> >UA known to have a particular capability, that could not go
> >into an INVITE request targeted at the AOR and with caller
> >preferences RECOMMENDING selection of a UA with that
> >particular capability. Apologies if such an example has
> >already been given earlier in the discussion.
>
> I don't think I have said that the INVITE would necessarily be targeted t=
o
> a particular UA. There MAY of course be use-cases where you would target
> also the INVITE to a specific UAS. For example, if an UAS supports a
> specific capability, and you don't want the call to proceed unless that
> UAS accepts the INVITE. Then, if the INVITE fails, it may still be
> possible for the call to be forwarded to a voice mail etc, depending on
> policies and service design in the network.
>
> But, the main point of the O-3xx mechanisms is that it allows you to
> target *OPTIONS* request to particular UAs, in order to get the
> capabilities of those.
>
> That is the only thing the mechanism is supposed to provide. What you the=
n
> do with that information is a separate issue, and that seems to be what
> the discussion is now about.
>
> Now, we can discuss use-cases etc, but I just want to make it clear that
> the O-3xx mechanism itself does not make any assumptions regarding use-
> cases. It only allows the usage of OPTIONS in forking scenarios - nothing
> more, nothing less. Apart from providing a way to deal with the forking
> issue, it does not solve any other issues, bugs or shortcomings there may
> be with OPTIONS.
>
> Regards,
>
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore=

From brett@broadsoft.com  Thu Apr  8 12:46:59 2010
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From: Brett Tate <brett@broadsoft.com>
To: Robert Sparks <rjsparks@nostrum.com>, Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Thu, 8 Apr 2010 12:46:06 -0700
Thread-Topic: [sipcore] RFC 4320: deprecation of non-INVITE 408 responses
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> 408's semantics in 3261 are clearly bound to the transaction=20
> machinery. The other cases you are describing should use a=20
> different response code.

If 408's semantics were so clear, I'd think RFC 3398, B2BUAs, and etcetera =
would not have generated or documented to generate a 408 response based upo=
n another interfaces timeouts or receipt of 408.

Should errata be opened against RFC 3398?

Thanks for the response,

Brett


From pkyzivat@cisco.com  Thu Apr  8 12:57:37 2010
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Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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[as individual]

Its good that subnot-etags will help with this.
But support for that will be optional, and it may not even make sense 
for some event packages. Yet not wanting the notify when ending the 
subscription can make sense for all event packages.

Is there a way that packages that support subnot-etags just for the "I 
don't want to see anything" case, without the complexity and bother of 
supporting for anything else?

	Thanks,
	Paul

Adam Roach wrote:
> [as individual]
> 
> On 4/8/10 9:17 AM, Paul Kyzivat wrote:
>> [as individual]
>>
>> Christer Holmberg wrote:
>>> Hi,
>>>
>>> In the case of DE-subscriptions (which is just one variant of 
>>> re-subscription), I still think there are some issues regarding 
>>> whether the SUB 200 or the NOT req acknowledges the SUB req.
>>>
>>> Assuming the sub dialog is terminated when an entity receives a SUB 
>>> 200 that contains 'Expires: 0'.
>>>
>>> Now, if there is then a NOT request (that contains 
>>> 'Subscription-State: terminated'), won't it be treated as an 
>>> out-of-dialog request, and rejected?
>>
>> That scenario has always bothered me. When my purpose is simply to 
>> request the termination of the existing dialog, and I have no further 
>> interest in the state, is there any reason for the NOTIFY? Can it be 
>> suppressed?
> 
> Yes. RFC 5839 (which will be published shortly) -- more widely known as 
> subnot-etags -- will provide exactly the mechanism to perform this 
> suppression.
> 
> /a
> 

From adam@nostrum.com  Thu Apr  8 13:02:34 2010
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Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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[as individual]

On 4/8/10 2:57 PM, Paul Kyzivat wrote:
> [as individual]
>
> Its good that subnot-etags will help with this.
> But support for that will be optional...

Which is pretty much true of anything we define after the fact. Horses 
have left the barn, etc.

> ...and it may not even make sense for some event packages.

I think you'll have a hard time coming up with an example of an entire 
event package for which etags doesn't make sense.

> Is there a way that packages that support subnot-etags just for the "I 
> don't want to see anything" case, without the complexity and bother of 
> supporting for anything else?

Not for the server. The client can blindly do a "Suppress-If-Match: *" 
on unsubscribe and not worry about the etag mechanism otherwise.

/a

From pkyzivat@cisco.com  Thu Apr  8 13:13:56 2010
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Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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[as individual]

Christer Holmberg wrote:
> But again, stateful intermediaries that do not support subnot will wait for the NOTIFY.

Are you aware of any *proxies* that are dialog stateful for sub/not?
(Its not an issue for B2BUAs since they must support it for the two ends 
to use it.)

If so, is there really some good reason for them to be doing so?

	Thanks,
	Paul

From christer.holmberg@ericsson.com  Thu Apr  8 13:14:20 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Thu, 8 Apr 2010 22:13:45 +0200
Thread-Topic: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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Hi,

IF we in 3265bis are going to say (see earlier discussion) that the SUB 200=
 (rather than the NOT) acknowledges a RE-subscription, is there are reason =
why we can't say the same thing for a DE-subscription?

Regards,

Christer

________________________________________
From: Adam Roach [adam@nostrum.com]
Sent: Thursday, April 08, 2010 11:02 PM
To: Paul Kyzivat
Cc: Christer Holmberg; sipcore@ietf.org
Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-=
)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]

[as individual]

On 4/8/10 2:57 PM, Paul Kyzivat wrote:
> [as individual]
>
> Its good that subnot-etags will help with this.
> But support for that will be optional...

Which is pretty much true of anything we define after the fact. Horses
have left the barn, etc.

> ...and it may not even make sense for some event packages.

I think you'll have a hard time coming up with an example of an entire
event package for which etags doesn't make sense.

> Is there a way that packages that support subnot-etags just for the "I
> don't want to see anything" case, without the complexity and bother of
> supporting for anything else?

Not for the server. The client can blindly do a "Suppress-If-Match: *"
on unsubscribe and not worry about the etag mechanism otherwise.

/a=

From christer.holmberg@ericsson.com  Thu Apr  8 13:21:30 2010
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To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Thu, 8 Apr 2010 22:21:08 +0200
Thread-Topic: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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Hi,

>>But again, stateful intermediaries that do not support subnot will wait f=
or the NOTIFY.
>
>Are you aware of any *proxies* that are dialog stateful for sub/not?

Yes, I am. They insert Record-Route (which is explicitly allowed), and main=
tain the subscription dialog state.

>(Its not an issue for B2BUAs since they must support it for the two ends t=
o use it.)

Correct. I am talking about proxies.

>If so, is there really some good reason for them to be doing so?

That information I unfortunately don't have at the moment.

Regards,

Christer=

From pkyzivat@cisco.com  Thu Apr  8 13:24:49 2010
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Rfc3265bis: NOTIFY request or response confirms (re-)subscription? [was rfc3265bis: SIP events redux [was Minutes Posted]]
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Adam Roach wrote:
> [as individual]
> 
> On 4/8/10 2:57 PM, Paul Kyzivat wrote:
>> [as individual]
>>
>> Its good that subnot-etags will help with this.
>> But support for that will be optional...
> 
> Which is pretty much true of anything we define after the fact. Horses 
> have left the barn, etc.
> 
>> ...and it may not even make sense for some event packages.
> 
> I think you'll have a hard time coming up with an example of an entire 
> event package for which etags doesn't make sense.

I'm no expert on this, but its my impression that it is *work* for the 
server developer to assign etags and then handle the suppression based 
on them. I'm thinking a lot of implementors would rather not bother to 
do that, and will only do so if the event package spec says they must. 
(Which none of the current ones do.)

Can a server never send etags in NOTIFY, yet still support 
"Suppress-If-Match: *" in SUBSCRIBE?

	Thanks,
	Paul

>> Is there a way that packages that support subnot-etags just for the "I 
>> don't want to see anything" case, without the complexity and bother of 
>> supporting for anything else?
> 
> Not for the server. The client can blindly do a "Suppress-If-Match: *" 
> on unsubscribe and not worry about the etag mechanism otherwise.
> 
> /a
> 

From rjsparks@nostrum.com  Thu Apr  8 13:33:08 2010
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Subject: Re: [sipcore] RFC 4320: deprecation of non-INVITE 408 responses
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We're talking about the non-INVITE transaction.
I think what you're looking at in 3398 is INVITE oriented (408's for =
INVITE have a broader meaning).

RjS

On Apr 8, 2010, at 2:46 PM, Brett Tate wrote:

>> 408's semantics in 3261 are clearly bound to the transaction=20
>> machinery. The other cases you are describing should use a=20
>> different response code.
>=20
> If 408's semantics were so clear, I'd think RFC 3398, B2BUAs, and =
etcetera would not have generated or documented to generate a 408 =
response based upon another interfaces timeouts or receipt of 408.
>=20
> Should errata be opened against RFC 3398?
>=20
> Thanks for the response,
>=20
> Brett
>=20


From dean.willis@softarmor.com  Thu Apr  8 15:42:14 2010
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On Apr 8, 2010, at 8:39 AM, Elwell, John wrote:
>
> In either case, nobody seems to have given a convincing use case  
> where probing all the devices behind an AOR is of significant benefit,

That's indeed the question.

What would you do differently if you knew about the capabilities of  
all the potential responding UAs before sending an INVITE?

Our traditional logic has been "put everything YOU support in the  
offer, and let the combination of RFC 3840/3841 working in the proxy  
pick a UA that best-matches what you have. Of course, this can  
sometimes lead to requests getting sent to the phone that is the  
closest feature match rather than the one that's closest to the called  
party.

Another alternative is to make an INVITE that offers only the bare  
minimum, then explore upgrading the call once it is established to  
engage additional features.


But perhaps we have a problem. I want to offer a "rich" INVITE, but  
the low-end client with no features keeps answering before the  
"better" box. Now, we need to know the "best" set of capabilities, so  
that we can construct an INVITE that only the "right" UA is going to  
answer.


> and if there is a significant benefit, why you can't just obtain the  
> device contact URIs using the reg event package and then send  
> OPTIONS to each.

That one's easy. I'm not letting you subscribe to my reg-event  
information. I don't even HAVE reg-event information, as my registrar  
doesn't do it.

--
Dean


From fluffy@cisco.com  Thu Apr  8 17:00:43 2010
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On Apr 7, 2010, at 7:49 , Adam Roach wrote:

> I think Hadriel is right.
>=20
> The more I read through this thread, the more it looks like we're =
running around with a hammer, desperately looking for nails to hit. =
Barring nails, we'll settle for screws. Or bolts. Or maybe kittens, if =
it comes to that.


+1 - I have a potato peeler and I'm not afraid to use it. But I'm not =
seeing a problem here that OPTIONS looks like it solves.=20



Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html



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To: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On Apr 8, 2010, at 15:41 , Dean Willis wrote:

>=20
> What would you do differently if you knew about the capabilities of =
all the potential responding UAs before sending an INVITE?

Well, one things I might do if I knew the capability of the far end =
before the INVITE is not do Offer/answer and instead just tell the far =
end what to do. More like how SDP is used in RTSP. All this offer answer =
stuff is hard to get right. Some times I wonder if the use case is =
actually it's hard to get offer / answer right so really what people =
want to send is an "offer you can't refuse" in the INVITE and are trying =
to discover capability's to make that possible. All a very interesting =
discussion but none of this has inspired me to start work on anything to =
do with OPTIONS.=20




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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Dean Willis wrote:
> 
> But perhaps we have a problem. I want to offer a "rich" INVITE, but the 
> low-end client with no features keeps answering before the "better" box. 
> Now, we need to know the "best" set of capabilities, so that we can 
> construct an INVITE that only the "right" UA is going to answer.

I would recommend that you then set up callerprefs to *prefer* (but not 
require) the "rich" capabilities. With luck the call will be offered to 
the rich device first, and if that one doesn't answer then the dumber 
ones will still be tried.

	Thanks,
	Paul

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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Cullen Jennings wrote:
> On Apr 7, 2010, at 7:49 , Adam Roach wrote:
> 
>> I think Hadriel is right.
>>
>> The more I read through this thread, the more it looks like we're running around with a hammer, desperately looking for nails to hit. Barring nails, we'll settle for screws. Or bolts. Or maybe kittens, if it comes to that.
> 
> 
> +1 - I have a potato peeler and I'm not afraid to use it. But I'm not seeing a problem here that OPTIONS looks like it solves. 

Maybe we could find a kitten for you to peel with it. :-)

	Thanks,
	Paul

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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On Apr 8, 2010, at 7:06 PM, Paul Kyzivat wrote:

>
>
> Dean Willis wrote:
>> But perhaps we have a problem. I want to offer a "rich" INVITE, but  
>> the low-end client with no features keeps answering before the  
>> "better" box. Now, we need to know the "best" set of capabilities,  
>> so that we can construct an INVITE that only the "right" UA is  
>> going to answer.
>
> I would recommend that you then set up callerprefs to *prefer* (but  
> not require) the "rich" capabilities. With luck the call will be  
> offered to the rich device first, and if that one doesn't answer  
> then the dumber ones will still be tried.
>

Do you feel lucky?

--
Dean

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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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=20

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: 08 April 2010 23:42
> To: Elwell, John
> Cc: Hans Erik van Elburg; WORLEY, DALE R (DALE); sipcore@ietf.org
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
>=20
> On Apr 8, 2010, at 8:39 AM, Elwell, John wrote:
> >
> > In either case, nobody seems to have given a convincing use case =20
> > where probing all the devices behind an AOR is of=20
> significant benefit,
>=20
> That's indeed the question.
>=20
> What would you do differently if you knew about the capabilities of =20
> all the potential responding UAs before sending an INVITE?
>=20
> Our traditional logic has been "put everything YOU support in the =20
> offer, and let the combination of RFC 3840/3841 working in the proxy =20
> pick a UA that best-matches what you have. Of course, this can =20
> sometimes lead to requests getting sent to the phone that is the =20
> closest feature match rather than the one that's closest to=20
> the called =20
> party.
>=20
> Another alternative is to make an INVITE that offers only the bare =20
> minimum, then explore upgrading the call once it is established to =20
> engage additional features.
>=20
>=20
> But perhaps we have a problem. I want to offer a "rich" INVITE, but =20
> the low-end client with no features keeps answering before the =20
> "better" box. Now, we need to know the "best" set of=20
> capabilities, so =20
> that we can construct an INVITE that only the "right" UA is going to =20
> answer.
>=20
>=20
> > and if there is a significant benefit, why you can't just=20
> obtain the =20
> > device contact URIs using the reg event package and then send =20
> > OPTIONS to each.
>=20
> That one's easy. I'm not letting you subscribe to my reg-event =20
> information.=20
[JRE] If my policy is not to let you subscribe to reg-event, I would antici=
pate having a similar policy not to reveal my registered contacts through 3=
xx to OPTIONS.

> I don't even HAVE reg-event information, as my=20
> registrar =20
> doesn't do it.
[JRE] Equally, I don't support 3xx to OPTIONS because my proxy doesn't do i=
t. I agree that 3xx to OPTIONS might be easier to implement than reg-event,=
 but you still need all the authentication/authorisation stuff.

John

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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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IMO it would be useful having in SIP a similar mechanism to the one used 
in XMPP
as an alternative way to do OPTIONS

/Sal

On 4/8/10 5:13 PM, Peter Saint-Andre wrote:
> FWIW, this is exactly what we do in XMPP.
>
> On 4/8/10 9:08 AM, Salvatore Loreto wrote:
>    
>> Hi Paul,
>>
>> the use case you have written down is exactly the one I have in mind and
>> was going to send to this ml.
>>
>> I want add one more benefit in knowing all the capabilities of the
>> different Bob devices in advance and
>> not just of one device.
>> An OPTIONS response (or another mechanism!) that contains a level of
>> granularity that let Alice (or rather Alices device)
>> know that Bob can take only audio on one device, but
>> audio/video/text/picture on his other device (as describe in Paul use
>> case),
>> would help Alice's device understand that starting a voice session with
>> only voice Bob device, there is no way a re-INVITE to add video would work;
>> so if she thinks that a voice call could be then transformed in a
>> voice+video, she knows that should contact directly the
>> audio/video/text/picture device.
>>
>> cheers
>> Sal
>>
>>
>> On 4/8/10 4:01 PM, Paul Kyzivat wrote:
>>      
>>> [as individual]
>>>
>>> It seems that Christer has come close to offering a use case, even if
>>> that was not his intent. And its still a little incomplete. I'll try to
>>> fill it out a bit:
>>>
>>> Alice wants to call Bob.
>>> Her phone app first queries Bob's AOR with OPTIONS.
>>> The response to that is a 3xx with Contact URIs for
>>> Bob's home phone and his mobile phone. So Alice's phone
>>> sends additional OPTIONS requests to each of those.
>>>    From the responses it learns that Bob's mobile phone support
>>> audio and video, while his home phone (behind a pstn gateway)
>>> only supports audio.
>>>
>>> Alice has video support on her phone, and since Bob may be able to
>>> support video too, decides to make an audio+video call. The
>>> phone sends an INVITE with both, and caller preferences indicating
>>> a preference for audio+video.
>>>
>>> The INVITE reaches the server handling Bob's AOR.
>>> It processes the callerprefs, and so decides to initially send the
>>> invite to Bob's mobile phone. But there is no answer there, so
>>> the server forks the invite toward Bob's home phone. Bob answers that
>>> one, accepting the audio and rejecting the video.
>>>
>>> The result of this is exactly the same as if no OPTIONS was ever sent,
>>> and the INVITE was sent indicating Alice's own preferences. That is the
>>> point that John already made.
>>>
>>> The only "benefit" I can see from use of options here would be if it was
>>> discovered that Bob had no video support. In that case, the invite could
>>> have been sent without offering video. Is that advantageous? Does it
>>> cost more to initiate a call that offers video? Does it take more
>>> network resources to do so, even when ultimately the video is rejected?
>>>
>>> The only real benefit I can see to this is if the calling user is given
>>> an enhanced user interface - akin to what one gets with buddy lists -
>>> where there is a display showing that Bob has two devices with differing
>>> capabilities, with Alice having the option to choose one or allowing the
>>> phone to make the choice.
>>>
>>> To provide that feature, presence would be a much better choice.
>>> OPTIONS is in a sense "poor man's presence". Christer has mentioned that
>>> he considers presence to be too "heavy weight" for this, but I remain to
>>> be convinced of that. Perhaps maintaining persistent subscriptions to
>>> presence status of everyone you might possibly call is too heavy weight,
>>> but doing a polling subscription prior to a call won't be any heavier
>>> than all this options stuff.
>>>
>>> It may be that lots of devices don't publish presence state. But that
>>> can be solved in many ways. A registrar can synthesize basic presence
>>> state from registration data. And it could even use OPTIONS to refine
>>> that.
>>>
>>> I suspect the real hidden issue here is about presence, and so I'd
>>> prefer to see it addressed from that direction. If that in turn leads to
>>> a desire to use OPTIONS to get some degree of presence state from
>>> devices that don't otherwise publish it, then lets confront it in that
>>> context. And if the issue is that presence is typically not authorized
>>> for as many requesters as OPTIONS is, then lets look at possible policy
>>> recommendations in that regard.
>>>
>>>      Thanks,
>>>      Paul
>>>
>>> Christer Holmberg wrote:
>>>
>>>        
>>>> Hi,
>>>>
>>>>
>>>>          
>>>>>>  From my perspective, the main use-case is not to discover services,
>>>>>>              
>>>>>
>>>>>            
>>>>>> but to discover capabilities in order to determine what services
>>>>>> can be applied.
>>>>>>
>>>>>> The idea is also that the OPTIONS request would be sent prior to the
>>>>>> INVITE, to try to make sure that the returned contacts are actually
>>>>>> available.
>>>>>>
>>>>>>              
>>>>> But you know that would never work in practice/the-real-world. Think
>>>>> about all the routing rules
>>>>> people have for INVITEs - in some cases *only* for INVITEs.
>>>>> And even if they route OPTIONS identically in the SIP domain,
>>>>> for any given AoR it'll go to any of: a SIP UA, or a PSTN
>>>>> gateway, or a H.323-gateway, or an MGCP call-agent, or an
>>>>> IP-PBX, or a foobar-protocol-gateway, or a vmail server, or
>>>>> an announcement server, or an app server, etc.  Just because
>>>>> the OPTIONS reaches the end of the SIP domain, means nothing
>>>>> about whether an INVITE to the higher-layer target (which in
>>>>> practice is a E.164 number, that crosses from SIP into many
>>>>> other protocol domains) is actually available to receive a call.
>>>>>
>>>>>            
>>>> The assumption is that the OPTIONS request reaches the registrar,
>>>> that returns the registered contacts. It is never forwarded to any UAS.
>>>>
>>>> Then, at some point an INVITE is sent, and it is addressed to the
>>>> same AOR (the INVITE request might be routed differently than the
>>>> OPTIONS request). Based on the UAS information, the INVITE may now
>>>> contain feature tags etc that indicates a wish to reach an UAS with
>>>> certain capabilities.
>>>>
>>>> Now, unless the INVITE is addressed to a specific UAS, it can of
>>>> course not be guranteed that the INVITE will actually reach such UAS.
>>>> But, that is a "feature" of RFC 3841, and it would be the case even
>>>> if you had never sent any OPTIONS. What OPTIONS allows provides you
>>>> is capability information about the UAS(s) associated with the AOR,
>>>> which might be useful when sending the INVITE.
>>>>
>>>> No, it does not guarantee that the user is available - it only
>>>> provides you with that same information as OPTIONS in a non-forking
>>>> case does.
>>>>
>>>> Regards,
>>>>
>>>> Christer
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>>>>          
>    


-- 
Salvatore Loreto
www.sloreto.com


From brett@broadsoft.com  Fri Apr  9 02:28:24 2010
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From: Brett Tate <brett@broadsoft.com>
To: Robert Sparks <rjsparks@nostrum.com>, Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Fri, 9 Apr 2010 02:27:26 -0700
Thread-Topic: [sipcore] RFC 4320: deprecation of non-INVITE 408 responses
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Subject: Re: [sipcore] RFC 4320: deprecation of non-INVITE 408 responses
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I am indicating that the logic that used to dismiss my concern over depreca=
ting all non-INVITE 408 responses is incorrect.

Per RFC 3261, a 408 Request Timeout response to INVITE has no different mea=
ning from a 408 Request Timeout to non-INVITE.  A request timeout is a requ=
est timeout.=20

RFC 3261 section 21.4.9: "The server could not produce a response within a =
suitable amount of time, for example, if it could not determine the locatio=
n of the user in time.  The client MAY repeat the request without modificat=
ions at any later time."

RFC 4320 unfortunately deprecated all 408 responses to non-INVITE instead o=
f only the use case where sending the 408 isn't desirable.

Since RFC 4320 unfortunately deprecated 408 Request Timeout responses to no=
n-INVITE, what response does the working group want to be sent to indicate =
"Request Timeout" when the timeout occurs quickly (like after 1 second) ins=
tead of after long delay associated with rfc4320/rfc4321's use case?  Is 50=
4 Server Time-out the best alternative to indicate Request Timeout?

I would rather see RFC 4320's 408 deprecation be limited to appropriate use=
 case instead of defining or using another response code to indicate Reques=
t Timeout.


> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]
> Sent: Thursday, April 08, 2010 4:33 PM
> To: Brett Tate
> Cc: Adam Roach; SIPCORE
> Subject: Re: [sipcore] RFC 4320: deprecation of non-INVITE 408
> responses
>=20
> We're talking about the non-INVITE transaction.
> I think what you're looking at in 3398 is INVITE oriented (408's for
> INVITE have a broader meaning).
>=20
> RjS
>=20
> On Apr 8, 2010, at 2:46 PM, Brett Tate wrote:
>=20
> >> 408's semantics in 3261 are clearly bound to the transaction
> >> machinery. The other cases you are describing should use a
> >> different response code.
> >
> > If 408's semantics were so clear, I'd think RFC 3398, B2BUAs, and
> etcetera would not have generated or documented to generate a 408
> response based upon another interfaces timeouts or receipt of 408.
> >
> > Should errata be opened against RFC 3398?
> >
> > Thanks for the response,
> >
> > Brett
> >


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Cc: "WORLEY, DALE R \(DALE\)" <dworley@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Dean Willis wrote:
> 
> On Apr 8, 2010, at 7:06 PM, Paul Kyzivat wrote:
> 
>>
>>
>> Dean Willis wrote:
>>> But perhaps we have a problem. I want to offer a "rich" INVITE, but 
>>> the low-end client with no features keeps answering before the 
>>> "better" box. Now, we need to know the "best" set of capabilities, so 
>>> that we can construct an INVITE that only the "right" UA is going to 
>>> answer.
>>
>> I would recommend that you then set up callerprefs to *prefer* (but 
>> not require) the "rich" capabilities. With luck the call will be 
>> offered to the rich device first, and if that one doesn't answer then 
>> the dumber ones will still be tried.
>>
> 
> Do you feel lucky?

Never. I'm definitely a "glass half empty" person.

OTOH, if it doesn't work, its the callee's fault, not mine. If he wants 
his "rich" devices to be useful he ought to get a smarter proxy.

	Thanks,
	Paul

From dean.willis@softarmor.com  Fri Apr  9 08:47:46 2010
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On Apr 9, 2010, at 1:24 AM, Elwell, John wrote:
>>> That one's easy. I'm not letting you subscribe to my reg-event
>> information.
> [JRE] If my policy is not to let you subscribe to reg-event, I would  
> anticipate having a similar policy not to reveal my registered  
> contacts through 3xx to OPTIONS.
>
>> I don't even HAVE reg-event information, as my
>> registrar
>> doesn't do it.
> [JRE] Equally, I don't support 3xx to OPTIONS because my proxy  
> doesn't do it. I agree that 3xx to OPTIONS might be easier to  
> implement than reg-event, but you still need all the authentication/ 
> authorisation stuff.
>

Both good arguments. I'm inclined more than ever to suggest that  
proxies should just never fork OPTIONS.

Besides using it as a very inefficient PING, what do we really use  
OPTIONS for in the real world? Even when used as a "ping", what does  
it mean to fork it? Does contact-routing of OPTIONS (including  
forking) create a security hole?

If we suggest proxies don't fork OPTIONs, how should they respond to  
an OPTIONS request for an AOR for which they are responsible? With  
their own capabilities? Perhaps with a 302 listing the public GRUUs,  
if any?

--
Dean



From john.elwell@siemens-enterprise.com  Fri Apr  9 09:40:43 2010
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To: Dean Willis <dean.willis@softarmor.com>
Date: Fri, 9 Apr 2010 18:40:33 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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=20

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: 09 April 2010 16:47
> To: Elwell, John
> Cc: Hans Erik van Elburg; WORLEY, DALE R (DALE); sipcore@ietf.org
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
>=20
> On Apr 9, 2010, at 1:24 AM, Elwell, John wrote:
> >>> That one's easy. I'm not letting you subscribe to my reg-event
> >> information.
> > [JRE] If my policy is not to let you subscribe to=20
> reg-event, I would =20
> > anticipate having a similar policy not to reveal my registered =20
> > contacts through 3xx to OPTIONS.
> >
> >> I don't even HAVE reg-event information, as my
> >> registrar
> >> doesn't do it.
> > [JRE] Equally, I don't support 3xx to OPTIONS because my proxy =20
> > doesn't do it. I agree that 3xx to OPTIONS might be easier to =20
> > implement than reg-event, but you still need all the=20
> authentication/=20
> > authorisation stuff.
> >
>=20
> Both good arguments. I'm inclined more than ever to suggest that =20
> proxies should just never fork OPTIONS.
>=20
> Besides using it as a very inefficient PING, what do we really use =20
> OPTIONS for in the real world? Even when used as a "ping", what does =20
> it mean to fork it? Does contact-routing of OPTIONS (including =20
> forking) create a security hole?
>=20
> If we suggest proxies don't fork OPTIONs, how should they respond to =20
> an OPTIONS request for an AOR for which they are responsible? With =20
> their own capabilities? Perhaps with a 302 listing the public GRUUs, =20
> if any?
[JRE] I think this misses my point. If the proxy returns a 302 with all my =
contacts (GRUUs), I have revealed to the caller how many devices I have and=
 their individual addresses, i.e., basically the same information as reg-ev=
ent would give the caller. So similar authorisation policies should apply.

John


>=20
> --
> Dean
>=20
>=20
> =

From dean.willis@softarmor.com  Fri Apr  9 12:18:44 2010
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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On Apr 9, 2010, at 11:40 AM, Elwell, John wrote:

>>
>> If we suggest proxies don't fork OPTIONs, how should they respond to
>> an OPTIONS request for an AOR for which they are responsible? With
>> their own capabilities? Perhaps with a 302 listing the public GRUUs,
>> if any?
> [JRE] I think this misses my point. If the proxy returns a 302 with  
> all my contacts (GRUUs), I have revealed to the caller how many  
> devices I have and their individual addresses, i.e., basically the  
> same information as reg-event would give the caller. So similar  
> authorisation policies should apply.

Well, the public GRUUs do list the different devices, but not their IP  
Addresses, per se. One could also use temp-gruu for the same function.

I like the idea of returning some sort of "OPTIONS aggregate", but  
this leads to questions of disjoint feature sets across devices. For  
example what happens if one picks two "must" features out of the  
aggregate and constructs a request around them, when in reality there  
are two different devices, each of which has only one of the features?

--
Dean



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Hadriel,

Sorry for the belated reply.  I was on vacation this past week (and still
am), and wasn't able to respond to everything this week.

> I was worried about this too ('cause I also suggested option-tags and
> was worried about code-changes), but ISTM this is basically already the
> case: unless most folks do exactly what this doc says today, they're
> non-compliant and would require changing either code or config.

I don't know what percentage of devices would be compliant vs.
non-compliant.  However, unless we stick a stake in the ground somewhere, it
will continue to be rather messy.
 
> The thing is, afaict, many folks DON'T do what the doc says.  Not
> exactly, which is the same thing (right?).  For example, of three
> rather popular vendors I just looked at OPTION-ping traces of, none
> used the 486 for anything.  I don't know what they'd do if they got
> one.  All 3 of the vendors use 503 to signal "out-of-service" back, but
> at least 2 of them still allow (and want/expect to receive) in-dialog
> messages - I can't tell with the 3rd.  And one of them still _sent_ all
> messages even if it got a 503.

Note that in the language of the draft, I often use the word "should" for
things, including the 486.  We could mandate strict behavior in some cases
(and should), but "guidance" is all that is necessary in some instances.  If
a device wishes to keep responding until it drops deaf entirely, that's OK.
But, we ought to encourage the use of a different response code, in my
opinion.

In any case, the fact that OPTIONS is used for a "ping" so widely and there
are so many variations on the usage suggests we really need to reach some
agreement on something.  I'm certainly open to input, but we're not going to
be able to accommodate every implementation.

Paul



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Hadriel Kaplan'" <HKaplan@acmepacket.com>, <sipcore@ietf.org>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Christer,

 
> >>But more importantly because it conflicts with two other pieces of
> >>work being done in the IETF:
> >>1) The keep-alive portion, which is done by RFC 5626 and also being
> >>worked on in draft-ietf-sipcore-keep.
> >
> >That document covers a different problem that's not related to an
> application "ping" between two devices.  One could view it as such for
> active sessions, I suppose, but it really has a different focus.
> 
> That is not true.
> 
> If you negotiate the keep-alive as part of the registration procedure,
> the "pings" will be sent during the duration of the registration - no
> matter whether you have active sessions or not.

True, but this draft is not focused on places where a REGISTER message is
sent between a UA and a Registrar.  Rather, the draft is focused on places
where a REGISTER is not likely used, such as between two peer SBCs.

> Now, that of course only applies to nodes that are involved in
> registration, and the keep draft currently says that keep alives
> negotiated during session setup are only valid for that session.
> However, I think that at some point there was a comment that it should
> also be allowed to negotiate to longer keep-alive durations during the
> session setup, for proxy-to-proxy "pings" etc. I guess we'll get back
> to that discussion once we get the keep discussion up again (next week,
> according to my plans).

We also need a solution for when there are no active sessions and haven't
been for hours, for example.  OPTIONS "pings" have been serving this role
for a while.  OPTIONS "pings" can also be used by diagnostic equipment to
alert admins that a device is down. 

Paul


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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Brett Tate'" <brett@broadsoft.com>, <sipcore@ietf.org>
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Cc: 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Brett,

I guess I'm still a bit confused.  What in this draft violates the rule?  I
think we're in agreement here, but apparently the text of the draft isn't
quite as clear as it ought to be, I guess.

Paul

> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]
> Sent: Sunday, April 04, 2010 10:23 AM
> To: Paul E. Jones; sipcore@ietf.org
> Cc: 'Hadriel Kaplan'
> Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
> 
> > As for the meaning, I'm confused with your question.
> > The intent is to obey the Retry-After header.  If the
> > reply to the OPTIONS has a Retry-After header, then
> > don't send another OPTIONS until that time expires.
> 
> Your response is appropriate for responses like 486 and 500 with Retry-
> After; however a 503 with Retry-After applies to more than OPTIONS.
> Per RFC 3261 section 21.5.4, it means you SHOULD NOT send any requests
> (INVITE, BYE, ACK, CANCEL, REGISTER, ...) during that interval.
> 



From paulej@packetizer.com  Sat Apr 10 17:08:31 2010
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Subject: Re: [sipcore] draft-jones-sip-options-ping-00: comments
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Brett,

> The current wording requires honoring the 503 with Retry-After overload
> mechanism.  RFC 3261 section 21.5.4 is at the SHOULD level instead of a
> MUST.
> 
> If you only intended the Retry-After to apply to OPTIONS (instead of
> all requests), you can't use a 503 response (unless maybe you used an
> option-tag or something to indicate a meaning other than RFC 3261
> section 21.5.4).

Ah, I think this was your point.  I believe a 503 would apply to the entire
server, just as 3261 says.  So, a device might return a 503 or 486.  In both
cases, the Retry-After header should be honored and these procedures do not
change the intent of 3261 -- at least not intentionally.

> I'm referring to devices with multiple ip-addresses such as 1 or more
> IPv4 addresses and/or 1 or more IPv6 addresses.  Thus the source ip-
> address of 503 response may be different from where UAC sent the
> OPTIONS; and the sender of 503 may have additional ip-addresses used
> for SIP communication between these two servers.  Since rfc3261 was
> somewhat vague, I'm attempting to ensure the draft is clear since it is
> requiring or recommending the use of 503 with Retry-After.

This is a problem general for SIP, since there is no way for a SIP entity to
know that a DNS query that returns 4 replies relates to the same or
different servers.  I can't fix that problem in this draft :-)

Paul



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Hadriel Kaplan'" <HKaplan@acmepacket.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Cc: "'Vivek Bhargava \(vbharga\)'" <vbharga@cisco.com>, sipcore@ietf.org, 'Gonzalo Salgueiro' <gsalguei@cisco.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hadriel,

Yeah, I think that's OK, too.  And, it's one reason for recommending setting
the Max-Forwards header.  These procedures really are intended for use
between two devices that have some kind of established relationship.  I
would not expect them to be used between any two devices on the Internet,
nor would I expect all devices to support an OPTIONS "ping".  We might see
service providers require this support at peer interconnection points or
perhaps between an enterprise or service provider.  We might also see this
kind of support required inside an enterprise or carrier network for
monitoring purposes.  But, such requirements would be explicit,
manufacturers would build accordingly, and things should work.  What we
need, I think, are the basic guidelines to help ensure (what I've
unfortunately seen) where all kinds of different response codes more-or-less
equate to a 503.

Paul

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> Sent: Monday, April 05, 2010 5:32 PM
> To: Paul Kyzivat; Paul E. Jones
> Cc: sipcore@ietf.org; 'Vivek Bhargava (vbharga)'; 'Gonzalo Salgueiro'
> Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
> 
> 
> One problem I've seen is when something like a UA or IP-PBX is OPTIONS
> pinging "sip:sip.ssp.com" - that will resolve in DNS to an ingress
> proxy, which actually is the real target of the ping request - but the
> ingress proxy itself may think the OPTIONS needs to get routed to the
> authoritative server for sip.ssp.com, which may well be something
> _after_ the proxy.
> 
> Setting max-forwards to 1 helps avoid that, but in some proxies that
> will get back a 483 too many hops, which is actually ok in my book -
> after all you know its alive and processing requests - but in some
> devices that increments error counters which people don't like.
> 
> -hadriel
> 
> 
> > -----Original Message-----
> > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > Sent: Monday, April 05, 2010 8:52 AM
> > To: Paul E. Jones
> > Cc: Hadriel Kaplan; sipcore@ietf.org; 'Vivek Bhargava (vbharga)';
> 'Gonzalo
> > Salgueiro'
> > Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
> >
> > [as an individual]
> >
> > Paul E. Jones wrote:
> >
> > >> Other, specific comments on the draft:
> > >> 0) You might want to consider specifying what the To/From-URIs
> should
> > >> be sent as, and that a receiver should be lenient.  We've seen
> interop
> > >> problems with those URIs for OPTIONS pings.
> > >
> > > I wanted to be rather strict in saying that only sip:192.168.4.56
> was
> > > acceptable, for example, but others I consulted with thought that
> was
> > too
> > > restrictive.  So, the language is soft here.
> >
> > I believe I was one of the "others".
> >
> > My personal concern was that making such a mandate starts to
> interfere
> > with the layering of the sip stack. If you "know" your neighbors via
> DNS
> > names, then the translation to ip addresses occurs in the code that
> > implements 3263, which is way below where you construct, send, and
> > receive responses to messages.
> >
> > I didn't want to see a set of recommendations that require violation
> of
> > such layering.
> >
> > But I'd be interested in hearing other points of view on this.
> >
> > 	Thanks,
> > 	Paul
> 



From HKaplan@acmepacket.com  Sat Apr 10 19:52:31 2010
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To: "Paul E. Jones" <paulej@packetizer.com>, 'Christer Holmberg' <christer.holmberg@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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Cc: "'Vivek Bhargava \(vbharga\)'" <vbharga@cisco.com>, 'Gonzalo Salgueiro' <gsalguei@cisco.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Saturday, April 10, 2010 8:00 PM
>=20
> We also need a solution for when there are no active sessions and haven't
> been for hours, for example. =20

It was my impression draft-sip-core-keep was also supposed to support that.=
  I agree that its current wording is wrong, imo, but when it started out m=
y impression was it could be used for permanent proxy-to-proxy (or b2bua-to=
-b2bua, etc.) connections/associations, regardless of there being an active=
 dialog between the two.  Without that I agree with you that it's not as us=
eful.


> OPTIONS "pings" have been serving this role
> for a while.  OPTIONS "pings" can also be used by diagnostic equipment to
> alert admins that a device is down.

The draft-keep does that (detect device failure), but what it doesn't do is=
 tell you when it's getting too busy.  I liked it over OPTIONS-pings becaus=
e draft-keep's a lot lighter-weight/more-efficient, trivial to implement, e=
xplicit in its purpose, and unavoidably single-hop.  In other words it woul=
d *replace* OPTIONS pings.  But if we go fix OPTIONS-pings, would we need d=
raft-keep anymore?  Would people implement it?

-hadriel

From HKaplan@acmepacket.com  Sat Apr 10 20:13:37 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Paul E. Jones" <paulej@packetizer.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Sat, 10 Apr 2010 23:13:30 -0400
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi Paul,
Inline...

> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Saturday, April 10, 2010 8:16 PM
>=20
> Yeah, I think that's OK, too.  And, it's one reason for recommending
> setting
> the Max-Forwards header.  These procedures really are intended for use
> between two devices that have some kind of established relationship.  I
> would not expect them to be used between any two devices on the Internet,
> nor would I expect all devices to support an OPTIONS "ping". =20

I'm not sure what comment I made elicited the second sentence.  You mean my=
 concern for error logs due to max-forwards=3D1?  I.e., you're saying that =
since these devices "know" they're doing it, they shouldn't log errors?  Su=
re.  But it's kinda hard to know the OPTIONS is for ping and not just some =
other OPTIONS that happens to have looped around enough to get down to a Ma=
x-Forwards of 1.=20

That's one of the reasons it would be nice if the OPTIONS actually had some=
 way to say "I'm a ping".  For example, my company's systems set the To/Fro=
m username to "ping" (e.g., "sip:ping@192.1.2.3") by default.


> We might see
> service providers require this support at peer interconnection points or
> perhaps between an enterprise or service provider. =20

That's already the case.

-hadriel

> > -----Original Message-----
> > From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> > Sent: Monday, April 05, 2010 5:32 PM
> > To: Paul Kyzivat; Paul E. Jones
> > Cc: sipcore@ietf.org; 'Vivek Bhargava (vbharga)'; 'Gonzalo Salgueiro'
> > Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> > 00.txt
> >
> >
> > One problem I've seen is when something like a UA or IP-PBX is OPTIONS
> > pinging "sip:sip.ssp.com" - that will resolve in DNS to an ingress
> > proxy, which actually is the real target of the ping request - but the
> > ingress proxy itself may think the OPTIONS needs to get routed to the
> > authoritative server for sip.ssp.com, which may well be something
> > _after_ the proxy.
> >
> > Setting max-forwards to 1 helps avoid that, but in some proxies that
> > will get back a 483 too many hops, which is actually ok in my book -
> > after all you know its alive and processing requests - but in some
> > devices that increments error counters which people don't like.
> >
> > -hadriel

From HKaplan@acmepacket.com  Sat Apr 10 20:16:26 2010
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To: "Paul E. Jones" <paulej@packetizer.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Sat, 10 Apr 2010 23:16:16 -0400
Thread-Topic: Options-ping vs. Draft-keep (was draft-jones-sip-options-ping-00.txt)
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Subject: [sipcore] Options-ping vs. Draft-keep (was draft-jones-sip-options-ping-00.txt)
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Hi Paul,

I suppose there's no real point in arguing over the nitty-gritty details ri=
ght now.  Let's assume whatever this turns out to be isn't 100% backwards-c=
ompatible with your or some people's implementations of OPTIONS-pings, and =
thus requires code changes for some or even many folks.  The big question i=
n my mind is: should we do this, or should we do draft-keep?

-hadriel


> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Saturday, April 10, 2010 8:16 PM
>
> These procedures really are intended for use
> between two devices that have some kind of established relationship.  I
> would not expect them to be used between any two devices on the Internet,
> nor would I expect all devices to support an OPTIONS "ping".  We might se=
e
> service providers require this support at peer interconnection points or
> perhaps between an enterprise or service provider.  We might also see thi=
s
> kind of support required inside an enterprise or carrier network for
> monitoring purposes.  But, such requirements would be explicit,
> manufacturers would build accordingly, and things should work.  What we
> need, I think, are the basic guidelines to help ensure (what I've
> unfortunately seen) where all kinds of different response codes more-or-
> less
> equate to a 503.
>=20
> Paul
>=20

From christer.holmberg@ericsson.com  Sun Apr 11 05:05:40 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Paul E. Jones" <paulej@packetizer.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Sun, 11 Apr 2010 14:05:30 +0200
Thread-Topic: [sipcore] Options-ping vs. Draft-keep (was draft-jones-sip-options-ping-00.txt)
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Subject: Re: [sipcore] Options-ping vs. Draft-keep (was draft-jones-sip-options-ping-00.txt)
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Hi,

Keep in mind that the keep-draft does not define a keep-alive/ping mechanis=
m - it defines a mechanism to negotiate the mechanism already defined in Ou=
tbound.

Now, the keep-draft also supports negotiation of "keep-alives" between prox=
ies, where the "keep-alive" is more likely to be used as a ping, so I guess=
 your question is whether you want to replace THAT part with options-ping?=
=20

Or, are you also saying that we should do NAT keep-alives using options-pin=
g??? I really hope not... :)

Regards,

Christer
________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Hadr=
iel Kaplan [HKaplan@acmepacket.com]
Sent: Sunday, April 11, 2010 6:16 AM
To: Paul E. Jones; 'Paul Kyzivat'
Cc: 'Vivek Bhargava (vbharga)'; sipcore@ietf.org; 'Gonzalo Salgueiro'
Subject: [sipcore] Options-ping vs. Draft-keep (was draft-jones-sip-options=
-ping-00.txt)

Hi Paul,

I suppose there's no real point in arguing over the nitty-gritty details ri=
ght now.  Let's assume whatever this turns out to be isn't 100% backwards-c=
ompatible with your or some people's implementations of OPTIONS-pings, and =
thus requires code changes for some or even many folks.  The big question i=
n my mind is: should we do this, or should we do draft-keep?

-hadriel


> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Saturday, April 10, 2010 8:16 PM
>
> These procedures really are intended for use
> between two devices that have some kind of established relationship.  I
> would not expect them to be used between any two devices on the Internet,
> nor would I expect all devices to support an OPTIONS "ping".  We might se=
e
> service providers require this support at peer interconnection points or
> perhaps between an enterprise or service provider.  We might also see thi=
s
> kind of support required inside an enterprise or carrier network for
> monitoring purposes.  But, such requirements would be explicit,
> manufacturers would build accordingly, and things should work.  What we
> need, I think, are the basic guidelines to help ensure (what I've
> unfortunately seen) where all kinds of different response codes more-or-
> less
> equate to a 503.
>
> Paul
>
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From: Brett Tate <brett@broadsoft.com>
To: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sun, 11 Apr 2010 06:03:26 -0700
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> What in this draft violates the rule?

The draft is currently upgrading rfc3261's honoring Retry-After requirement=
s from MAY (MAY NOT) and SHOULD (SHOULD NOT) levels to MUST (MUST NOT) leve=
ls.  Thus it is preventing the UAC from making appropriate/configurable dec=
isions concerning if/how it is willing to honor/apply the reception of 503 =
with Retry-After.

For instance, the draft doesn't allow UAC to reduce the honoring of 503 wit=
h Retry-After value of 1 year to only apply for 30 seconds.

For instance, the draft doesn't allow UAC to avoid applying the 503 with Re=
try-After value to sending requests associated with emergency calls.

For instance, the draft doesn't allow UAC to avoid applying the 503 with Re=
try-After to sending requests within dialog.

For instance, the draft doesn't allow UAC to avoid honoring a 503 with Retr=
y-After when response source ip-address is not equivalent to where the requ=
est was sent.  Concerning their not matching, the draft currently does not =
mention it from a security risk or otherwise.

For instance, the draft doesn't allow UAC to only apply the 503 with Retry-=
After to specific interfaces.  More specifically if a UAC knows it should n=
ot trust/honor the 503 Retry-After value (because of a proxy rfc3261/draft =
non compliance or an insecure interface), the draft doesn't allow the devic=
e to instead not honor the 503 with Retry-After.


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To: Christer Holmberg <christer.holmberg@ericsson.com>, "Paul E. Jones" <paulej@packetizer.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Sun, 11 Apr 2010 11:02:41 -0400
Thread-Topic: [sipcore] Options-ping vs. Draft-keep (was draft-jones-sip-options-ping-00.txt)
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Subject: Re: [sipcore] Options-ping vs. Draft-keep (was draft-jones-sip-options-ping-00.txt)
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Sunday, April 11, 2010 8:06 AM
> To: Hadriel Kaplan; Paul E. Jones; 'Paul Kyzivat'
>=20
> Keep in mind that the keep-draft does not define a keep-alive/ping
> mechanism - it defines a mechanism to negotiate the mechanism already
> defined in Outbound.
>=20
> Now, the keep-draft also supports negotiation of "keep-alives" between
> proxies, where the "keep-alive" is more likely to be used as a ping, so I
> guess your question is whether you want to replace THAT part with options=
-
> ping?

Right, I'm questioning if we need to standardize a way to negotiate "keep-a=
lives" between proxies, if we decide to go and fix OPTIONS-pings. =20

Today OPTIONS-pings are already used as keep-alives between proxies, but th=
ey have interop and operational issues.  I had assumed that draft-keep woul=
d help us avoid those issues, and would be the "standard" way to do it.

It doesn't make much sense to run *both* between proxies, for example. I do=
n't think having two ways to solve the same problems is a good idea.

I don't much care which one it is - I think the stun/crlf method is better =
from a technical perspective, but OPTIONS would probably get wider adoption=
 faster.

=20
> Or, are you also saying that we should do NAT keep-alives using options-
> ping??? I really hope not... :)

Definitely not. :)

-hadriel

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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> > Yeah, I think that's OK, too.  And, it's one reason for recommending
> > setting
> > the Max-Forwards header.  These procedures really are intended for
> use
> > between two devices that have some kind of established relationship.
> I
> > would not expect them to be used between any two devices on the
> Internet,
> > nor would I expect all devices to support an OPTIONS "ping".
> 
> I'm not sure what comment I made elicited the second sentence.  You
> mean my concern for error logs due to max-forwards=1?  I.e., you're
> saying that since these devices "know" they're doing it, they shouldn't
> log errors?  

Oh, it was nothing you said.  I'm just adding more commentary :)

> Sure.  But it's kinda hard to know the OPTIONS is for ping
> and not just some other OPTIONS that happens to have looped around
> enough to get down to a Max-Forwards of 1.

I would expect it to know that it's an options ping based on the address to
which it is directed.  For example, sip:192.168.2.34 would suggest it's
destined for the particular device.  Perhaps one might also use
sip:ping@192.168.2.34 for this purpose.  I think each box that supports an
options ping ought to have a defined ping "address", with the IP address of
the machine serving as a reasonable default.  (That might not make sense for
IP phones, but I think it makes sense for B2BUAs.)
 
> That's one of the reasons it would be nice if the OPTIONS actually had
> some way to say "I'm a ping".  For example, my company's systems set
> the To/From username to "ping" (e.g., "sip:ping@192.1.2.3") by default.

And I'd say that's reasonable to do.
 
> > We might see
> > service providers require this support at peer interconnection points
> or
> > perhaps between an enterprise or service provider.
> 
> That's already the case.

Indeed :)  And, that's why I thought it would be reasonable to put this
draft out there.

Paul



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To: "'Brett Tate'" <brett@broadsoft.com>, <sipcore@ietf.org>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Brett,

So, I think the only change we need in the draft is to change the word
"MUST" to "SHOULD" in the paragraph just before Section 8?

Paul

> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]
> Sent: Sunday, April 11, 2010 9:03 AM
> To: Paul E. Jones; sipcore@ietf.org
> Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
> 
> > What in this draft violates the rule?
> 
> The draft is currently upgrading rfc3261's honoring Retry-After
> requirements from MAY (MAY NOT) and SHOULD (SHOULD NOT) levels to MUST
> (MUST NOT) levels.  Thus it is preventing the UAC from making
> appropriate/configurable decisions concerning if/how it is willing to
> honor/apply the reception of 503 with Retry-After.
> 
> For instance, the draft doesn't allow UAC to reduce the honoring of 503
> with Retry-After value of 1 year to only apply for 30 seconds.
> 
> For instance, the draft doesn't allow UAC to avoid applying the 503
> with Retry-After value to sending requests associated with emergency
> calls.
> 
> For instance, the draft doesn't allow UAC to avoid applying the 503
> with Retry-After to sending requests within dialog.
> 
> For instance, the draft doesn't allow UAC to avoid honoring a 503 with
> Retry-After when response source ip-address is not equivalent to where
> the request was sent.  Concerning their not matching, the draft
> currently does not mention it from a security risk or otherwise.
> 
> For instance, the draft doesn't allow UAC to only apply the 503 with
> Retry-After to specific interfaces.  More specifically if a UAC knows
> it should not trust/honor the 503 Retry-After value (because of a proxy
> rfc3261/draft non compliance or an insecure interface), the draft
> doesn't allow the device to instead not honor the 503 with Retry-After.
> 



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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Sun, 11 Apr 2010 18:21:45 -0400
Thread-Topic: [sipcore] Options-ping vs. Draft-keep (was draft-jones-sip-options-ping-00.txt)
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Subject: Re: [sipcore] Options-ping vs. Draft-keep (was draft-jones-sip-options-ping-00.txt)
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Oh wait, I totally misunderstood what you were saying.  You're saying "draf=
t-keep is just a negotiation mechanism, so it could negotiate to use option=
s-pings".

Right, so what's the value in that?  We needed an indication of support for=
 stun/crlf, because there was no way to indicate it between two proxies - e=
xcept of course them sending an OPTIONS to each other. :)

I guess in my mind the main use-case for draft-keep was for inter-proxy (or=
 inter-b2bua/etc.) on a permanent basis, and I was trying to avoid yet one =
more set of things to be configured/provisioned/get-wrong on systems. =20

So I was thinking "Great, let's do this simple indication on the wire, and =
if the next-hop does it I'll use the stun/crlf which is better anyway, othe=
rwise I'll use OPTIONS pings and hope for the best".

As an example, my system has half a dozen configurable settings for OPTIONS=
-pings, on a per next-hop basis, just to get it to work right with differen=
t vendors. :( =20

I'd like the next setting to be the Staples "Easy" button. :)

-hadriel


> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Hadriel Kaplan
> Sent: Sunday, April 11, 2010 11:03 AM
>=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: Sunday, April 11, 2010 8:06 AM
> > To: Hadriel Kaplan; Paul E. Jones; 'Paul Kyzivat'
> >
> > Keep in mind that the keep-draft does not define a keep-alive/ping
> > mechanism - it defines a mechanism to negotiate the mechanism already
> > defined in Outbound.
> >
> > Now, the keep-draft also supports negotiation of "keep-alives" between
> > proxies, where the "keep-alive" is more likely to be used as a ping, so
> I
> > guess your question is whether you want to replace THAT part with
> options-
> > ping?
>=20
> Right, I'm questioning if we need to standardize a way to negotiate "keep=
-
> alives" between proxies, if we decide to go and fix OPTIONS-pings.
>=20
> Today OPTIONS-pings are already used as keep-alives between proxies, but
> they have interop and operational issues.  I had assumed that draft-keep
> would help us avoid those issues, and would be the "standard" way to do i=
t.
>=20
> It doesn't make much sense to run *both* between proxies, for example. I
> don't think having two ways to solve the same problems is a good idea.
>=20
> I don't much care which one it is - I think the stun/crlf method is bette=
r
> from a technical perspective, but OPTIONS would probably get wider
> adoption faster.
>=20
>=20
> > Or, are you also saying that we should do NAT keep-alives using options=
-
> > ping??? I really hope not... :)
>=20
> Definitely not. :)
>=20
> -hadriel
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From christer.holmberg@ericsson.com  Mon Apr 12 00:24:58 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Mon, 12 Apr 2010 09:24:32 +0200
Thread-Topic: [sipcore] Options-ping vs. Draft-keep (was draft-jones-sip-options-ping-00.txt)
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Cc: "'Vivek Bhargava \(vbharga\)'" <vbharga@cisco.com>, "sipcore@ietf.org" <sipcore@ietf.org>, 'Gonzalo Salgueiro' <gsalguei@cisco.com>
Subject: Re: [sipcore] Options-ping vs. Draft-keep (was draft-jones-sip-options-ping-00.txt)
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Hi,=20

>Oh wait, I totally misunderstood what you were saying. =20
>You're saying "draft-keep is just a negotiation mechanism, so=20
>it could negotiate to use options-pings".

That is not what I am saying :)

I am saying that the draft is used to negotiate the mechanism defined in Ou=
tbound.

>Right, so what's the value in that?  We needed an indication=20
>of support for stun/crlf, because there was no way to=20
>indicate it between two proxies - except of course them=20
>sending an OPTIONS to each other. :)

That is what the keep-draft provides.

>I guess in my mind the main use-case for draft-keep was for=20
>inter-proxy (or inter-b2bua/etc.) on a permanent basis, and I=20
>was trying to avoid yet one more set of things to be=20
>configured/provisioned/get-wrong on systems. =20

Permanent proxy-to-proxy was not the main use-case of the keep-draft (at le=
ast not initially). The main use-cases were UA-to-proxy, to:

1. Allow negotiation of keep-alives during registration without having to s=
upport Outbound; and
2. Allow negotiation of keep-alives during session establishment

The inter-proxy was added later, since there were requests to do so.

The question is now whether we should REMOVE the inter-proxy case from the =
keep-draft all together, and ONLY use options-ping instead.

>So I was thinking "Great, let's do this simple indication on=20
>the wire, and if the next-hop does it I'll use the stun/crlf=20
>which is better anyway, otherwise I'll use OPTIONS pings and=20
>hope for the best".

Yes.

Regards,

Christer





=20
>As an example, my system has half a dozen configurable=20
>settings for OPTIONS-pings, on a per next-hop basis, just to=20
>get it to work right with different vendors. :( =20
>=20
>I'd like the next setting to be the Staples "Easy" button. :)



>=20
> -hadriel
>=20
>=20
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On=20
> > Behalf Of Hadriel Kaplan
> > Sent: Sunday, April 11, 2010 11:03 AM
> >=20
> > > -----Original Message-----
> > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > Sent: Sunday, April 11, 2010 8:06 AM
> > > To: Hadriel Kaplan; Paul E. Jones; 'Paul Kyzivat'
> > >
> > > Keep in mind that the keep-draft does not define a=20
> keep-alive/ping=20
> > > mechanism - it defines a mechanism to negotiate the mechanism=20
> > > already defined in Outbound.
> > >
> > > Now, the keep-draft also supports negotiation of "keep-alives"=20
> > > between proxies, where the "keep-alive" is more likely to=20
> be used as=20
> > > a ping, so
> > I
> > > guess your question is whether you want to replace THAT part with
> > options-
> > > ping?
> >=20
> > Right, I'm questioning if we need to standardize a way to negotiate=20
> > "keep- alives" between proxies, if we decide to go and fix=20
> OPTIONS-pings.
> >=20
> > Today OPTIONS-pings are already used as keep-alives between=20
> proxies,=20
> > but they have interop and operational issues.  I had assumed that=20
> > draft-keep would help us avoid those issues, and would be=20
> the "standard" way to do it.
> >=20
> > It doesn't make much sense to run *both* between proxies,=20
> for example.=20
> > I don't think having two ways to solve the same problems is=20
> a good idea.
> >=20
> > I don't much care which one it is - I think the stun/crlf method is=20
> > better from a technical perspective, but OPTIONS would probably get=20
> > wider adoption faster.
> >=20
> >=20
> > > Or, are you also saying that we should do NAT keep-alives using=20
> > > options- ping??? I really hope not... :)
> >=20
> > Definitely not. :)
> >=20
> > -hadriel
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> =

From john.elwell@siemens-enterprise.com  Mon Apr 12 00:45:24 2010
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To: Dean Willis <dean.willis@softarmor.com>
Date: Mon, 12 Apr 2010 09:44:54 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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=20

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: 09 April 2010 20:18
> To: Elwell, John
> Cc: Hans Erik van Elburg; WORLEY, DALE R (DALE); sipcore@ietf.org
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
>=20
> On Apr 9, 2010, at 11:40 AM, Elwell, John wrote:
>=20
> >>
> >> If we suggest proxies don't fork OPTIONs, how should they=20
> respond to
> >> an OPTIONS request for an AOR for which they are responsible? With
> >> their own capabilities? Perhaps with a 302 listing the=20
> public GRUUs,
> >> if any?
> > [JRE] I think this misses my point. If the proxy returns a=20
> 302 with =20
> > all my contacts (GRUUs), I have revealed to the caller how many =20
> > devices I have and their individual addresses, i.e., basically the =20
> > same information as reg-event would give the caller. So similar =20
> > authorisation policies should apply.
>=20
> Well, the public GRUUs do list the different devices, but not=20
> their IP =20
> Addresses, per se. One could also use temp-gruu for the same function.
[JRE] In fact RFC 5628 considers temp-gruu to be a bigger security risk, an=
d states in section 5:
"This element SHOULD NOT be
   included if the subscriber is not authorized to register to the AOR,
   unless there is an explicitly configured policy directing that it be
   included."
So if 3xx to OPTIONS reveals temp-gruus, this should be subject to the same=
 authorisation policy as reg-event.

John

From christer.holmberg@ericsson.com  Mon Apr 12 06:23:08 2010
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To: Dean Willis <dean.willis@softarmor.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>
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Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

As has been said, there can always be policies regarding what is returned -=
 no matter what mechanism is used.

So, in the case of OPTIONS, if there is a policy to not return the contacts=
, the proxy might return some kind of "OPTIONS aggregate".=20

We don't need to define those policies, we just say how things are done if =
applied.

Regards,

Christer


=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Dean Willis
> Sent: 9. huhtikuuta 2010 22:18
> To: Elwell, John
> Cc: WORLEY, DALE R (DALE); sipcore@ietf.org
> Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
>=20
>=20
> On Apr 9, 2010, at 11:40 AM, Elwell, John wrote:
>=20
> >>
> >> If we suggest proxies don't fork OPTIONs, how should they=20
> respond to=20
> >> an OPTIONS request for an AOR for which they are responsible? With=20
> >> their own capabilities? Perhaps with a 302 listing the=20
> public GRUUs,=20
> >> if any?
> > [JRE] I think this misses my point. If the proxy returns a 302 with=20
> > all my contacts (GRUUs), I have revealed to the caller how many=20
> > devices I have and their individual addresses, i.e., basically the=20
> > same information as reg-event would give the caller. So similar=20
> > authorisation policies should apply.
>=20
> Well, the public GRUUs do list the different devices, but not=20
> their IP Addresses, per se. One could also use temp-gruu for=20
> the same function.
>=20
> I like the idea of returning some sort of "OPTIONS=20
> aggregate", but this leads to questions of disjoint feature=20
> sets across devices. For example what happens if one picks=20
> two "must" features out of the aggregate and constructs a=20
> request around them, when in reality there are two different=20
> devices, each of which has only one of the features?
>=20
> --
> Dean
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

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Cc: sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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[as individual]

Paul E. Jones wrote:
> Brett,
> 
> So, I think the only change we need in the draft is to change the word
> "MUST" to "SHOULD" in the paragraph just before Section 8?

Let me just interject here that in the past we have frequently gotten 
into trouble by using SHOULD without qualification of what constitutes a 
valid reason for going against the SHOULD. Without actual text that 
clarifies that, SHOULD is often treated as if it said "you can claim 
conformance without implementing this."

	Thanks,
	Paul

> Paul
> 
>> -----Original Message-----
>> From: Brett Tate [mailto:brett@broadsoft.com]
>> Sent: Sunday, April 11, 2010 9:03 AM
>> To: Paul E. Jones; sipcore@ietf.org
>> Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
>> 00.txt
>>
>>> What in this draft violates the rule?
>> The draft is currently upgrading rfc3261's honoring Retry-After
>> requirements from MAY (MAY NOT) and SHOULD (SHOULD NOT) levels to MUST
>> (MUST NOT) levels.  Thus it is preventing the UAC from making
>> appropriate/configurable decisions concerning if/how it is willing to
>> honor/apply the reception of 503 with Retry-After.
>>
>> For instance, the draft doesn't allow UAC to reduce the honoring of 503
>> with Retry-After value of 1 year to only apply for 30 seconds.
>>
>> For instance, the draft doesn't allow UAC to avoid applying the 503
>> with Retry-After value to sending requests associated with emergency
>> calls.
>>
>> For instance, the draft doesn't allow UAC to avoid applying the 503
>> with Retry-After to sending requests within dialog.
>>
>> For instance, the draft doesn't allow UAC to avoid honoring a 503 with
>> Retry-After when response source ip-address is not equivalent to where
>> the request was sent.  Concerning their not matching, the draft
>> currently does not mention it from a security risk or otherwise.
>>
>> For instance, the draft doesn't allow UAC to only apply the 503 with
>> Retry-After to specific interfaces.  More specifically if a UAC knows
>> it should not trust/honor the 503 Retry-After value (because of a proxy
>> rfc3261/draft non compliance or an insecure interface), the draft
>> doesn't allow the device to instead not honor the 503 with Retry-After.
>>
> 
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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Cc: sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul,

> > So, I think the only change we need in the draft is to change the
> word
> > "MUST" to "SHOULD" in the paragraph just before Section 8?
> 
> Let me just interject here that in the past we have frequently gotten
> into trouble by using SHOULD without qualification of what constitutes
> a
> valid reason for going against the SHOULD. Without actual text that
> clarifies that, SHOULD is often treated as if it said "you can claim
> conformance without implementing this."

I think that's OK here, since what Brett was suggesting was that it is
already a SHOULD in effect in 3261.  In any case, the worst thing that
happens here is that a device ignores the Retry-After header and claims
compliance.  Apparently, some feel there are plenty of good reasons to do
so.

Paul



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Cc: "'Vivek Bhargava \(vbharga\)'" <vbharga@cisco.com>, sipcore@ietf.org, 'Gonzalo Salgueiro' <gsalguei@cisco.com>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul,

Paul E. Jones wrote:
> Hadriel,
> 
> Yeah, I think that's OK, too.  And, it's one reason for recommending setting
> the Max-Forwards header.  These procedures really are intended for use
> between two devices that have some kind of established relationship.  I
> would not expect them to be used between any two devices on the Internet,
> nor would I expect all devices to support an OPTIONS "ping".  We might see
> service providers require this support at peer interconnection points or
> perhaps between an enterprise or service provider.  We might also see this
> kind of support required inside an enterprise or carrier network for
> monitoring purposes.  But, such requirements would be explicit,
> manufacturers would build accordingly, and things should work.  What we
> need, I think, are the basic guidelines to help ensure (what I've
> unfortunately seen) where all kinds of different response codes more-or-less
> equate to a 503.

I don't understand how you envision this working.
Would you expect that in those places where this mechanism is to be used 
that the policy to do so will be explicitly configured mutually on both 
of the peers, and that support for this draft would then be required of 
both?

(As opposed to a situation where a particular device just has a policy 
of using this mechanism with all peers that meet some generic criterion 
for their role, and that are discovered to support the mechanism.)

	Thanks,
	Paul

> Paul
> 
>> -----Original Message-----
>> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
>> Sent: Monday, April 05, 2010 5:32 PM
>> To: Paul Kyzivat; Paul E. Jones
>> Cc: sipcore@ietf.org; 'Vivek Bhargava (vbharga)'; 'Gonzalo Salgueiro'
>> Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
>> 00.txt
>>
>>
>> One problem I've seen is when something like a UA or IP-PBX is OPTIONS
>> pinging "sip:sip.ssp.com" - that will resolve in DNS to an ingress
>> proxy, which actually is the real target of the ping request - but the
>> ingress proxy itself may think the OPTIONS needs to get routed to the
>> authoritative server for sip.ssp.com, which may well be something
>> _after_ the proxy.
>>
>> Setting max-forwards to 1 helps avoid that, but in some proxies that
>> will get back a 483 too many hops, which is actually ok in my book -
>> after all you know its alive and processing requests - but in some
>> devices that increments error counters which people don't like.
>>
>> -hadriel
>>
>>
>>> -----Original Message-----
>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>> Sent: Monday, April 05, 2010 8:52 AM
>>> To: Paul E. Jones
>>> Cc: Hadriel Kaplan; sipcore@ietf.org; 'Vivek Bhargava (vbharga)';
>> 'Gonzalo
>>> Salgueiro'
>>> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
>> 00.txt
>>> [as an individual]
>>>
>>> Paul E. Jones wrote:
>>>
>>>>> Other, specific comments on the draft:
>>>>> 0) You might want to consider specifying what the To/From-URIs
>> should
>>>>> be sent as, and that a receiver should be lenient.  We've seen
>> interop
>>>>> problems with those URIs for OPTIONS pings.
>>>> I wanted to be rather strict in saying that only sip:192.168.4.56
>> was
>>>> acceptable, for example, but others I consulted with thought that
>> was
>>> too
>>>> restrictive.  So, the language is soft here.
>>> I believe I was one of the "others".
>>>
>>> My personal concern was that making such a mandate starts to
>> interfere
>>> with the layering of the sip stack. If you "know" your neighbors via
>> DNS
>>> names, then the translation to ip addresses occurs in the code that
>>> implements 3263, which is way below where you construct, send, and
>>> receive responses to messages.
>>>
>>> I didn't want to see a set of recommendations that require violation
>> of
>>> such layering.
>>>
>>> But I'd be interested in hearing other points of view on this.
>>>
>>> 	Thanks,
>>> 	Paul
> 
> 
> 

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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul,
 
> I don't understand how you envision this working.
> Would you expect that in those places where this mechanism is to be
> used
> that the policy to do so will be explicitly configured mutually on both
> of the peers, and that support for this draft would then be required of
> both?

That is my expectation.  It's not really different from today, except today
we have a variety of OPTIONS "ping" approaches and I'd like to narrow those
down to one, if possible.
 
Paul



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Cc: sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul E. Jones wrote:
> Paul,
> 
>>> So, I think the only change we need in the draft is to change the
>> word
>>> "MUST" to "SHOULD" in the paragraph just before Section 8?
>> Let me just interject here that in the past we have frequently gotten
>> into trouble by using SHOULD without qualification of what constitutes
>> a
>> valid reason for going against the SHOULD. Without actual text that
>> clarifies that, SHOULD is often treated as if it said "you can claim
>> conformance without implementing this."
> 
> I think that's OK here, since what Brett was suggesting was that it is
> already a SHOULD in effect in 3261.  In any case, the worst thing that
> happens here is that a device ignores the Retry-After header and claims
> compliance.  Apparently, some feel there are plenty of good reasons to do
> so.

I don't think that lets you off the hook.
We've learned from experience that we were too vague in the past.
So I would expect you to be less vague here - being more explicit about 
what conditions justify violating the SHOULD.

	Thanks,
	Paul

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Mon, 12 Apr 2010 12:22:04 -0400
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Paul Kyzivat
>=20
> I don't think that lets you off the hook.
> We've learned from experience that we were too vague in the past.
> So I would expect you to be less vague here - being more explicit about
> what conditions justify violating the SHOULD.

And really using "MUST..., unless..." instead.

-hadriel

From brett@broadsoft.com  Mon Apr 12 10:14:31 2010
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From: Brett Tate <brett@broadsoft.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, Paul Kyzivat <pkyzivat@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Mon, 12 Apr 2010 10:10:31 -0700
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> > I don't think that lets you off the hook.
> > We've learned from experience that we were too vague in=20
> > the past.  So I would expect you to be less vague here -=20
> > being more explicit about what conditions justify violating=20
> > the SHOULD.
>=20
> And really using "MUST..., unless..." instead.

Is making 503 with Retry-After usable (i.e. so it does more good than harm)=
 something that should be done within this draft and in this working group?=
  Or is that something that should be addressed within the hopefully dispat=
ched overload working group?


From victor.pascual.avila@gmail.com  Mon Apr 12 10:48:17 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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On Mon, Apr 12, 2010 at 7:10 PM, Brett Tate <brett@broadsoft.com> wrote:
>> > I don't think that lets you off the hook.
>> > We've learned from experience that we were too vague in
>> > the past. =C2=A0So I would expect you to be less vague here -
>> > being more explicit about what conditions justify violating
>> > the SHOULD.
>>
>> And really using "MUST..., unless..." instead.
>
> Is making 503 with Retry-After usable (i.e. so it does more good than har=
m) something that should be done within this draft and in this working grou=
p? =C2=A0Or is that something that should be addressed within the hopefully=
 dispatched overload working group?

Since, to some extent, the 503 + Retry-after mechanism represents an
on/off overload control approach (w/ some known problems [1]), IMHO it
deserves special attention within the so-expected overload control WG.

[1] http://tools.ietf.org/html/rfc5390#section-5

Cheers,
--=20
Victor Pascual =C3=81vila

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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Brett Tate wrote:
>>> I don't think that lets you off the hook.
>>> We've learned from experience that we were too vague in 
>>> the past.  So I would expect you to be less vague here - 
>>> being more explicit about what conditions justify violating 
>>> the SHOULD.
>> And really using "MUST..., unless..." instead.
> 
> Is making 503 with Retry-After usable (i.e. so it does more good than harm) something that should be done within this draft and in this working group?  Or is that something that should be addressed within the hopefully dispatched overload working group?

Well, now you ask the hard questions.

With my chair hat on, IMO any changes to the behavior of 503 will at 
least need to be blessed by sipcore. (But I'm new as a sipcore chair, so 
I'm willing to learn from the other chair or the AD that is isn't so.)

After that, I think it depends on what is to happen with this draft.

If it is intended to be informational, then it can't make any normative 
changes. It can make any recommendations it wants as long as they don't 
require violating normative behavior.

If its aspirations are higher, and maybe even if they aren't, then I 
think it makes sense to coordinate it with the work of the overload wg.
(AFAIK we don't yet *have* an overload WG, but there is a mailing list, 
and I gather its expected that we will have the WG before long.)

With my individual hat on I will recommend that it go there for some 
discussion of if/how it relates.

	Thanks,
	Paul

From christer.holmberg@ericsson.com  Mon Apr 12 22:49:58 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Tue, 13 Apr 2010 07:49:48 +0200
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

I agree with Paul.

I think that we should try to have a generic procedure, where we in the cas=
e of SHOULD describe when something does not apply. Otherwise it's useless =
when it comes to writing test specifications etc.

Regards,

Christer=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 12. huhtikuuta 2010 16:59
> To: Paul E. Jones
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] FW: I-D=20
> Action:draft-jones-sip-options-ping-00.txt
>=20
> [as individual]
>=20
> Paul E. Jones wrote:
> > Brett,
> >=20
> > So, I think the only change we need in the draft is to=20
> change the word=20
> > "MUST" to "SHOULD" in the paragraph just before Section 8?
>=20
> Let me just interject here that in the past we have=20
> frequently gotten into trouble by using SHOULD without=20
> qualification of what constitutes a valid reason for going=20
> against the SHOULD. Without actual text that clarifies that,=20
> SHOULD is often treated as if it said "you can claim=20
> conformance without implementing this."
>=20
> 	Thanks,
> 	Paul
>=20
> > Paul
> >=20
> >> -----Original Message-----
> >> From: Brett Tate [mailto:brett@broadsoft.com]
> >> Sent: Sunday, April 11, 2010 9:03 AM
> >> To: Paul E. Jones; sipcore@ietf.org
> >> Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> >> 00.txt
> >>
> >>> What in this draft violates the rule?
> >> The draft is currently upgrading rfc3261's honoring Retry-After=20
> >> requirements from MAY (MAY NOT) and SHOULD (SHOULD NOT) levels to=20
> >> MUST (MUST NOT) levels.  Thus it is preventing the UAC from making=20
> >> appropriate/configurable decisions concerning if/how it is=20
> willing to=20
> >> honor/apply the reception of 503 with Retry-After.
> >>
> >> For instance, the draft doesn't allow UAC to reduce the=20
> honoring of=20
> >> 503 with Retry-After value of 1 year to only apply for 30 seconds.
> >>
> >> For instance, the draft doesn't allow UAC to avoid=20
> applying the 503=20
> >> with Retry-After value to sending requests associated with=20
> emergency=20
> >> calls.
> >>
> >> For instance, the draft doesn't allow UAC to avoid=20
> applying the 503=20
> >> with Retry-After to sending requests within dialog.
> >>
> >> For instance, the draft doesn't allow UAC to avoid honoring a 503=20
> >> with Retry-After when response source ip-address is not=20
> equivalent to=20
> >> where the request was sent.  Concerning their not=20
> matching, the draft=20
> >> currently does not mention it from a security risk or otherwise.
> >>
> >> For instance, the draft doesn't allow UAC to only apply=20
> the 503 with=20
> >> Retry-After to specific interfaces.  More specifically if=20
> a UAC knows=20
> >> it should not trust/honor the 503 Retry-After value (because of a=20
> >> proxy rfc3261/draft non compliance or an insecure interface), the=20
> >> draft doesn't allow the device to instead not honor the=20
> 503 with Retry-After.
> >>
> >=20
> >=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
> =

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Subject: Re: [sipcore] draft-jones-sip-options-ping-00: comments
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Paul E. Jones wrote:

> This is a problem general for SIP, since there is no way for a SIP entity to
> know that a DNS query that returns 4 replies relates to the same or
> different servers.  I can't fix that problem in this draft :-)

No, but in the 'normal' case, an INVITE is not directed to a RURI that
contains an IP address and port number, it's directed to a RURI with a
logical name for the 'service', and in that case, it's very logical for
Retry-After returned in the 503 to apply to the entire 'service'.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

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> I think that's OK here, since what Brett was suggesting was that it
> is
> > already a SHOULD in effect in 3261.  In any case, the worst thing
> that
> > happens here is that a device ignores the Retry-After header and
> claims
> > compliance.  Apparently, some feel there are plenty of good reasons
> to do
> > so.
> 
> I don't think that lets you off the hook.
> We've learned from experience that we were too vague in the past.
> So I would expect you to be less vague here - being more explicit about
> what conditions justify violating the SHOULD.

Shall I put it back to MUST?  Keep in mind, we're not trying to re-define
RFC 3261.

Paul



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From: Christer Holmberg <christer.holmberg@ericsson.com>
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Date: Thu, 15 Apr 2010 07:38:34 +0200
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Hi,

I've submitted a new version of the keep-alive draft.

Based on the "lessons learned" during the work on the info-events draft, I'=
ve done some editorial changes.

The TECHNICAL change, however, is that there is no longer a "=3Dyes" value =
added to a "keep" parameter inserted by another entity, because it did not =
work in all cases.

The new solution is much more simple. An entity simply adds a "keep" parame=
ter if it supports keep-alive, and once the route set/path is established e=
ntities will then see whether the adjacent node supports keep-alives or not=
.

An open issue is whether, in the proxy-to-proxy heartbeat/ping case, whethe=
r it should be possible to negotiate a keep-alive lifetime that extends the=
 dialog for which the keep-alives were negotiated. I guess the outcome of t=
hat depends on what/if we are going to do with the options-ping draft.

The draft can also be found at:

http://users.piuha.net/cholmber/drafts/draft-ietf-sipcore-keep-02.txt

Regards,

Christer


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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
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<body>
<font face=3D"Arial, sans-serif" size=3D"3">
<div>&nbsp;</div>
<div><font size=3D"2">Hi,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">I've submitted a new version of the keep-alive draft.=
</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Based on the &quot;lessons learned&quot; during the w=
ork on the info-events draft, I've done some editorial changes.</font></div=
>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">The TECHNICAL change, however, is that there is no lo=
nger a &quot;=3Dyes&quot; value added to a &quot;keep&quot; parameter inser=
ted by another entity, because it did not work in all cases.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">The new solution is much more simple. An entity simpl=
y adds a &quot;keep&quot; parameter if it supports keep-alive, and once the=
 route set/path is established entities will then see whether the adjacent =
node supports keep-alives or not.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">An open issue is whether, in the proxy-to-proxy heart=
beat/ping case, whether it should be possible to negotiate a keep-alive lif=
etime that extends the dialog for which the keep-alives were negotiated. I =
guess the outcome of that depends
on what/if we are going to do with the options-ping draft.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">The draft can also be found at:</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2"><a href=3D"http://users.piuha.net/cholmber/drafts/dra=
ft-ietf-sipcore-keep-02.txt"><font color=3D"#0000FF"><u>http://users.piuha.=
net/cholmber/drafts/draft-ietf-sipcore-keep-02.txt</u></font></a></font></d=
iv>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Regards,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Christer</font></div>
<div><font size=3D"2">&nbsp;</font></div>
</font>
</body>
</html>

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


	Title           : Indication of support for keep-alive
	Author(s)       : C. Holmberg
	Filename        : draft-ietf-sipcore-keep-02.txt
	Pages           : 15
	Date            : 2010-04-14

This specification defines a new Session Initiation Protocol (SIP)
header field parameter, "keep", that SIP entities can use to
negotiate explicit support of the NAT keep-alive mechanisms defined
in the SIP Outbound specification.  The parameter is defined for
cases where the SIP Outbound mechanism is not supported, or in cases
where it cannot be applied.

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Cc: sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul, Brett,

I looked at the OPTIONS ping draft and 3261 again.  Perhaps I'm missing it
and need to be reminded (and I note it's 2AM here), but what in the OPTIONS
draft changes anything stated in 3261.  The statement of concern is this,
right?

   To avoid unduly taxing a receiving SIP entity, transmitters of
   OPTIONS messages MUST honor the Retry-After header field if
   received.

When a 503 is used is in line with 3261: we're not suggesting new
opportunities to send one. We're just saying that code should be used vs.
other codes that don't have the same meaning.

And, Retry-After should be honored.  We say MUST, which I understand is a
stronger word than SHOULD (and why I'm willing to soften it).  Changing it
to SHOULD would be in stronger agreement with 21.5.4 of 3261.

A device would not send 503 with Retry-After lightly, nor are we proposing a
change as to when a 503 is sent.  So, is using "SHOULD" acceptable? Or is
this really a problem?  I understand the problems one can get into with the
current wording in 3261, but what I'm failing to see, I guess, is how the
OPTIONS ping draft makes this worse.

Paul

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Monday, April 12, 2010 8:25 PM
> To: Brett Tate
> Cc: Hadriel Kaplan; Paul E. Jones; sipcore@ietf.org
> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
> 
> 
> 
> Brett Tate wrote:
> >>> I don't think that lets you off the hook.
> >>> We've learned from experience that we were too vague in
> >>> the past.  So I would expect you to be less vague here -
> >>> being more explicit about what conditions justify violating
> >>> the SHOULD.
> >> And really using "MUST..., unless..." instead.
> >
> > Is making 503 with Retry-After usable (i.e. so it does more good than
> harm) something that should be done within this draft and in this
> working group?  Or is that something that should be addressed within
> the hopefully dispatched overload working group?
> 
> 
> Well, now you ask the hard questions.
> 
> With my chair hat on, IMO any changes to the behavior of 503 will at
> least need to be blessed by sipcore. (But I'm new as a sipcore chair,
> so
> I'm willing to learn from the other chair or the AD that is isn't so.)
> 
> After that, I think it depends on what is to happen with this draft.
> 
> If it is intended to be informational, then it can't make any normative
> changes. It can make any recommendations it wants as long as they don't
> require violating normative behavior.
> 
> If its aspirations are higher, and maybe even if they aren't, then I
> think it makes sense to coordinate it with the work of the overload wg.
> (AFAIK we don't yet *have* an overload WG, but there is a mailing list,
> and I gather its expected that we will have the WG before long.)
> 
> With my individual hat on I will recommend that it go there for some
> discussion of if/how it relates.
> 
> 	Thanks,
> 	Paul
> 



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> > This is a problem general for SIP, since there is no way for a SIP
> entity to
> > know that a DNS query that returns 4 replies relates to the same or
> > different servers.  I can't fix that problem in this draft :-)
> 
> No, but in the 'normal' case, an INVITE is not directed to a RURI that
> contains an IP address and port number, it's directed to a RURI with a
> logical name for the 'service', and in that case, it's very logical for
> Retry-After returned in the 503 to apply to the entire 'service'.

Messages are still transmitted to a certain server.  RFC 3261 provides
guidance as to when an entity should stop transmitting messages to a given
server. The OPTIONS ping draft does not change that logic.

Paul



From john.elwell@siemens-enterprise.com  Thu Apr 15 01:19:35 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 15 Apr 2010 10:19:08 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-02
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Christer,

Why is this draft-ietf? I don't recall adopting anything of this nature as =
a WG item. I have a major concern that the draft assumes a very specific to=
pology. I remember discussing similar concerns on earlier iterations, but t=
his new draft makes things worse, if anything. Details below:

1. For UAs that don't perform registration (e.g., gateways, media servers, =
B2BUAs typically don't do registration), the provisions of section 4 seem a=
 bit restrictive. For example, between a proxy and a gateway, the gateway m=
ust do the keep-alives, not the proxy. I don't necessarily have a problem w=
ith this, but I wondered why there is this constraint. What I do have a pro=
blem with is the UA-to-UA case (in particular, B2BUA-to-B2BUA), where, if t=
his mechanism is to be used, clearly one of the UAs has to receive keep-ali=
ves.

2. Also section 4.1 starts off by saying that this is all about keep-alive =
between a UA and its edge proxy, which tends to rule out the cases above, a=
nd also cases where a registering UA registers directly with its registrar,=
 not via an edge proxy. Also it seems to rule out cases where the proxy doe=
s not record-route, so that for dialog keep-alive the UA would need to send=
 keep-alives to some other entity.

3. In 4.2.2:
"When the UAC receives the response(s) to the initial request for the
   dialog, if the top-most Record-Route header field (edge proxy) of
   response(s) contains a "keep" parameter, the UAC MUST start to send
   keep-alives towards the edge proxy for the remaining duration of the
   dialog."
This doesn't make sense. The response containing the "keep" parameter will =
be received on the path of the  request, but depending which entities have =
record-routed, the dialog path may be different. A transport connection to =
the first entity on the dialog path is not established until the next outbo=
und or inbound request on that dialog, so "MUST start to send keep-alives t=
owards the edge proxy" is unachievable in such cases.

4. In 4.3.1:
"When a UAS receives an initial request for a dialog, if the top-most
   Record-Route header filed contains a "keep" parameter, and if the UAS
   is willing to send keep-alives associated with the dialog, it MUST
   insert a "keep" parameter in its Contact header field in the
   response(s) to the request.  After that the UAS MUST start to send
   keep-alives towards the edge proxy, and it MUST send keep-alives for
   the remaining duration of the registration."
I have a similar problem with this.

5. In 5.3.1:
"When a proxy receives a dialog initiation request from its previous-
   hop proxy, and the top-most Record-Route header field of the request
   contains a "keep" parameter, if the proxy is willing to send and
   receive keep-alives from the previous-hop proxy for the associated
   dialog, the proxy MUST insert a "keep" parameter in its Record-Route
   header field of the associated response(s) sent towards the previous-
   hop proxy.  After that the proxy MUST send keep-alives, and accept
   keep-alives sent from the previous-hop proxy, for the duration of the
   dialog."
I have a similar problem with this. If the top-most Record-Route header fie=
ld does not represent the previous hop proxy, there is no transport connect=
ion established to allow you to start sending keep-alives at this time.

6. "When a proxy forwards a dialog initiation request towards its next-
   hop proxy, and it wants to negotiate the usage of keep-alives with
   that next-hop proxy, it MUST include a "keep" parameter in its
   Record-Record header field of the request."
How does the proxy know that the next hop is a proxy, as opposed to a UA? A=
ssuming it does know that the next hop is a proxy, how does it know that th=
e next hop proxy will record-route?

John


> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: 15 April 2010 06:39
> To: sipcore@ietf.org
> Subject: [sipcore] Draft new version: draft-ietf-sipcore-keep-02
>=20
> =20
> Hi,
> =20
> I've submitted a new version of the keep-alive draft.
> =20
> Based on the "lessons learned" during the work on the=20
> info-events draft, I've done some editorial changes.
> =20
> The TECHNICAL change, however, is that there is no longer a=20
> "=3Dyes" value added to a "keep" parameter inserted by another=20
> entity, because it did not work in all cases.
> =20
> The new solution is much more simple. An entity simply adds a=20
> "keep" parameter if it supports keep-alive, and once the=20
> route set/path is established entities will then see whether=20
> the adjacent node supports keep-alives or not.
> =20
> An open issue is whether, in the proxy-to-proxy=20
> heartbeat/ping case, whether it should be possible to=20
> negotiate a keep-alive lifetime that extends the dialog for=20
> which the keep-alives were negotiated. I guess the outcome of=20
> that depends on what/if we are going to do with the=20
> options-ping draft.
> =20
> The draft can also be found at:
> =20
> http://users.piuha.net/cholmber/drafts/draft-ietf-sipcore-keep
> -02.txt=20
> <http://users.piuha.net/cholmber/drafts/draft-ietf-sipcore-kee
> p-02.txt>=20
> =20
> Regards,
> =20
> Christer
> =

From christer.holmberg@ericsson.com  Thu Apr 15 02:43:40 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 15 Apr 2010 11:43:30 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-02
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Hi John,

>Why is this draft-ietf? I don't recall adopting anything of this nature as=
 a WG item.=20

The draft has been adopted a long time ago. It has just been on hold for a =
while, partly because of the INFO work.=20

Before answering to your specific comments, I want to clarify one thing:

When I talk about "next hop" and "previous hop", I refer to the adjacent no=
de in the established registration/dialog path - not a hop which will not b=
e part of that path. That needs to be clarfied.

And, this is the reason, based on a comment from yourself, that we decided =
to NOT define the "keep" parameter for the Via header field, since not all =
entities that insert Via will be part of the path.

--------

>I have a major concern that the draft assumes a very specific topology. I =
remember discussing similar concerns on earlier iterations, but this new dr=
aft=20
>makes things worse, if anything. Details below:
>=20
>1. For UAs that don't perform registration (e.g., gateways,=20
>media servers, B2BUAs typically don't do registration), the=20
>provisions of section 4 seem a bit restrictive. For example,=20
>between a proxy and a gateway, the gateway must do the=20
>keep-alives, not the proxy. I don't necessarily have a=20
>problem with this, but I wondered why there is this=20
>constraint.

The idea is that keep-alives would not be sent towards UAs. It is the UAs t=
hat need to send keep-alives in order to keep the NAT binding open.

--------

>What I do have a problem with is the UA-to-UA case (in particular, B2BUA-t=
o-B2BUA), where, if this mechanism is to be used, clearly one of the UAs ha=
s to=20
>receive keep-alives.

The initial scope of the draft was to use the Outbound topology, and assume=
 there is an edge proxy. After that people asked to also cover the proxy-to=
-proxy case.

However, if we want to cover the direct UA-to-UA case I guess we could say =
that the UAs check for a Record-Route with "keep", and if there is none the=
n check for a Contact with "keep".

--------

>2. Also section 4.1 starts off by saying that this is all=20
>about keep-alive between a UA and its edge proxy, which tends=20
>to rule out the cases above, and also cases where a=20
>registering UA registers directly with its registrar, not via=20
>an edge proxy. Also it seems to rule out cases where the=20
>proxy does not record-route, so that for dialog keep-alive=20
>the UA would need to send keep-alives to some other entity.

See my comment about the scope of the draft above.

However, as for the UA-to-UA case, I don't have a problem if we want to als=
o cover the direct UA-to-Registrar case. I guess the question then is how t=
he registrar will indicate its support of keep-alives, since it normally do=
esn't insert a Path header field.

--------

>3. In 4.2.2:
>"When the UAC receives the response(s) to the initial request for the
>dialog, if the top-most Record-Route header field (edge proxy) of
>response(s) contains a "keep" parameter, the UAC MUST start to send
>keep-alives towards the edge proxy for the remaining=20
>duration of the dialog."
>This doesn't make sense. The response containing the "keep"=20
>parameter will be received on the path of the  request, but=20
>depending which entities have record-routed, the dialog path=20
>may be different. A transport connection to the first entity=20
>on the dialog path is not established until the next outbound=20
>or inbound request on that dialog, so "MUST start to send=20
>keep-alives towards the edge proxy" is unachievable in such cases.

The UAC will send the keep-alives towards the address indicated in the Reco=
rd-Route - NOT to where it sent the INVITE request.

--------

>4. In 4.3.1:
>"When a UAS receives an initial request for a dialog, if the top-most
>Record-Route header filed contains a "keep" parameter, and=20
>if the UAS is willing to send keep-alives associated with the dialog, it M=
UST
>insert a "keep" parameter in its Contact header field in the
>response(s) to the request.  After that the UAS MUST start to send
>keep-alives towards the edge proxy, and it MUST send keep-alives for
>the remaining duration of the registration."
>I have a similar problem with this.

Again, the UAS will send the keep-alives towards the address indicated in t=
he Record-Route.

--------

>5. In 5.3.1:
>"When a proxy receives a dialog initiation request from its previous-
>hop proxy, and the top-most Record-Route header field of the request
>contains a "keep" parameter, if the proxy is willing to send and
>receive keep-alives from the previous-hop proxy for the associated
>dialog, the proxy MUST insert a "keep" parameter in its=20
>Record-Route header field of the associated response(s) sent towards=20
>the previous-hop proxy.  After that the proxy MUST send keep-alives, and a=
ccept
>keep-alives sent from the previous-hop proxy, for the duration of the
>dialog."
>I have a similar problem with this. If the top-most=20
>Record-Route header field does not represent the previous hop=20
>proxy, there is no transport connection established to allow=20
>you to start sending keep-alives at this time.

As I said in the beginning of the reply, "next hop" and "previous hop" are =
probably confusing terminology - at least without proper explanation.

--------

>6. "When a proxy forwards a dialog initiation request towards=20
>its next- hop proxy, and it wants to negotiate the usage of keep-alives wi=
th
>that next-hop proxy, it MUST include a "keep" parameter in its
>Record-Record header field of the request."
>How does the proxy know that the next hop is a proxy, as=20
>opposed to a UA?=20

I guess that would be a edge-to-UA case. But, I guess that could be clarifi=
ed.

--------

>Assuming it does know that the next hop is a proxy, how does it know that =
the next hop proxy will record-route?

See previous comments on "next hop" and "previous hop" terminology.

Regards,

Christer

From john.elwell@siemens-enterprise.com  Thu Apr 15 03:27:13 2010
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To: Christer Holmberg <christer.holmberg@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 15 April 2010 10:44
> To: Elwell, John; sipcore@ietf.org
> Subject: RE: Draft new version: draft-ietf-sipcore-keep-02
>=20
>=20
> Hi John,
>=20
> >Why is this draft-ietf? I don't recall adopting anything of=20
> this nature as a WG item.=20
>=20
> The draft has been adopted a long time ago. It has just been=20
> on hold for a while, partly because of the INFO work.=20
[JRE] I know, but this draft looks very different from the one adopted as W=
G item, but never mind, the important thing is to address the comments.

>=20
> Before answering to your specific comments, I want to clarify=20
> one thing:
>=20
> When I talk about "next hop" and "previous hop", I refer to=20
> the adjacent node in the established registration/dialog path=20
> - not a hop which will not be part of that path. That needs=20
> to be clarfied.
>=20
> And, this is the reason, based on a comment from yourself,=20
> that we decided to NOT define the "keep" parameter for the=20
> Via header field, since not all entities that insert Via will=20
> be part of the path.
[JRE] Yes, I understand the problems with the old Via-based proposal, but o=
n reading the present draft it didn't seem to help much.

>=20
> --------
>=20
> >I have a major concern that the draft assumes a very=20
> specific topology. I remember discussing similar concerns on=20
> earlier iterations, but this new draft=20
> >makes things worse, if anything. Details below:
> >=20
> >1. For UAs that don't perform registration (e.g., gateways,=20
> >media servers, B2BUAs typically don't do registration), the=20
> >provisions of section 4 seem a bit restrictive. For example,=20
> >between a proxy and a gateway, the gateway must do the=20
> >keep-alives, not the proxy. I don't necessarily have a=20
> >problem with this, but I wondered why there is this=20
> >constraint.
>=20
> The idea is that keep-alives would not be sent towards UAs.=20
> It is the UAs that need to send keep-alives in order to keep=20
> the NAT binding open.
>=20
> --------
>=20
> >What I do have a problem with is the UA-to-UA case (in=20
> particular, B2BUA-to-B2BUA), where, if this mechanism is to=20
> be used, clearly one of the UAs has to=20
> >receive keep-alives.
>=20
> The initial scope of the draft was to use the Outbound=20
> topology, and assume there is an edge proxy. After that=20
> people asked to also cover the proxy-to-proxy case.
>=20
> However, if we want to cover the direct UA-to-UA case I guess=20
> we could say that the UAs check for a Record-Route with=20
> "keep", and if there is none then check for a Contact with "keep".
[JRE] Many people use B2BUAs rather than proxies, and I would expect the dr=
aft to be equally applicable between B2BUAs. Also I would have expected the=
 draft to be applicable between a proxy/B2BUA and a non-registering UA, or =
between a proxy/B2BUA and a directly registering UA (without an edge proxy)=
. In other words, filling all the gaps that aren't covered by sip-outbound.

>=20
> --------
>=20
> >2. Also section 4.1 starts off by saying that this is all=20
> >about keep-alive between a UA and its edge proxy, which tends=20
> >to rule out the cases above, and also cases where a=20
> >registering UA registers directly with its registrar, not via=20
> >an edge proxy. Also it seems to rule out cases where the=20
> >proxy does not record-route, so that for dialog keep-alive=20
> >the UA would need to send keep-alives to some other entity.
>=20
> See my comment about the scope of the draft above.
>=20
> However, as for the UA-to-UA case, I don't have a problem if=20
> we want to also cover the direct UA-to-Registrar case. I=20
> guess the question then is how the registrar will indicate=20
> its support of keep-alives, since it normally doesn't insert=20
> a Path header field.
[JRE] Well, the previous proposal used the Via header field, so in moving a=
way from Via we seem to have lost a capability. Perhaps a UA wanting to use=
 this mechanism must insert a Path header field - I haven't thought through=
 the consequences, however.

>=20
> --------
>=20
> >3. In 4.2.2:
> >"When the UAC receives the response(s) to the initial request for the
> >dialog, if the top-most Record-Route header field (edge proxy) of
> >response(s) contains a "keep" parameter, the UAC MUST start to send
> >keep-alives towards the edge proxy for the remaining=20
> >duration of the dialog."
> >This doesn't make sense. The response containing the "keep"=20
> >parameter will be received on the path of the  request, but=20
> >depending which entities have record-routed, the dialog path=20
> >may be different. A transport connection to the first entity=20
> >on the dialog path is not established until the next outbound=20
> >or inbound request on that dialog, so "MUST start to send=20
> >keep-alives towards the edge proxy" is unachievable in such cases.
>=20
> The UAC will send the keep-alives towards the address=20
> indicated in the Record-Route - NOT to where it sent the=20
> INVITE request.
[JRE] How can it sent keep-alives when it hasn't established a transport co=
nnection yet? It must wait for the transport connection to be established, =
which means waiting for an inbound or outbound mid-dialog request. If it's =
an inbound request, however, which entity should be responsible for keep-al=
ives?

>=20
> --------
>=20
> >4. In 4.3.1:
> >"When a UAS receives an initial request for a dialog, if the top-most
> >Record-Route header filed contains a "keep" parameter, and=20
> >if the UAS is willing to send keep-alives associated with=20
> the dialog, it MUST
> >insert a "keep" parameter in its Contact header field in the
> >response(s) to the request.  After that the UAS MUST start to send
> >keep-alives towards the edge proxy, and it MUST send keep-alives for
> >the remaining duration of the registration."
> >I have a similar problem with this.
>=20
> Again, the UAS will send the keep-alives towards the address=20
> indicated in the Record-Route.
[JRE] Same problem.

>=20
> --------
>=20
> >5. In 5.3.1:
> >"When a proxy receives a dialog initiation request from its previous-
> >hop proxy, and the top-most Record-Route header field of the request
> >contains a "keep" parameter, if the proxy is willing to send and
> >receive keep-alives from the previous-hop proxy for the associated
> >dialog, the proxy MUST insert a "keep" parameter in its=20
> >Record-Route header field of the associated response(s) sent towards=20
> >the previous-hop proxy.  After that the proxy MUST send=20
> keep-alives, and accept
> >keep-alives sent from the previous-hop proxy, for the duration of the
> >dialog."
> >I have a similar problem with this. If the top-most=20
> >Record-Route header field does not represent the previous hop=20
> >proxy, there is no transport connection established to allow=20
> >you to start sending keep-alives at this time.
>=20
> As I said in the beginning of the reply, "next hop" and=20
> "previous hop" are probably confusing terminology - at least=20
> without proper explanation.
[JRE] OK, but this still doesn't address my concern about the absence of a =
transport connection.

>=20
> --------
>=20
> >6. "When a proxy forwards a dialog initiation request towards=20
> >its next- hop proxy, and it wants to negotiate the usage of=20
> keep-alives with
> >that next-hop proxy, it MUST include a "keep" parameter in its
> >Record-Record header field of the request."
> >How does the proxy know that the next hop is a proxy, as=20
> >opposed to a UA?=20
>=20
> I guess that would be a edge-to-UA case. But, I guess that=20
> could be clarified.
[JRE] As a proxy, how do I know I am the "edge", i.e., how do I know the ne=
xt entity is a UA? Even in the IMS P-CSCF case, I don't know - there could =
be a PBX behind it.

>=20
> --------
>=20
> >Assuming it does know that the next hop is a proxy, how does=20
> it know that the next hop proxy will record-route?
>=20
> See previous comments on "next hop" and "previous hop" terminology.
[JRE] The words "When a proxy forwards a dialog initiation request towards =
its next-hop proxy" do not make sense - a proxy simply forwards a request t=
o the next hop entity. Until it receives a response, it has no idea whether=
 the entity it forwarded the request to is a UA, a non-record-routing proxy=
 or a record-routing proxy, i.e., it has no idea whether it has sent it to =
something that complies with your definition of "next-hop proxy".

John

From christer.holmberg@ericsson.com  Thu Apr 15 03:57:32 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 15 Apr 2010 12:57:13 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-02
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-keep-02
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Hi John,=20

>>Why is this draft-ietf? I don't recall adopting anything of this nature a=
s a WG item.=20
>>=20
>>The draft has been adopted a long time ago. It has just been on hold=20
>>for a while, partly because of the INFO work.
>[JRE] I know, but this draft looks very different from the one adopted as =
WG item,

Please compare the first and last version of the Info-events draft :)

Seriously, I agree there were editorial changes, but the only technical cha=
nge was to not use the "=3Dyes" parameter. And, some of the editorial chang=
es are also based on that change.
=20
>>Before answering to your specific comments, I want to clarify one thing:
>>=20
>>When I talk about "next hop" and "previous hop", I refer to=20
>>the adjacent node in the established registration/dialog path=20
>>- not a hop which will not be part of that path. That needs=20
>>to be clarfied.
>>=20
>>And, this is the reason, based on a comment from yourself,=20
>>that we decided to NOT define the "keep" parameter for the=20
>>Via header field, since not all entities that insert Via will=20
>>be part of the path.
>[JRE] Yes, I understand the problems with the old Via-based=20
>proposal, but on reading the present draft it didn't seem to=20
>help much.

Keep in mind that in Stockholm we decided to use Contact/Path/Record-Route =
instead of Via, so that is an existing agreement.

The change in this version is to not use the "=3Dyes" value.

But, as I said earlier, the "next/previous hop" terminology is confusing, b=
ut I *AM* referring to adjacent nodes in the established path :)

--------

>>>I have a major concern that the draft assumes a very=20
>>>specific topology. I remember discussing similar concerns on=20
>>>earlier iterations, but this new draft=20
>>>makes things worse, if anything. Details below:
>>>=20
>>>1. For UAs that don't perform registration (e.g., gateways,=20
>>>media servers, B2BUAs typically don't do registration), the=20
>>>provisions of section 4 seem a bit restrictive. For example,=20
>>>between a proxy and a gateway, the gateway must do the=20
>>>keep-alives, not the proxy. I don't necessarily have a=20
>>>problem with this, but I wondered why there is this=20
>>>constraint.
>>=20
>>The idea is that keep-alives would not be sent towards UAs.=20
>>It is the UAs that need to send keep-alives in order to keep=20
>>the NAT binding open.
>>=20
>>--------
>>=20
>>>What I do have a problem with is the UA-to-UA case (in=20
>>>particular, B2BUA-to-B2BUA), where, if this mechanism is to=20
>>>be used, clearly one of the UAs has to=20
>>>receive keep-alives.
>>=20
>>The initial scope of the draft was to use the Outbound=20
>>topology, and assume there is an edge proxy. After that=20
>>people asked to also cover the proxy-to-proxy case.
>>=20
>>However, if we want to cover the direct UA-to-UA case I guess=20
>>we could say that the UAs check for a Record-Route with=20
>>"keep", and if there is none then check for a Contact with "keep".
>[JRE] Many people use B2BUAs rather than proxies, and I would=20
>expect the draft to be equally applicable between B2BUAs.=20
>Also I would have expected the draft to be applicable between=20
>a proxy/B2BUA and a non-registering UA, or between a=20
>proxy/B2BUA and a directly registering UA (without an edge=20
>proxy). In other words, filling all the gaps that aren't=20
>covered by sip-outbound.

Since a B2BUA acts as a UA, I don't we need to separate them. If we solve U=
A-to-UA we also solve UA-to-B2BUA.


----------


>>>2. Also section 4.1 starts off by saying that this is all=20
>>>about keep-alive between a UA and its edge proxy, which tends=20
>>>to rule out the cases above, and also cases where a=20
>>>registering UA registers directly with its registrar, not via=20
>>>an edge proxy. Also it seems to rule out cases where the=20
>>>proxy does not record-route, so that for dialog keep-alive=20
>>>the UA would need to send keep-alives to some other entity.
>>=20
>>See my comment about the scope of the draft above.
>>=20
>>However, as for the UA-to-UA case, I don't have a problem if=20
>>we want to also cover the direct UA-to-Registrar case. I=20
>>guess the question then is how the registrar will indicate=20
>>its support of keep-alives, since it normally doesn't insert=20
>>a Path header field.
>[JRE] Well, the previous proposal used the Via header field,=20
>so in moving away from Via we seem to have lost a capability.=20
>Perhaps a UA wanting to use this mechanism must insert a Path=20
>header field - I haven't thought through the consequences, however.

The UA inserts Contact so that is not a problem. The question is how the re=
gistrar would indicate support.


---------


>>>3. In 4.2.2:
>>>"When the UAC receives the response(s) to the initial=20
>>>request for the dialog, if the top-most Record-Route header field (edge =
proxy) of
>>>response(s) contains a "keep" parameter, the UAC MUST start to send
>>>keep-alives towards the edge proxy for the remaining=20
>>>duration of the dialog."
>>>This doesn't make sense. The response containing the "keep"=20
>>>parameter will be received on the path of the  request, but=20
>>>depending which entities have record-routed, the dialog path=20
>>>may be different. A transport connection to the first entity=20
>>>on the dialog path is not established until the next outbound=20
>>>or inbound request on that dialog, so "MUST start to send=20
>>>keep-alives towards the edge proxy" is unachievable in such cases.
>>=20
>>The UAC will send the keep-alives towards the address=20
>>indicated in the Record-Route - NOT to where it sent the=20
>>INVITE request.
>[JRE] How can it sent keep-alives when it hasn't established=20
>a transport connection yet? It must wait for the transport=20
>connection to be established, which means waiting for an=20
>inbound or outbound mid-dialog request. If it's an inbound=20
>request, however, which entity should be responsible for keep-alives?

What prevents the UAC from establishing the transport connection without wa=
iting for any mid-dialog request to be sent?

And, unless the UAC establishes the connnection, it won't even be able to r=
eceive inbound requests in case there is a NAT.


---------


>>>4. In 4.3.1:
>>>"When a UAS receives an initial request for a dialog, if=20
>>>the top-most Record-Route header filed contains a "keep" parameter, and=
=20
>>>if the UAS is willing to send keep-alives associated with=20
>>>the dialog, it MUST insert a "keep" parameter in its Contact header fiel=
d in the
>>>response(s) to the request.  After that the UAS MUST start to send
>>>keep-alives towards the edge proxy, and it MUST send keep-alives for
>>>the remaining duration of the registration."
>>>I have a similar problem with this.
>>=20
>>Again, the UAS will send the keep-alives towards the address=20
>>indicated in the Record-Route.
>[JRE] Same problem.

Same answer :)


---------


>>>5. In 5.3.1:
>>>"When a proxy receives a dialog initiation request from=20
>>>its previous-hop proxy, and the top-most Record-Route header field of=20
>>>the request contains a "keep" parameter, if the proxy is willing to send=
 and
>>>receive keep-alives from the previous-hop proxy for the associated
>>>dialog, the proxy MUST insert a "keep" parameter in its=20
>>>Record-Route header field of the associated response(s) sent towards=20
>>>the previous-hop proxy.  After that the proxy MUST send keep-alives, and=
 accept
>>>keep-alives sent from the previous-hop proxy, for the duration of the
>>>dialog."
>>>I have a similar problem with this. If the top-most=20
>>>Record-Route header field does not represent the previous hop=20
>>>proxy, there is no transport connection established to allow=20
>>>you to start sending keep-alives at this time.
>>=20
>>As I said in the beginning of the reply, "next hop" and=20
>>"previous hop" are probably confusing terminology - at least=20
>>without proper explanation.
>[JRE] OK, but this still doesn't address my concern about the=20
>absence of a transport connection.

See above.


--------


>>>6. "When a proxy forwards a dialog initiation request towards=20
>>>its next- hop proxy, and it wants to negotiate the usage of=20
>>>keep-alives with
>>>that next-hop proxy, it MUST include a "keep" parameter in its
>>>Record-Record header field of the request."
>>>How does the proxy know that the next hop is a proxy, as=20
>>>opposed to a UA?=20
>>=20
>>I guess that would be a edge-to-UA case. But, I guess that=20
>>could be clarified.
>[JRE] As a proxy, how do I know I am the "edge", i.e., how do=20
>I know the next entity is a UA? Even in the IMS P-CSCF case,=20
>I don't know - there could be a PBX behind it.

When the proxy get the response, if there are no Record-Routes between the =
proxy and the UA, the proxy is adjacent to the UA.


---------

>>>Assuming it does know that the next hop is a proxy, how does=20
>>>it know that the next hop proxy will record-route?
>>=20
>>See previous comments on "next hop" and "previous hop" terminology.
>[JRE] The words "When a proxy forwards a dialog initiation=20
>request towards its next-hop proxy" do not make sense - a=20
>proxy simply forwards a request to the next hop entity. Until=20
>it receives a response, it has no idea whether the entity it=20
>forwarded the request to is a UA, a non-record-routing proxy=20
>or a record-routing proxy, i.e., it has no idea whether it=20
>has sent it to something that complies with your definition=20
>of "next-hop proxy".

I agree. So, the text could probably be more generic, and the actions then =
depend on what comes in the answer (Record-Route or Contact).


Regards,

Christer

From pkyzivat@cisco.com  Thu Apr 15 05:55:12 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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I'm ok with the SHOULD since that is what is in 3261. What I would 
prefer to see is some enumeration of the cases when its appropriate to 
violate the SHOULD.

	Thanks,
	Paul

Paul E. Jones wrote:
>> I think that's OK here, since what Brett was suggesting was that it
>> is
>>> already a SHOULD in effect in 3261.  In any case, the worst thing
>> that
>>> happens here is that a device ignores the Retry-After header and
>> claims
>>> compliance.  Apparently, some feel there are plenty of good reasons
>> to do
>>> so.
>> I don't think that lets you off the hook.
>> We've learned from experience that we were too vague in the past.
>> So I would expect you to be less vague here - being more explicit about
>> what conditions justify violating the SHOULD.
> 
> Shall I put it back to MUST?  Keep in mind, we're not trying to re-define
> RFC 3261.
> 
> Paul
> 
> 
> 

From john.elwell@siemens-enterprise.com  Thu Apr 15 06:31:39 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 15 Apr 2010 15:27:41 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-02
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-keep-02
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 15 April 2010 11:57
> To: Elwell, John; sipcore@ietf.org
> Subject: RE: Draft new version: draft-ietf-sipcore-keep-02
>=20
>=20
> Hi John,=20
>=20
> >>Why is this draft-ietf? I don't recall adopting anything of=20
> this nature as a WG item.=20
> >>=20
> >>The draft has been adopted a long time ago. It has just=20
> been on hold=20
> >>for a while, partly because of the INFO work.
> >[JRE] I know, but this draft looks very different from the=20
> one adopted as WG item,
>=20
> Please compare the first and last version of the Info-events draft :)
>=20
> Seriously, I agree there were editorial changes, but the only=20
> technical change was to not use the "=3Dyes" parameter. And,=20
> some of the editorial changes are also based on that change.
[JRE] This is not true. -00 and -01 both used Via, now we use Path, Record-=
Route and Contact. That is a completely different solution. I agree that -0=
2 might be moving in the right direction.

> =20
> >>Before answering to your specific comments, I want to=20
> clarify one thing:
> >>=20
> >>When I talk about "next hop" and "previous hop", I refer to=20
> >>the adjacent node in the established registration/dialog path=20
> >>- not a hop which will not be part of that path. That needs=20
> >>to be clarfied.
> >>=20
> >>And, this is the reason, based on a comment from yourself,=20
> >>that we decided to NOT define the "keep" parameter for the=20
> >>Via header field, since not all entities that insert Via will=20
> >>be part of the path.
> >[JRE] Yes, I understand the problems with the old Via-based=20
> >proposal, but on reading the present draft it didn't seem to=20
> >help much.
>=20
> Keep in mind that in Stockholm we decided to use=20
> Contact/Path/Record-Route instead of Via, so that is an=20
> existing agreement.
[JRE] Well, I could not find any mailing list endorsement of this following=
 Stockholm, and that is nearly one year ago, so my memory failed me. The mi=
nutes just state: "Option 2 from slides: "Nodes which are adjacent in a rou=
te set/registration path"", and don't say anything about the header fields.=
 Unfortunately the link to the slides is broken. Anyway, we should focus on=
 fixing the present draft.

>=20
> The change in this version is to not use the "=3Dyes" value.
>=20
> But, as I said earlier, the "next/previous hop" terminology=20
> is confusing, but I *AM* referring to adjacent nodes in the=20
> established path :)
>=20
> --------
>=20
> >>>I have a major concern that the draft assumes a very=20
> >>>specific topology. I remember discussing similar concerns on=20
> >>>earlier iterations, but this new draft=20
> >>>makes things worse, if anything. Details below:
> >>>=20
> >>>1. For UAs that don't perform registration (e.g., gateways,=20
> >>>media servers, B2BUAs typically don't do registration), the=20
> >>>provisions of section 4 seem a bit restrictive. For example,=20
> >>>between a proxy and a gateway, the gateway must do the=20
> >>>keep-alives, not the proxy. I don't necessarily have a=20
> >>>problem with this, but I wondered why there is this=20
> >>>constraint.
> >>=20
> >>The idea is that keep-alives would not be sent towards UAs.=20
> >>It is the UAs that need to send keep-alives in order to keep=20
> >>the NAT binding open.
> >>=20
> >>--------
> >>=20
> >>>What I do have a problem with is the UA-to-UA case (in=20
> >>>particular, B2BUA-to-B2BUA), where, if this mechanism is to=20
> >>>be used, clearly one of the UAs has to=20
> >>>receive keep-alives.
> >>=20
> >>The initial scope of the draft was to use the Outbound=20
> >>topology, and assume there is an edge proxy. After that=20
> >>people asked to also cover the proxy-to-proxy case.
> >>=20
> >>However, if we want to cover the direct UA-to-UA case I guess=20
> >>we could say that the UAs check for a Record-Route with=20
> >>"keep", and if there is none then check for a Contact with "keep".
> >[JRE] Many people use B2BUAs rather than proxies, and I would=20
> >expect the draft to be equally applicable between B2BUAs.=20
> >Also I would have expected the draft to be applicable between=20
> >a proxy/B2BUA and a non-registering UA, or between a=20
> >proxy/B2BUA and a directly registering UA (without an edge=20
> >proxy). In other words, filling all the gaps that aren't=20
> >covered by sip-outbound.
>=20
> Since a B2BUA acts as a UA, I don't we need to separate them.=20
> If we solve UA-to-UA we also solve UA-to-B2BUA.
[JRE] Fine, but the present draft doesn't seem to work for UA-to-UA.

>=20
> >>>2. Also section 4.1 starts off by saying that this is all=20
> >>>about keep-alive between a UA and its edge proxy, which tends=20
> >>>to rule out the cases above, and also cases where a=20
> >>>registering UA registers directly with its registrar, not via=20
> >>>an edge proxy. Also it seems to rule out cases where the=20
> >>>proxy does not record-route, so that for dialog keep-alive=20
> >>>the UA would need to send keep-alives to some other entity.
> >>=20
> >>See my comment about the scope of the draft above.
> >>=20
> >>However, as for the UA-to-UA case, I don't have a problem if=20
> >>we want to also cover the direct UA-to-Registrar case. I=20
> >>guess the question then is how the registrar will indicate=20
> >>its support of keep-alives, since it normally doesn't insert=20
> >>a Path header field.
> >[JRE] Well, the previous proposal used the Via header field,=20
> >so in moving away from Via we seem to have lost a capability.=20
> >Perhaps a UA wanting to use this mechanism must insert a Path=20
> >header field - I haven't thought through the consequences, however.
>=20
> The UA inserts Contact so that is not a problem. The question=20
> is how the registrar would indicate support.
[JRE] If the registrar receives Path in the request, it could send Path in =
the response.

>=20
>=20
> ---------
>=20
>=20
> >>>3. In 4.2.2:
> >>>"When the UAC receives the response(s) to the initial=20
> >>>request for the dialog, if the top-most Record-Route=20
> header field (edge proxy) of
> >>>response(s) contains a "keep" parameter, the UAC MUST start to send
> >>>keep-alives towards the edge proxy for the remaining=20
> >>>duration of the dialog."
> >>>This doesn't make sense. The response containing the "keep"=20
> >>>parameter will be received on the path of the  request, but=20
> >>>depending which entities have record-routed, the dialog path=20
> >>>may be different. A transport connection to the first entity=20
> >>>on the dialog path is not established until the next outbound=20
> >>>or inbound request on that dialog, so "MUST start to send=20
> >>>keep-alives towards the edge proxy" is unachievable in such cases.
> >>=20
> >>The UAC will send the keep-alives towards the address=20
> >>indicated in the Record-Route - NOT to where it sent the=20
> >>INVITE request.
> >[JRE] How can it sent keep-alives when it hasn't established=20
> >a transport connection yet? It must wait for the transport=20
> >connection to be established, which means waiting for an=20
> >inbound or outbound mid-dialog request. If it's an inbound=20
> >request, however, which entity should be responsible for keep-alives?
>=20
> What prevents the UAC from establishing the transport=20
> connection without waiting for any mid-dialog request to be sent?
[JRE] Well, nothing I guess, except that isn't SIP. What would the next hop=
 do if it receives a TCP connection without any SIP message for a while, or=
 receives an empty UDP packet?

>=20
> And, unless the UAC establishes the connnection, it won't=20
> even be able to receive inbound requests in case there is a NAT.
[JRE] True, and in such cases I would expect the first proxy to record-rout=
e, so no issue. The point is, there are situations that the present draft d=
oesn't deal with. I am happy to be convinced that you don't need keep-alive=
 in cases that aren't covered by the draft, but we haven't had that discuss=
ion.


>=20
>=20
> ---------
>=20
>=20
> >>>4. In 4.3.1:
> >>>"When a UAS receives an initial request for a dialog, if=20
> >>>the top-most Record-Route header filed contains a "keep"=20
> parameter, and=20
> >>>if the UAS is willing to send keep-alives associated with=20
> >>>the dialog, it MUST insert a "keep" parameter in its=20
> Contact header field in the
> >>>response(s) to the request.  After that the UAS MUST start to send
> >>>keep-alives towards the edge proxy, and it MUST send=20
> keep-alives for
> >>>the remaining duration of the registration."
> >>>I have a similar problem with this.
> >>=20
> >>Again, the UAS will send the keep-alives towards the address=20
> >>indicated in the Record-Route.
> >[JRE] Same problem.
>=20
> Same answer :)
>=20
>=20
> ---------
>=20
>=20
> >>>5. In 5.3.1:
> >>>"When a proxy receives a dialog initiation request from=20
> >>>its previous-hop proxy, and the top-most Record-Route=20
> header field of=20
> >>>the request contains a "keep" parameter, if the proxy is=20
> willing to send and
> >>>receive keep-alives from the previous-hop proxy for the associated
> >>>dialog, the proxy MUST insert a "keep" parameter in its=20
> >>>Record-Route header field of the associated response(s)=20
> sent towards=20
> >>>the previous-hop proxy.  After that the proxy MUST send=20
> keep-alives, and accept
> >>>keep-alives sent from the previous-hop proxy, for the=20
> duration of the
> >>>dialog."
> >>>I have a similar problem with this. If the top-most=20
> >>>Record-Route header field does not represent the previous hop=20
> >>>proxy, there is no transport connection established to allow=20
> >>>you to start sending keep-alives at this time.
> >>=20
> >>As I said in the beginning of the reply, "next hop" and=20
> >>"previous hop" are probably confusing terminology - at least=20
> >>without proper explanation.
> >[JRE] OK, but this still doesn't address my concern about the=20
> >absence of a transport connection.
>=20
> See above.
>=20
>=20
> --------
>=20
>=20
> >>>6. "When a proxy forwards a dialog initiation request towards=20
> >>>its next- hop proxy, and it wants to negotiate the usage of=20
> >>>keep-alives with
> >>>that next-hop proxy, it MUST include a "keep" parameter in its
> >>>Record-Record header field of the request."
> >>>How does the proxy know that the next hop is a proxy, as=20
> >>>opposed to a UA?=20
> >>=20
> >>I guess that would be a edge-to-UA case. But, I guess that=20
> >>could be clarified.
> >[JRE] As a proxy, how do I know I am the "edge", i.e., how do=20
> >I know the next entity is a UA? Even in the IMS P-CSCF case,=20
> >I don't know - there could be a PBX behind it.
>=20
> When the proxy get the response, if there are no=20
> Record-Routes between the proxy and the UA, the proxy is=20
> adjacent to the UA.
[JRE] This still doesn't make sense. The text talks about "When a proxy for=
wards a dialog initiation request" - at the time of forwarding you just don=
't know what the next entity is - not until the response arrives.

John



>=20
>=20
> ---------
>=20
> >>>Assuming it does know that the next hop is a proxy, how does=20
> >>>it know that the next hop proxy will record-route?
> >>=20
> >>See previous comments on "next hop" and "previous hop" terminology.
> >[JRE] The words "When a proxy forwards a dialog initiation=20
> >request towards its next-hop proxy" do not make sense - a=20
> >proxy simply forwards a request to the next hop entity. Until=20
> >it receives a response, it has no idea whether the entity it=20
> >forwarded the request to is a UA, a non-record-routing proxy=20
> >or a record-routing proxy, i.e., it has no idea whether it=20
> >has sent it to something that complies with your definition=20
> >of "next-hop proxy".
>=20
> I agree. So, the text could probably be more generic, and the=20
> actions then depend on what comes in the answer (Record-Route=20
> or Contact).
>=20
>=20
> Regards,
>=20
> Christer
> =

From pkyzivat@cisco.com  Thu Apr 15 06:42:06 2010
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Cc: sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul E. Jones wrote:
> Paul, Brett,
> 
> I looked at the OPTIONS ping draft and 3261 again.  Perhaps I'm missing it
> and need to be reminded (and I note it's 2AM here), but what in the OPTIONS
> draft changes anything stated in 3261.  The statement of concern is this,
> right?
> 
>    To avoid unduly taxing a receiving SIP entity, transmitters of
>    OPTIONS messages MUST honor the Retry-After header field if
>    received.
> 
> When a 503 is used is in line with 3261: we're not suggesting new
> opportunities to send one. We're just saying that code should be used vs.
> other codes that don't have the same meaning.

> And, Retry-After should be honored.  We say MUST, which I understand is a
> stronger word than SHOULD (and why I'm willing to soften it).  Changing it
> to SHOULD would be in stronger agreement with 21.5.4 of 3261.
> 
> A device would not send 503 with Retry-After lightly, nor are we proposing a
> change as to when a 503 is sent.  So, is using "SHOULD" acceptable? Or is
> this really a problem?  I understand the problems one can get into with the
> current wording in 3261, but what I'm failing to see, I guess, is how the
> OPTIONS ping draft makes this worse.

Let me see if I fully understand your intent here.
If you said nothing about this, and the result of an OPTIONS ping was a 
503 with a Retry-After, then the recipient would be bound by 3261, which 
says:

    A client ... receiving a 503 ... SHOULD NOT
    forward any other requests to that server for the duration specified
    in the Retry-After header field, if present.

Is the intent of the MUST in your draft to strengthen that SHOULD NOT in 
3261 to MUST NOT for those devices that claim conformance to your draft?

If that is the case, then I think it is ok as long as this draft itself 
becomes normative, or a BCP. (I have to check the rules for BCPs - I'm 
not certain if they can have normative statements of their own.) I'm 
pretty sure this would be inappropriate in an informative draft, though 
some non-normative language with the same intent should be ok.

	Thanks,
	Paul



> Paul
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Monday, April 12, 2010 8:25 PM
>> To: Brett Tate
>> Cc: Hadriel Kaplan; Paul E. Jones; sipcore@ietf.org
>> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
>> 00.txt
>>
>>
>>
>> Brett Tate wrote:
>>>>> I don't think that lets you off the hook.
>>>>> We've learned from experience that we were too vague in
>>>>> the past.  So I would expect you to be less vague here -
>>>>> being more explicit about what conditions justify violating
>>>>> the SHOULD.
>>>> And really using "MUST..., unless..." instead.
>>> Is making 503 with Retry-After usable (i.e. so it does more good than
>> harm) something that should be done within this draft and in this
>> working group?  Or is that something that should be addressed within
>> the hopefully dispatched overload working group?
>>
>>
>> Well, now you ask the hard questions.
>>
>> With my chair hat on, IMO any changes to the behavior of 503 will at
>> least need to be blessed by sipcore. (But I'm new as a sipcore chair,
>> so
>> I'm willing to learn from the other chair or the AD that is isn't so.)
>>
>> After that, I think it depends on what is to happen with this draft.
>>
>> If it is intended to be informational, then it can't make any normative
>> changes. It can make any recommendations it wants as long as they don't
>> require violating normative behavior.
>>
>> If its aspirations are higher, and maybe even if they aren't, then I
>> think it makes sense to coordinate it with the work of the overload wg.
>> (AFAIK we don't yet *have* an overload WG, but there is a mailing list,
>> and I gather its expected that we will have the WG before long.)
>>
>> With my individual hat on I will recommend that it go there for some
>> discussion of if/how it relates.
>>
>> 	Thanks,
>> 	Paul
>>
> 
> 
> 

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Subject: Re: [sipcore] draft-jones-sip-options-ping-00: comments
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Paul E. Jones wrote:
>>> This is a problem general for SIP, since there is no way for a SIP
>> entity to
>>> know that a DNS query that returns 4 replies relates to the same or
>>> different servers.  I can't fix that problem in this draft :-)
>> No, but in the 'normal' case, an INVITE is not directed to a RURI that
>> contains an IP address and port number, it's directed to a RURI with a
>> logical name for the 'service', and in that case, it's very logical for
>> Retry-After returned in the 503 to apply to the entire 'service'.
> 
> Messages are still transmitted to a certain server.  RFC 3261 provides
> guidance as to when an entity should stop transmitting messages to a given
> server. The OPTIONS ping draft does not change that logic.

The 3261 language could be clearer on this subject.
In reality, messages are transmitted toward an address (typically an IP 
address), and arrive (if at all) at some server listening on that 
address. As you pointed out already, when the sender has a list of 
alternate addresses that it could have sent the request to, it has no 
way of knowing which ones are served by the same server.

So, in general I think the best the client can do is refrain from 
sending to the same *address*.

But maybe somebody else has better insight on this.

	Thanks,
	Paul

From christer.holmberg@ericsson.com  Thu Apr 15 09:03:13 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 15 Apr 2010 18:03:01 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-02
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-keep-02
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Hi,

>>>Why is this draft-ietf? I don't recall adopting anything of this nature =
as a WG item.
>>>
>>>The draft has been adopted a long time ago. It has just been on hold
>>>for a while, partly because of the INFO work.
>>[JRE] I know, but this draft looks very different from the one adopted as=
 WG item,
>
> Please compare the first and last version of the Info-events draft :)
>
> Seriously, I agree there were editorial changes, but the only
> technical change was to not use the "=3Dyes" parameter. And,
> some of the editorial changes are also based on that change.
>[JRE] This is not true. -00 and -01 both used Via, now we use Path, Record=
-Route and Contact.=20

True. What I meant was that we in Stockholm agreed to use P/RR/C instead of=
 Via, with the "=3Dyes" parameter value, but it was never documented in a d=
raft. In addition, at one point I indicated that there are problems with "=
=3Dyes", and that something else is needed. That else is now presented in -=
02.

----------

>That is a completely different solution. I agree that -02 might be moving =
in the right direction.

Good.

>>>>Before answering to your specific comments, I want to clarify one thing=
:
>>>>
>>>>When I talk about "next hop" and "previous hop", I refer to
>>>>the adjacent node in the established registration/dialog path
>>>>- not a hop which will not be part of that path. That needs
>>>>to be clarfied.
>>>>
>>>>And, this is the reason, based on a comment from yourself,
>>>>that we decided to NOT define the "keep" parameter for the
>>>>Via header field, since not all entities that insert Via will
>>>>be part of the path.
>>>[JRE] Yes, I understand the problems with the old Via-based
>>>proposal, but on reading the present draft it didn't seem to
>>>help much.
>>
>>Keep in mind that in Stockholm we decided to use Contact/Path/Record-Rout=
e instead of Via, so that is an existing agreement.
>[JRE] Well, I could not find any mailing list endorsement of this followin=
g Stockholm, and that is nearly one year ago, so my memory failed me.=20
>The minutes just state: "Option 2 from slides: "Nodes which are adjacent i=
n a route set/registration path"", and don't say anything about the header=
=20
>fields. Unfortunately the link to the slides is broken.

I can look for them, if you want.

----------

>Anyway, we should focus on fixing the present draft.

Yes.

----------

>>>>>What I do have a problem with is the UA-to-UA case (in
>>>>>particular, B2BUA-to-B2BUA), where, if this mechanism is to
>>>>>be used, clearly one of the UAs has to
>>>>>receive keep-alives.
>>>>
>>>>The initial scope of the draft was to use the Outbound
>>>>topology, and assume there is an edge proxy. After that
>>>>people asked to also cover the proxy-to-proxy case.
>>>>
>>>>However, if we want to cover the direct UA-to-UA case I guess
>>>>we could say that the UAs check for a Record-Route with
>>>>"keep", and if there is none then check for a Contact with "keep".
>>>[JRE] Many people use B2BUAs rather than proxies, and I would
>>>expect the draft to be equally applicable between B2BUAs.
>>>Also I would have expected the draft to be applicable between
>>>a proxy/B2BUA and a non-registering UA, or between a
>>>proxy/B2BUA and a directly registering UA (without an edge
>>>proxy). In other words, filling all the gaps that aren't
>>>covered by sip-outbound.
>>
>>Since a B2BUA acts as a UA, I don't we need to separate them.
>>If we solve UA-to-UA we also solve UA-to-B2BUA.
>[JRE] Fine, but the present draft doesn't seem to work for UA-to-UA.

I agree, and the reason for that is I assumed it was not within the scope o=
f the solution.

But, as I said earlier, there should be no problem to make it work for UA-t=
o-UA. Currently an UA only checks if the adjacent node supports keep-alive =
by looking at Record-Route, but we can add text saying that, if there is no=
 Record-Route, the UA checks whether the Contact contains "keep".

However, the use-cases described in the beginning of the draft are the ones=
 I used to justfiy the work, and UA-to-UA is not covered in any of those. S=
o, I just want to be sure people are ok with adding it.

----------

>>>>>2. Also section 4.1 starts off by saying that this is all
>>>>>about keep-alive between a UA and its edge proxy, which tends
>>>>>to rule out the cases above, and also cases where a
>>>>>registering UA registers directly with its registrar, not via
>>>>>an edge proxy. Also it seems to rule out cases where the
>>>>>proxy does not record-route, so that for dialog keep-alive
>>>>>the UA would need to send keep-alives to some other entity.
>>>>
>>>>See my comment about the scope of the draft above.
>>>>
>>>>However, as for the UA-to-UA case, I don't have a problem if
>>>>we want to also cover the direct UA-to-Registrar case. I
>>>>guess the question then is how the registrar will indicate
>>>>its support of keep-alives, since it normally doesn't insert
>>>>a Path header field.
>>>[JRE] Well, the previous proposal used the Via header field,
>>>so in moving away from Via we seem to have lost a capability.
>>>Perhaps a UA wanting to use this mechanism must insert a Path
>>>header field - I haven't thought through the consequences, however.
>>
>>The UA inserts Contact so that is not a problem. The question
>>is how the registrar would indicate support.
>[JRE] If the registrar receives Path in the request, it could send Path in=
 the response.

Yes. It would insert a Path pointing to itself, and add a "keep" parameter.

----------

>>>>3. In 4.2.2:
>>>>"When the UAC receives the response(s) to the initial
>>>>request for the dialog, if the top-most Record-Route
>>>>header field (edge proxy) of
>>>>response(s) contains a "keep" parameter, the UAC MUST start to send
>>>>keep-alives towards the edge proxy for the remaining
>>>>duration of the dialog."
>>>>This doesn't make sense. The response containing the "keep"
>>>>parameter will be received on the path of the  request, but
>>>>depending which entities have record-routed, the dialog path
>>>>may be different. A transport connection to the first entity
>>>>on the dialog path is not established until the next outbound
>>>>or inbound request on that dialog, so "MUST start to send
>>>>keep-alives towards the edge proxy" is unachievable in such cases.
>>>
>>>The UAC will send the keep-alives towards the address
>>>indicated in the Record-Route - NOT to where it sent the
>>>INVITE request.
>>[JRE] How can it sent keep-alives when it hasn't established
>>a transport connection yet? It must wait for the transport
>>connection to be established, which means waiting for an
>>inbound or outbound mid-dialog request. If it's an inbound
>>request, however, which entity should be responsible for keep-alives?
>
>What prevents the UAC from establishing the transport
>connection without waiting for any mid-dialog request to be sent?
>[JRE] Well, nothing I guess, except that isn't SIP. What would the next ho=
p do if it receives a TCP connection without any SIP message for a while, o=
r receives an empty UDP packet?

The UA establishes the connection when the first keep-alive is sent, and si=
nce the next hop has indicated support of keep-alive it needs to be ready t=
o handle it.

I guess we could add a Note indicating that.

----------

>>And, unless the UAC establishes the connnection, it won't
>>even be able to receive inbound requests in case there is a NAT.
>[JRE] True, and in such cases I would expect the first proxy to record-rou=
te, so no issue. The point is, there are situations that the present draft =
doesn't deal with.=20
>I am happy to be convinced that you don't need keep-alive in cases that ar=
en't covered by the draft, but we haven't had that discussion.

I am NOT trying to convince you about that :) =20

I wrote the draft based on the use-cases we had agreed upon (at least accor=
ding to my understanding), but I would be happy to cover additional use-cas=
es, if people are ok with it.

----------

>>>>6. "When a proxy forwards a dialog initiation request towards
>>>>its next- hop proxy, and it wants to negotiate the usage of
>>>>keep-alives with
>>>>that next-hop proxy, it MUST include a "keep" parameter in its
>>>>Record-Record header field of the request."
>>>>How does the proxy know that the next hop is a proxy, as
>>>>opposed to a UA?
>>>
>>>I guess that would be a edge-to-UA case. But, I guess that
>>>could be clarified.
>>[JRE] As a proxy, how do I know I am the "edge", i.e., how do
>>I know the next entity is a UA? Even in the IMS P-CSCF case,
>>I don't know - there could be a PBX behind it.
>
>When the proxy get the response, if there are no
>Record-Routes between the proxy and the UA, the proxy is
>adjacent to the UA.
>[JRE] This still doesn't make sense. The text talks about "When a proxy fo=
rwards a dialog initiation request" - at the time of forwarding you just do=
n't know what the next entity is - not until the response arrives.

That is what I tried to say :)=20

The text about the request being forward would NOT make assumption about wh=
at type of node the next hope is - it would be detected when the response a=
rrives.

Regards,

Christer=

From christer.holmberg@ericsson.com  Thu Apr 15 09:04:43 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Thu, 15 Apr 2010 18:03:49 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

>I'm ok with the SHOULD since that is what is in 3261. What I would
>prefer to see is some enumeration of the cases when its appropriate to
>violate the SHOULD.

If it is a reference to 3261 it would be good to say:

"As specified in RFC 3261, blah blah blah..."

Regards,

Christer




Paul E. Jones wrote:
>> I think that's OK here, since what Brett was suggesting was that it
>> is
>>> already a SHOULD in effect in 3261.  In any case, the worst thing
>> that
>>> happens here is that a device ignores the Retry-After header and
>> claims
>>> compliance.  Apparently, some feel there are plenty of good reasons
>> to do
>>> so.
>> I don't think that lets you off the hook.
>> We've learned from experience that we were too vague in the past.
>> So I would expect you to be less vague here - being more explicit about
>> what conditions justify violating the SHOULD.
>
> Shall I put it back to MUST?  Keep in mind, we're not trying to re-define
> RFC 3261.
>
> Paul
>
>
>
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 15 Apr 2010 18:53:36 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-02
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Christer,

The use cases in 1.1 to 1.4 seem to cover most of what I mentioned in my co=
mments. For example, they make no assumption about the presence of an edge =
proxy between a UA and its registrar/proxy, and they make no assumption abo=
ut which entities, if any, between two UAs, perform record-route. Therefore=
 I would expect such situations to be covered by the solution.

What is missing from the use cases is the B2BUA case, I guess. I think if w=
e include the proxy-to-proxy case (1.4), we should also include the B2BUA-B=
2BUA case, since B2BUAs are commonly used instead of proxies.

(Rest of thread stripped).

John


From christer.holmberg@ericsson.com  Thu Apr 15 10:07:48 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 15 Apr 2010 19:04:42 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-02
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Hi,

>The use cases in 1.1 to 1.4 seem to cover most of what I mentioned in my c=
omments. For example, they=20
>make no assumption about the presence of an edge proxy between a UA and it=
s registrar/proxy, and they=20
>make no assumption about which entities, if any, between two UAs, perform =
record-route.

Well, there is text talking about the fact why Outbound can't be used in th=
ose cases, so the assumption is that the Outbound topology is used.

But, we don't need to argue about it. I am ok to make it more general :)

>What is missing from the use cases is the B2BUA case, I guess. I think if =
we include the proxy-to-proxy case=20
>(1.4), we should also include the B2BUA-B2BUA case, since B2BUAs are commo=
nly used instead of=20
>proxies.

Won't UA-UA case also cover B2BUA-B2BUA?

Regards,

Christer=

From john.elwell@siemens-enterprise.com  Thu Apr 15 12:37:21 2010
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To: Christer Holmberg <christer.holmberg@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 15 Apr 2010 21:37:10 +0200
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 15 April 2010 18:05
> To: Elwell, John; sipcore@ietf.org
> Subject: RE: Draft new version: draft-ietf-sipcore-keep-02
>=20
> Hi,
>=20
> >The use cases in 1.1 to 1.4 seem to cover most of what I=20
> mentioned in my comments. For example, they=20
> >make no assumption about the presence of an edge proxy=20
> between a UA and its registrar/proxy, and they=20
> >make no assumption about which entities, if any, between two=20
> UAs, perform record-route.
>=20
> Well, there is text talking about the fact why Outbound can't=20
> be used in those cases, so the assumption is that the=20
> Outbound topology is used.
[JRE] Outbound does not mandate an edge proxy.

>=20
> But, we don't need to argue about it. I am ok to make it more=20
> general :)
>=20
> >What is missing from the use cases is the B2BUA case, I=20
> guess. I think if we include the proxy-to-proxy case=20
> >(1.4), we should also include the B2BUA-B2BUA case, since=20
> B2BUAs are commonly used instead of=20
> >proxies.
>=20
> Won't UA-UA case also cover B2BUA-B2BUA?
[JRE] But where is the UA-UA case in 1.1 to 1.4? If it's there, it's not ob=
vious.

John

From christer.holmberg@ericsson.com  Thu Apr 15 13:34:48 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 15 Apr 2010 22:30:07 +0200
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-keep-02
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Hi,

>>>The use cases in 1.1 to 1.4 seem to cover most of what I
>>>mentioned in my comments. For example, they
>>>make no assumption about the presence of an edge proxy
>>>between a UA and its registrar/proxy, and they
>>>make no assumption about which entities, if any, between two
>>>UAs, perform record-route.
>>
>>Well, there is text talking about the fact why Outbound can't
>>be used in those cases, so the assumption is that the
>>Outbound topology is used.
>[JRE] Outbound does not mandate an edge proxy.
>
>But, we don't need to argue about it. I am ok to make it more
>general :)
>
>>What is missing from the use cases is the B2BUA case, I
>>guess. I think if we include the proxy-to-proxy case
>>(1.4), we should also include the B2BUA-B2BUA case, since
>>B2BUAs are commonly used instead of
>>proxies.
>
>Won't UA-UA case also cover B2BUA-B2BUA?
>[JRE] But where is the UA-UA case in 1.1 to 1.4? If it's there, it's not o=
bvious.

It's not there. My point was that if we add the UA-UA case, we'll also cove=
r the B2BUA-B2BUA case.

Regards,

Christer=

From dean.willis@softarmor.com  Thu Apr 15 16:42:42 2010
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------- Original message -------
> From: Paul Kyzivat <pkyzivat@cisco.com>
>
> Is the intent of the MUST in your draft to strengthen that SHOULD NOT in 
> 3261 to MUST NOT for those devices that claim conformance to your draft?
>
> If that is the case, then I think it is ok as long as this draft itself 
> becomes normative, or a BCP. (I have to check the rules for BCPs - I'm not 
> certain if they can have normative statements of their own.) I'm pretty 
> sure this would be inappropriate in an informative draft, though some 
> non-normative language with the same intent should be ok.

BCP can use 2119 language, but is for documenting practices for which we 
have in-the-field experience, so it doesn't really apply here.

An informative or experimental draft can use such language, but has to lay 
out the conditions of "what happens if" and explain "why" in a very high 
level of detail. Remember: 2119 is about requirements, and naked 
requirements are weaker than naked emperors. They need to be well-dressed 
in explanation to be respected.

That said, if there is adequate justification to change the language of RFC 
3261, then we should probably just change it, and that means a 
standards-track draft.

--
Dean 

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
References: <01a201cad2a5$ae1096c0$0a31c440$@com>	<430FC6BDED356B4C8498F634416644A91A79E9327E@mail>	<003201cad3a5$6f263540$4d729fc0$@com>	<747A6506A991724FB09B129B79D5FEB61480BBFACB@EXMBXCLUS01.citservers.local>	<002901cad90a$6e6fee10$4b4fca30$@com>	<747A6506A991724FB09B129B79D5FEB61480D10784@EXMBXCLUS01.citservers.local>	<00d701cad9c0$394a0160$abde0420$@com> <4BC3273D.7090902@cisco.com>	<003501cada4a$44b220c0$ce166240$@com> <4BC338F9.80604@cisco.com>	<018a01cadc5c$167a5ba0$436f12e0$@com>, <4BC70BD6.4060601@cisco.com> <FF84A09F50A6DC48ACB6714F4666CC745E21B30AC4@ESESSCMS0354.eemea.ericsson.se>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Capturing everyone's comments, I changed the paragraph as follows:

   If a requesting entity fails to receive a response to an
   OPTIONS message, it MAY retransmit that message following
   the procedures defined in RFC 3261.  If a requesting entity
   receives a 486 or 503 response code, it MAY send a subsequent
   OPTIONS messages in order to detect a change in operational
   status, but it SHOULD, as per RFC 3261, honor the Retry-After
   header field received in the previous response.

Pail
> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Thursday, April 15, 2010 12:04 PM
> To: Paul Kyzivat; Paul E. Jones
> Cc: sipcore@ietf.org
> Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
> 
> Hi,
> 
> >I'm ok with the SHOULD since that is what is in 3261. What I would
> >prefer to see is some enumeration of the cases when its appropriate to
> >violate the SHOULD.
> 
> If it is a reference to 3261 it would be good to say:
> 
> "As specified in RFC 3261, blah blah blah..."
> 
> Regards,
> 
> Christer
> 
> 
> 
> 
> Paul E. Jones wrote:
> >> I think that's OK here, since what Brett was suggesting was that it
> >> is
> >>> already a SHOULD in effect in 3261.  In any case, the worst thing
> >> that
> >>> happens here is that a device ignores the Retry-After header and
> >> claims
> >>> compliance.  Apparently, some feel there are plenty of good reasons
> >> to do
> >>> so.
> >> I don't think that lets you off the hook.
> >> We've learned from experience that we were too vague in the past.
> >> So I would expect you to be less vague here - being more explicit
> about
> >> what conditions justify violating the SHOULD.
> >
> > Shall I put it back to MUST?  Keep in mind, we're not trying to re-
> define
> > RFC 3261.
> >
> > Paul
> >
> >
> >
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore



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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Elwell, John
> Sent: Thursday, April 15, 2010 6:26 AM
>
> [JRE] Yes, I understand the problems with the old Via-based proposal, but
> on reading the present draft it didn't seem to help much.

Can someone explain the problems with the Via approach again?  I'm sorry, I=
 must just be thick.  It seemed like such a clean/simple approach.

-hadriel

From christer.holmberg@ericsson.com  Thu Apr 15 22:37:52 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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Hi,=20

>>[JRE] Yes, I understand the problems with the old Via-based=20
>>proposal,=20
>>but on reading the present draft it didn't seem to help much.
>=20
>Can someone explain the problems with the Via approach again?=20
>I'm sorry, I must just be thick.  It seemed like such a=20
>clean/simple approach.

The main problem is that all entities, including those that will not be pat=
h of the registration/dialog path, insert Via, so when an entity receives a=
 request it does not know which Via header represents the entity that will =
be its "neighbour" in the path, and can't therefor check whether that neigh=
bour supports keep-alives or not.

Example:

P1                      P2                        P3                       =
P4

--- INVITE ------------->
    Via: P1;keep
    Record-Route: P1
                         --- INVITE ------------->
				     Via: P1;keep
                             Via: P2
                             Record-Route: P1
                                                  --- INVITE ------------->=
         =20
                                                      Via: P1;keep
                                                      Via: P2
                                                      Via: P3
                                                      Record-Route: P1



Now, when P4 receives the INVITE, how does it know that P1 will be its dial=
og path neighbour (since P2 and P3 do not record-route)? We cannot assume t=
hat P4 would be able to figure that out by starting to compare the Vias wit=
h the Record-Routes etc...


Regards,

Christer

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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Friday, April 16, 2010 1:38 AM
>=20
> The main problem is that all entities, including those that will not be
> path of the registration/dialog path, insert Via, so when an entity
> receives a request it does not know which Via header represents the entit=
y
> that will be its "neighbour" in the path, and can't therefor check whethe=
r
> that neighbour supports keep-alives or not.


So??  What does it matter?  P4 received a request with a top-most Via that =
does not have the keep param, and it thereby knows the previous hop of that=
 Via's address:port (P3) does not do keep-alive.  Likewise, P1 will get bac=
k a response with the top-most Via (its own) without the keep param's value=
 being "yes", and knows that next-hop address:port (P2) cannot do keepalive=
. =20

That's all that's important at that point. =20

At some later time P1 will send an ACK, and P4 will get it and see the top-=
most Via (now P1) has the keep param, and know that previous hop addr:port =
of P1 can do keep-alive.  P1 won't get a response for the ACK of course (un=
less there's a PRACK) so it won't learn about P4's ability, but P4 can send=
 a stun/crlf keep-alive if it wants to.  And if P1 really cares it can send=
 an Update or re-Invite and find out for itself.

That may sound annoying to have to send a re-invite/update, but it's not th=
e common use-case for this thing anyway, right?

The common use-case would be for a UA registering to its proxy/registrar, w=
here the Via's will be fine.  And the other use-case might be between proxi=
es/b2bua's, where the Via's will be fine too.

I mean basically this whole thing is an optimization at the end of the day =
to avoid doing OPTIONS or REGISTER refreshes, for specific common use-cases=
, so why make it complicated for the common use-cases just to be less annoy=
ing for the not-common use-cases?

-hadriel

From john.elwell@siemens-enterprise.com  Thu Apr 15 23:28:23 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Hadriel Kaplan <HKaplan@acmepacket.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 16 Apr 2010 08:28:13 +0200
Thread-Topic: Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Subject: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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One of the consequences of the changes in the current version of keep is as=
 follows. Taking the example in the diagram Christer drew below, we have P1=
 and P4 on the dialog path, but not P3 and P4. The dialog-forming INVITE tr=
ansaction follows the path through P1, P2, P3 and P4, so none of the transp=
ort connections used for that transaction need to be kept. The only transpo=
rt connection that needs to be kept is the one from P1 to P4. However, this=
 does not exist yet. So what the draft seems to imply is that, when a trans=
port connection directly between P1 and P4 is established, it needs to be k=
ept alive.

In an earlier mail, Christer suggested we establish it straight away, just =
to send the keep-alive, and then, when the first mid-dialog transaction tak=
es place, the connection will be there ready and waiting. This seems wastef=
ul to me. In particular, for some calls the only mid-dialog transaction mig=
ht be the BYE, so establishing a transport connection early and keeping it =
alive, just to send the BYE, seems particularly pointless. Of course, other=
 INVITE-initiated dialogs might have more traffic, e.g., UPDATE, INFO, re-I=
NVITE, REFER, PRACK.

The alternative would be to wait for the first mid-dialog request, and then=
 establish the connection in the normal manner and keep it alive (assuming =
the transaction is not a BYE).

But then, we could equally well use Via in the first mid-dialog transaction=
 to signal keep-alive, and in fact that would probably be simpler.

We seem to have confusion between the dialog-forming transaction and the tr=
ansaction that establishes connections for the dialog path (the first mid-d=
ialog transaction).

Finally, we need to say something about dialog termination - there is no po=
int in keeping the path open permanently, except in the context of connecti=
on reuse.

John

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 16 April 2010 06:38
> To: Hadriel Kaplan; Elwell, John; sipcore@ietf.org
> Subject: RE: Draft new version: draft-ietf-sipcore-keep-02
>=20
>=20
> Hi,=20
>=20
> >>[JRE] Yes, I understand the problems with the old Via-based=20
> >>proposal,=20
> >>but on reading the present draft it didn't seem to help much.
> >=20
> >Can someone explain the problems with the Via approach again?=20
> >I'm sorry, I must just be thick.  It seemed like such a=20
> >clean/simple approach.
>=20
> The main problem is that all entities, including those that=20
> will not be path of the registration/dialog path, insert Via,=20
> so when an entity receives a request it does not know which=20
> Via header represents the entity that will be its "neighbour"=20
> in the path, and can't therefor check whether that neighbour=20
> supports keep-alives or not.
>=20
> Example:
>=20
> P1                      P2                        P3         =20
>              P4
>=20
> --- INVITE ------------->
>     Via: P1;keep
>     Record-Route: P1
>                          --- INVITE ------------->
> 				     Via: P1;keep
>                              Via: P2
>                              Record-Route: P1
>                                                   --- INVITE -------->   =
      =20
>                                                       Via: P1;keep
>                                                       Via: P2
>                                                       Via: P3
>                                                       Record-Route: P1
>=20
>=20
>=20
> Now, when P4 receives the INVITE, how does it know that P1=20
> will be its dialog path neighbour (since P2 and P3 do not=20
> record-route)? We cannot assume that P4 would be able to=20
> figure that out by starting to compare the Vias with the=20
> Record-Routes etc...
>=20
>=20
> Regards,
>=20
> Christer
> =

From john.elwell@siemens-enterprise.com  Thu Apr 15 23:31:49 2010
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To: Christer Holmberg <christer.holmberg@ericsson.com>, Hadriel Kaplan <HKaplan@acmepacket.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 16 Apr 2010 08:31:36 +0200
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Subject: Re: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Sorry, I forgot to mention ACK in the list of mid-dialog requests below. Th=
at has specific considerations, as Hadriel has just pointed out on the old =
thread.

John=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Elwell, John
> Sent: 16 April 2010 07:28
> To: Christer Holmberg; Hadriel Kaplan; sipcore@ietf.org
> Subject: [sipcore] Keeping alive connections on a dialog path=20
> (was RE: Draft new version: draft-ietf-sipcore-keep-02)
>=20
> One of the consequences of the changes in the current version=20
> of keep is as follows. Taking the example in the diagram=20
> Christer drew below, we have P1 and P4 on the dialog path,=20
> but not P3 and P4. The dialog-forming INVITE transaction=20
> follows the path through P1, P2, P3 and P4, so none of the=20
> transport connections used for that transaction need to be=20
> kept. The only transport connection that needs to be kept is=20
> the one from P1 to P4. However, this does not exist yet. So=20
> what the draft seems to imply is that, when a transport=20
> connection directly between P1 and P4 is established, it=20
> needs to be kept alive.
>=20
> In an earlier mail, Christer suggested we establish it=20
> straight away, just to send the keep-alive, and then, when=20
> the first mid-dialog transaction takes place, the connection=20
> will be there ready and waiting. This seems wasteful to me.=20
> In particular, for some calls the only mid-dialog transaction=20
> might be the BYE, so establishing a transport connection=20
> early and keeping it alive, just to send the BYE, seems=20
> particularly pointless. Of course, other INVITE-initiated=20
> dialogs might have more traffic, e.g., UPDATE, INFO,=20
> re-INVITE, REFER, PRACK.
>=20
> The alternative would be to wait for the first mid-dialog=20
> request, and then establish the connection in the normal=20
> manner and keep it alive (assuming the transaction is not a BYE).
>=20
> But then, we could equally well use Via in the first=20
> mid-dialog transaction to signal keep-alive, and in fact that=20
> would probably be simpler.
>=20
> We seem to have confusion between the dialog-forming=20
> transaction and the transaction that establishes connections=20
> for the dialog path (the first mid-dialog transaction).
>=20
> Finally, we need to say something about dialog termination -=20
> there is no point in keeping the path open permanently,=20
> except in the context of connection reuse.
>=20
> John
>=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> > Sent: 16 April 2010 06:38
> > To: Hadriel Kaplan; Elwell, John; sipcore@ietf.org
> > Subject: RE: Draft new version: draft-ietf-sipcore-keep-02
> >=20
> >=20
> > Hi,=20
> >=20
> > >>[JRE] Yes, I understand the problems with the old Via-based=20
> > >>proposal,=20
> > >>but on reading the present draft it didn't seem to help much.
> > >=20
> > >Can someone explain the problems with the Via approach again?=20
> > >I'm sorry, I must just be thick.  It seemed like such a=20
> > >clean/simple approach.
> >=20
> > The main problem is that all entities, including those that=20
> > will not be path of the registration/dialog path, insert Via,=20
> > so when an entity receives a request it does not know which=20
> > Via header represents the entity that will be its "neighbour"=20
> > in the path, and can't therefor check whether that neighbour=20
> > supports keep-alives or not.
> >=20
> > Example:
> >=20
> > P1                      P2                        P3         =20
> >              P4
> >=20
> > --- INVITE ------------->
> >     Via: P1;keep
> >     Record-Route: P1
> >                          --- INVITE ------------->
> > 				     Via: P1;keep
> >                              Via: P2
> >                              Record-Route: P1
> >                                                   ---=20
> INVITE -------->         =20
> >                                                       Via: P1;keep
> >                                                       Via: P2
> >                                                       Via: P3
> >                                                      =20
> Record-Route: P1
> >=20
> >=20
> >=20
> > Now, when P4 receives the INVITE, how does it know that P1=20
> > will be its dialog path neighbour (since P2 and P3 do not=20
> > record-route)? We cannot assume that P4 would be able to=20
> > figure that out by starting to compare the Vias with the=20
> > Record-Routes etc...
> >=20
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From christer.holmberg@ericsson.com  Fri Apr 16 00:03:06 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, Hadriel Kaplan <HKaplan@acmepacket.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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Subject: Re: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Hi,

Just for my understanding, how would Via be simpler, since it requires the =
insertion of "=3Dyes" by the adjacent entity in order to work?

Regards,

Christer


=20

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
> Sent: 16. huhtikuuta 2010 9:32
> To: Christer Holmberg; Hadriel Kaplan; sipcore@ietf.org
> Subject: RE: [sipcore] Keeping alive connections on a dialog=20
> path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
>=20
> Sorry, I forgot to mention ACK in the list of mid-dialog=20
> requests below. That has specific considerations, as Hadriel=20
> has just pointed out on the old thread.
>=20
> John=20
>=20
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Elwell, John
> > Sent: 16 April 2010 07:28
> > To: Christer Holmberg; Hadriel Kaplan; sipcore@ietf.org
> > Subject: [sipcore] Keeping alive connections on a dialog=20
> path (was RE:=20
> > Draft new version: draft-ietf-sipcore-keep-02)
> >=20
> > One of the consequences of the changes in the current=20
> version of keep=20
> > is as follows. Taking the example in the diagram Christer=20
> drew below,=20
> > we have P1 and P4 on the dialog path, but not P3 and P4. The=20
> > dialog-forming INVITE transaction follows the path through=20
> P1, P2, P3=20
> > and P4, so none of the transport connections used for that=20
> transaction=20
> > need to be kept. The only transport connection that needs=20
> to be kept=20
> > is the one from P1 to P4. However, this does not exist yet. So what=20
> > the draft seems to imply is that, when a transport=20
> connection directly=20
> > between P1 and P4 is established, it needs to be kept alive.
> >=20
> > In an earlier mail, Christer suggested we establish it=20
> straight away,=20
> > just to send the keep-alive, and then, when the first mid-dialog=20
> > transaction takes place, the connection will be there ready and=20
> > waiting. This seems wasteful to me.
> > In particular, for some calls the only mid-dialog=20
> transaction might be=20
> > the BYE, so establishing a transport connection early and=20
> keeping it=20
> > alive, just to send the BYE, seems particularly pointless.=20
> Of course,=20
> > other INVITE-initiated dialogs might have more traffic,=20
> e.g., UPDATE,=20
> > INFO, re-INVITE, REFER, PRACK.
> >=20
> > The alternative would be to wait for the first mid-dialog=20
> request, and=20
> > then establish the connection in the normal manner and keep=20
> it alive=20
> > (assuming the transaction is not a BYE).
> >=20
> > But then, we could equally well use Via in the first mid-dialog=20
> > transaction to signal keep-alive, and in fact that would=20
> probably be=20
> > simpler.
> >=20
> > We seem to have confusion between the dialog-forming=20
> transaction and=20
> > the transaction that establishes connections for the dialog=20
> path (the=20
> > first mid-dialog transaction).
> >=20
> > Finally, we need to say something about dialog termination=20
> - there is=20
> > no point in keeping the path open permanently, except in=20
> the context=20
> > of connection reuse.
> >=20
> > John
> >=20
> > > -----Original Message-----
> > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > Sent: 16 April 2010 06:38
> > > To: Hadriel Kaplan; Elwell, John; sipcore@ietf.org
> > > Subject: RE: Draft new version: draft-ietf-sipcore-keep-02
> > >=20
> > >=20
> > > Hi,
> > >=20
> > > >>[JRE] Yes, I understand the problems with the old Via-based=20
> > > >>proposal, but on reading the present draft it didn't=20
> seem to help=20
> > > >>much.
> > > >=20
> > > >Can someone explain the problems with the Via approach again?=20
> > > >I'm sorry, I must just be thick.  It seemed like such a=20
> > > >clean/simple approach.
> > >=20
> > > The main problem is that all entities, including those=20
> that will not=20
> > > be path of the registration/dialog path, insert Via, so when an=20
> > > entity receives a request it does not know which Via header=20
> > > represents the entity that will be its "neighbour"
> > > in the path, and can't therefor check whether that neighbour=20
> > > supports keep-alives or not.
> > >=20
> > > Example:
> > >=20
> > > P1                      P2                        P3         =20
> > >              P4
> > >=20
> > > --- INVITE ------------->
> > >     Via: P1;keep
> > >     Record-Route: P1
> > >                          --- INVITE ------------->
> > > 				     Via: P1;keep
> > >                              Via: P2
> > >                              Record-Route: P1
> > >                                                   ---
> > INVITE -------->         =20
> > >                                                       Via: P1;keep
> > >                                                       Via: P2
> > >                                                       Via: P3
> > >                                                      =20
> > Record-Route: P1
> > >=20
> > >=20
> > >=20
> > > Now, when P4 receives the INVITE, how does it know that=20
> P1 will be=20
> > > its dialog path neighbour (since P2 and P3 do not=20
> record-route)? We=20
> > > cannot assume that P4 would be able to figure that out by=20
> starting=20
> > > to compare the Vias with the Record-Routes etc...
> > >=20
> > >=20
> > > Regards,
> > >=20
> > > Christer
> > >=20
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> > =

From HKaplan@acmepacket.com  Fri Apr 16 00:15:59 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 16 Apr 2010 03:15:46 -0400
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Friday, April 16, 2010 3:03 AM
> To: Elwell, John; Hadriel Kaplan; sipcore@ietf.org
>=20
> Hi,
>=20
> Just for my understanding, how would Via be simpler, since it requires th=
e
> insertion of "=3Dyes" by the adjacent entity in order to work?

Compared to using several different header types, each used in different ca=
ses, and having a bunch of if/else checks and such?  How *isn't* it simpler=
?=20

Setting a received top-most Via param isn't hard is it?  Like the Via "rece=
ived" param. =20

In fact, instead of doing the new Flow-Timer header, just make the "keep" p=
aram's value the timer.  So instead of setting it to "yes", it sets it to t=
he number of seconds.

-hadriel

From john.elwell@siemens-enterprise.com  Fri Apr 16 00:35:58 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, Christer Holmberg <christer.holmberg@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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In addition to Hadriel's point, mid-dialog requests do not necessarily cont=
ain Record-Route and Contact, but always contain Via. So if we rely on Reco=
rd-Route and Contact, the decision on keeping connections alive for followi=
ng the first mid-dialog request would have to be based on retained state fr=
om the initial request. Does a record-routing proxy necessarily store Recor=
d-route and Contact header fields from the dialog-forming transaction?

John

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> Sent: 16 April 2010 08:16
> To: Christer Holmberg; Elwell, John; sipcore@ietf.org
> Subject: RE: [sipcore] Keeping alive connections on a dialog=20
> path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
>=20
>=20
>=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: Friday, April 16, 2010 3:03 AM
> > To: Elwell, John; Hadriel Kaplan; sipcore@ietf.org
> >=20
> > Hi,
> >=20
> > Just for my understanding, how would Via be simpler, since=20
> it requires the
> > insertion of "=3Dyes" by the adjacent entity in order to work?
>=20
> Compared to using several different header types, each used=20
> in different cases, and having a bunch of if/else checks and=20
> such?  How *isn't* it simpler?=20
>=20
> Setting a received top-most Via param isn't hard is it?  Like=20
> the Via "received" param. =20
>=20
> In fact, instead of doing the new Flow-Timer header, just=20
> make the "keep" param's value the timer.  So instead of=20
> setting it to "yes", it sets it to the number of seconds.
>=20
> -hadriel
> =

From christer.holmberg@ericsson.com  Fri Apr 16 00:37:45 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 16 Apr 2010 09:32:57 +0200
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Subject: Re: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Hi,

So, let's look at an example, to make sure I understand you.

An route set has been established, and ACK is sent:



UAC ----------- P1 ----------- UAS

---ACK --------->
---Via: UAC;keep
                ---ACK --------->
                ---Via:P1;keep
                ---Via:UAC;keep=3Dyes

Since there is no reply to ACK, how does the UAS tell P1 that it supports k=
eep-alives?

To make it work, wouldn't we need to require UPDATE or PRACK, so that UAS c=
an send back a response with "Via:P1;keep=3Dyes"?

Regards,

Christer





> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> Sent: 16. huhtikuuta 2010 10:16
> To: Christer Holmberg; Elwell, John; sipcore@ietf.org
> Subject: RE: [sipcore] Keeping alive connections on a dialog=20
> path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
>=20
>=20
>=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: Friday, April 16, 2010 3:03 AM
> > To: Elwell, John; Hadriel Kaplan; sipcore@ietf.org
> >=20
> > Hi,
> >=20
> > Just for my understanding, how would Via be simpler, since=20
> it requires=20
> > the insertion of "=3Dyes" by the adjacent entity in order to work?
>=20
> Compared to using several different header types, each used=20
> in different cases, and having a bunch of if/else checks and=20
> such?  How *isn't* it simpler?=20
>=20
> Setting a received top-most Via param isn't hard is it?  Like=20
> the Via "received" param. =20
>=20
> In fact, instead of doing the new Flow-Timer header, just=20
> make the "keep" param's value the timer.  So instead of=20
> setting it to "yes", it sets it to the number of seconds.
>=20
> -hadriel
> =

From christer.holmberg@ericsson.com  Fri Apr 16 00:39:43 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, Hadriel Kaplan <HKaplan@acmepacket.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 16 Apr 2010 09:35:46 +0200
Thread-Topic: [sipcore] Keeping alive connections on a dialog path
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Subject: Re: [sipcore] Keeping alive connections on a dialog path
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Hi,=20

>In addition to Hadriel's point, mid-dialog requests do not=20
>necessarily contain Record-Route and Contact, but always=20
>contain Via. So if we rely on Record-Route and Contact, the=20
>decision on keeping connections alive for following the first=20
>mid-dialog request would have to be based on retained state=20
>from the initial request. Does a record-routing proxy=20
>necessarily store Record-route and Contact header fields from=20
>the dialog-forming transaction?

I agree that it might not be possible to rely on R-R and Contact in a mid-d=
ialog request. But, the current draft does not require a mid-dialog request=
.

Regards,

Christer



> > -----Original Message-----
> > From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> > Sent: 16 April 2010 08:16
> > To: Christer Holmberg; Elwell, John; sipcore@ietf.org
> > Subject: RE: [sipcore] Keeping alive connections on a=20
> dialog path (was=20
> > RE: Draft new version: draft-ietf-sipcore-keep-02)
> >=20
> >=20
> >=20
> > > -----Original Message-----
> > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > Sent: Friday, April 16, 2010 3:03 AM
> > > To: Elwell, John; Hadriel Kaplan; sipcore@ietf.org
> > >=20
> > > Hi,
> > >=20
> > > Just for my understanding, how would Via be simpler, since
> > it requires the
> > > insertion of "=3Dyes" by the adjacent entity in order to work?
> >=20
> > Compared to using several different header types, each used in=20
> > different cases, and having a bunch of if/else checks and=20
> such?  How=20
> > *isn't* it simpler?
> >=20
> > Setting a received top-most Via param isn't hard is it? =20
> Like the Via=20
> > "received" param.
> >=20
> > In fact, instead of doing the new Flow-Timer header, just make the=20
> > "keep" param's value the timer.  So instead of setting it=20
> to "yes", it=20
> > sets it to the number of seconds.
> >=20
> > -hadriel
> > =

From john.elwell@siemens-enterprise.com  Fri Apr 16 01:04:43 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Hadriel Kaplan <HKaplan@acmepacket.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 16 Apr 2010 10:03:16 +0200
Thread-Topic: [sipcore] Keeping alive connections on a dialog path
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Subject: Re: [sipcore] Keeping alive connections on a dialog path
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Christer,

But the current draft requires retention of state from the dialog initiatio=
n request to be used when the first mid-dialog request occurs or, as I thin=
k you are suggesting, we establish the connection for the mid-dialog path i=
mmediately, in anticipation of the first mid-dialog request.

Example 1: INVITE.
The first mid-dialog request is normally ACK, so following the dialog-formi=
ng response to INVITE (e.g., 180), a record-routing proxy establishes a con=
nection to the next record-routing proxy and keeps it alive. When the ACK a=
rrives, the first record-routing proxy knows it can send it over that trans=
port connection. I think this is what you are suggesting.

But what if the first mid-dialog request is UPDATE, sent in the backwards d=
irection (legal without having used PRACK, as long as UPDATE doesn't contai=
n an SDP offer)?

Example 2: SUBSCRIBE or REFER.
The first mid-dialog request is NOTIFY. If the first record-routing proxy e=
stablishes a connection to the second record-routing proxy at the time of t=
he SUBSCRIBE response, how does the second record-routing proxy know that t=
his can be used for the NOTIFY request?

Also what if the NOTIFY request reaches the first record-routing proxy befo=
re the SUBSCRIBE response?

John
=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 16 April 2010 08:36
> To: Elwell, John; Hadriel Kaplan; sipcore@ietf.org
> Subject: RE: [sipcore] Keeping alive connections on a dialog path
>=20
>=20
> Hi,=20
>=20
> >In addition to Hadriel's point, mid-dialog requests do not=20
> >necessarily contain Record-Route and Contact, but always=20
> >contain Via. So if we rely on Record-Route and Contact, the=20
> >decision on keeping connections alive for following the first=20
> >mid-dialog request would have to be based on retained state=20
> >from the initial request. Does a record-routing proxy=20
> >necessarily store Record-route and Contact header fields from=20
> >the dialog-forming transaction?
>=20
> I agree that it might not be possible to rely on R-R and=20
> Contact in a mid-dialog request. But, the current draft does=20
> not require a mid-dialog request.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
> > > -----Original Message-----
> > > From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> > > Sent: 16 April 2010 08:16
> > > To: Christer Holmberg; Elwell, John; sipcore@ietf.org
> > > Subject: RE: [sipcore] Keeping alive connections on a=20
> > dialog path (was=20
> > > RE: Draft new version: draft-ietf-sipcore-keep-02)
> > >=20
> > >=20
> > >=20
> > > > -----Original Message-----
> > > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > > Sent: Friday, April 16, 2010 3:03 AM
> > > > To: Elwell, John; Hadriel Kaplan; sipcore@ietf.org
> > > >=20
> > > > Hi,
> > > >=20
> > > > Just for my understanding, how would Via be simpler, since
> > > it requires the
> > > > insertion of "=3Dyes" by the adjacent entity in order to work?
> > >=20
> > > Compared to using several different header types, each used in=20
> > > different cases, and having a bunch of if/else checks and=20
> > such?  How=20
> > > *isn't* it simpler?
> > >=20
> > > Setting a received top-most Via param isn't hard is it? =20
> > Like the Via=20
> > > "received" param.
> > >=20
> > > In fact, instead of doing the new Flow-Timer header, just=20
> make the=20
> > > "keep" param's value the timer.  So instead of setting it=20
> > to "yes", it=20
> > > sets it to the number of seconds.
> > >=20
> > > -hadriel
> > > =

From keith.drage@alcatel-lucent.com  Fri Apr 16 04:42:50 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "Paul E. Jones" <paulej@packetizer.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
Date: Fri, 16 Apr 2010 13:37:24 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Being pedantic, I would note that the second "MAY" (as in "it MAY send a su=
bsequent OPTIONS messages") can apparently be replaced with "can" as all th=
is sentence is doing is the possibility of repeating the option expressed b=
y the "MAY" in the first sentence. It is not expressing a new requirement.

regards

Keith

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul E. Jones
> Sent: Friday, April 16, 2010 1:34 AM
> To: 'Christer Holmberg'; 'Paul Kyzivat'
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] FW: I-D=20
> Action:draft-jones-sip-options-ping-00.txt
>=20
> Capturing everyone's comments, I changed the paragraph as follows:
>=20
>    If a requesting entity fails to receive a response to an
>    OPTIONS message, it MAY retransmit that message following
>    the procedures defined in RFC 3261.  If a requesting entity
>    receives a 486 or 503 response code, it MAY send a subsequent
>    OPTIONS messages in order to detect a change in operational
>    status, but it SHOULD, as per RFC 3261, honor the Retry-After
>    header field received in the previous response.
>=20
> Pail
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: Thursday, April 15, 2010 12:04 PM
> > To: Paul Kyzivat; Paul E. Jones
> > Cc: sipcore@ietf.org
> > Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> > 00.txt
> >=20
> > Hi,
> >=20
> > >I'm ok with the SHOULD since that is what is in 3261. What I would=20
> > >prefer to see is some enumeration of the cases when its=20
> appropriate=20
> > >to violate the SHOULD.
> >=20
> > If it is a reference to 3261 it would be good to say:
> >=20
> > "As specified in RFC 3261, blah blah blah..."
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >=20
> > Paul E. Jones wrote:
> > >> I think that's OK here, since what Brett was suggesting=20
> was that it=20
> > >> is
> > >>> already a SHOULD in effect in 3261.  In any case, the=20
> worst thing
> > >> that
> > >>> happens here is that a device ignores the Retry-After header and
> > >> claims
> > >>> compliance.  Apparently, some feel there are plenty of good=20
> > >>> reasons
> > >> to do
> > >>> so.
> > >> I don't think that lets you off the hook.
> > >> We've learned from experience that we were too vague in the past.
> > >> So I would expect you to be less vague here - being more explicit
> > about
> > >> what conditions justify violating the SHOULD.
> > >
> > > Shall I put it back to MUST?  Keep in mind, we're not=20
> trying to re-
> > define
> > > RFC 3261.
> > >
> > > Paul
> > >
> > >
> > >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

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Cc: paulej@packetizer.com, sipcore@ietf.org, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Dean Willis wrote:
> 
> 
> ------- Original message -------
>> From: Paul Kyzivat <pkyzivat@cisco.com>
>>
>> Is the intent of the MUST in your draft to strengthen that SHOULD NOT 
>> in 3261 to MUST NOT for those devices that claim conformance to your 
>> draft?
>>
>> If that is the case, then I think it is ok as long as this draft 
>> itself becomes normative, or a BCP. (I have to check the rules for 
>> BCPs - I'm not certain if they can have normative statements of their 
>> own.) I'm pretty sure this would be inappropriate in an informative 
>> draft, though some non-normative language with the same intent should 
>> be ok.
> 
> BCP can use 2119 language, but is for documenting practices for which we 
> have in-the-field experience, so it doesn't really apply here.

If I understand Paul Jones's intent, I think that does apply.
I think he is suggesting this behavior just for those following the 
practices he is proposing.

> An informative or experimental draft can use such language, but has to 
> lay out the conditions of "what happens if" and explain "why" in a very 
> high level of detail. Remember: 2119 is about requirements, and naked 
> requirements are weaker than naked emperors. They need to be 
> well-dressed in explanation to be respected.
> 
> That said, if there is adequate justification to change the language of 
> RFC 3261, then we should probably just change it, and that means a 
> standards-track draft.

I don't think we would change the SHOULD NOT to MUST NOT in 3261. But it 
would be helpful to change it to SHOULD NOT, with exception for the 
following cases: ...

But that is a different thing than what Paul J is trying to achieve.

	Thanks,
	Paul

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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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All of this (Paul's revision and Keith's tweak) all work for me.

	Thanks,
	Paul

DRAGE, Keith (Keith) wrote:
> Being pedantic, I would note that the second "MAY" (as in "it MAY send a subsequent OPTIONS messages") can apparently be replaced with "can" as all this sentence is doing is the possibility of repeating the option expressed by the "MAY" in the first sentence. It is not expressing a new requirement.
> 
> regards
> 
> Keith
> 
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org 
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul E. Jones
>> Sent: Friday, April 16, 2010 1:34 AM
>> To: 'Christer Holmberg'; 'Paul Kyzivat'
>> Cc: sipcore@ietf.org
>> Subject: Re: [sipcore] FW: I-D 
>> Action:draft-jones-sip-options-ping-00.txt
>>
>> Capturing everyone's comments, I changed the paragraph as follows:
>>
>>    If a requesting entity fails to receive a response to an
>>    OPTIONS message, it MAY retransmit that message following
>>    the procedures defined in RFC 3261.  If a requesting entity
>>    receives a 486 or 503 response code, it MAY send a subsequent
>>    OPTIONS messages in order to detect a change in operational
>>    status, but it SHOULD, as per RFC 3261, honor the Retry-After
>>    header field received in the previous response.
>>
>> Pail
>>> -----Original Message-----
>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>> Sent: Thursday, April 15, 2010 12:04 PM
>>> To: Paul Kyzivat; Paul E. Jones
>>> Cc: sipcore@ietf.org
>>> Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
>>> 00.txt
>>>
>>> Hi,
>>>
>>>> I'm ok with the SHOULD since that is what is in 3261. What I would 
>>>> prefer to see is some enumeration of the cases when its 
>> appropriate 
>>>> to violate the SHOULD.
>>> If it is a reference to 3261 it would be good to say:
>>>
>>> "As specified in RFC 3261, blah blah blah..."
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>>
>>>
>>>
>>> Paul E. Jones wrote:
>>>>> I think that's OK here, since what Brett was suggesting 
>> was that it 
>>>>> is
>>>>>> already a SHOULD in effect in 3261.  In any case, the 
>> worst thing
>>>>> that
>>>>>> happens here is that a device ignores the Retry-After header and
>>>>> claims
>>>>>> compliance.  Apparently, some feel there are plenty of good 
>>>>>> reasons
>>>>> to do
>>>>>> so.
>>>>> I don't think that lets you off the hook.
>>>>> We've learned from experience that we were too vague in the past.
>>>>> So I would expect you to be less vague here - being more explicit
>>> about
>>>>> what conditions justify violating the SHOULD.
>>>> Shall I put it back to MUST?  Keep in mind, we're not 
>> trying to re-
>>> define
>>>> RFC 3261.
>>>>
>>>> Paul
>>>>
>>>>
>>>>
>>> _______________________________________________
>>> 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
>>

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To: Paul Kyzivat <pkyzivat@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Fri, 16 Apr 2010 11:25:54 -0400
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Thursday, April 15, 2010 9:42 AM
> To: Paul E. Jones
>=20
> Let me see if I fully understand your intent here.
> If you said nothing about this, and the result of an OPTIONS ping was a
> 503 with a Retry-After, then the recipient would be bound by 3261, which
> says:
>=20
>     A client ... receiving a 503 ... SHOULD NOT
>     forward any other requests to that server for the duration specified
>     in the Retry-After header field, if present.
>=20
> Is the intent of the MUST in your draft to strengthen that SHOULD NOT in
> 3261 to MUST NOT for those devices that claim conformance to your draft?
>=20
> If that is the case, then I think it is ok as long as this draft itself
> becomes normative, or a BCP. (I have to check the rules for BCPs - I'm
> not certain if they can have normative statements of their own.) I'm
> pretty sure this would be inappropriate in an informative draft, though
> some non-normative language with the same intent should be ok.

Afaik you can keep the language even in an Informational.  At least 2119 an=
d 2026 don't say otherwise, and there are existing published Informational'=
s which use the word MUST.=20

The real question is if you SHOULD or SHOULD NOT have this Retry-After stat=
ement to begin with. ;)

If your goal is interoperability from the customer's perspective, then I th=
ink explicitly specifying what the Retry-After really does/means *is* impor=
tant.  Unfortunately, if you specify that it means no subsequent requests (=
other than OPTIONS) are sent, some of us will have to submit another Inform=
ational which contradicts this one because for a lot of customers it's the =
wrong behavior.=20

-hadriel

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Subject: Re: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Friday, April 16, 2010 3:33 AM
> To: Hadriel Kaplan; Elwell, John; sipcore@ietf.org
>
> So, let's look at an example, to make sure I understand you.
> An route set has been established, and ACK is sent:
>=20
>=20
>=20
> UAC ----------- P1 ----------- UAS
>=20
> ---ACK --------->
> ---Via: UAC;keep
>                 ---ACK --------->
>                 ---Via:P1;keep
>                 ---Via:UAC;keep=3Dyes
>=20
> Since there is no reply to ACK, how does the UAS tell P1 that it supports
> keep-alives?

As I said in my email: it doesn't.  But that's ok.  SIP isn't broken.


> To make it work, wouldn't we need to require UPDATE or PRACK, so that UAS
> can send back a response with "Via:P1;keep=3Dyes"?

No, we don't need to require anything.  The UAS learned P1 supports keep.  =
If it wants to, it can now send keepalives to P1.  P1 has learned that the =
UAC supports keep.  If it wants to, it can send keepalives to it.

If the UAC really wants to send keepalive for some reason, it can send a UP=
DATE, re-INVITE, or OPTIONS at its leisure, to find out about P1.  If P1 re=
ally wants to send keepalive for some reason, it can act as a UA of its own=
 and send an out-of-dialog OPTIONS.

That's it.  It's not *broken* - SIP still works, the dialog still exists, e=
tc.  It's just not "optimal" for this specific call-flow scenario, but who =
cares?!?  Why double the complexity for the common use-cases, just to make =
it less complex for the rare use-cases??

-hadriel

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'DRAGE, Keith \(Keith\)'" <keith.drage@alcatel-lucent.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
References: <01a201cad2a5$ae1096c0$0a31c440$@com>	<430FC6BDED356B4C8498F634416644A91A79E9327E@mail>	<003201cad3a5$6f263540$4d729fc0$@com>	<747A6506A991724FB09B129B79D5FEB61480BBFACB@EXMBXCLUS01.citservers.local>	<002901cad90a$6e6fee10$4b4fca30$@com>	<747A6506A991724FB09B129B79D5FEB61480D10784@EXMBXCLUS01.citservers.local>	<00d701cad9c0$394a0160$abde0420$@com>	<4BC3273D.7090902@cisco.com>	<003501cada4a$44b220c0$ce166240$@com>	<4BC338F9.80604@cisco.com>	<018a01cadc5c$167a5ba0$436f12e0$@com>, <4BC70BD6.4060601@cisco.com>	<FF84A09F50A6DC48ACB6714F4666CC745E21B30AC4@ESESSCMS0354.eemea.ericsson.se>	<00cd01cadcfc$8f1f8f80$ad5eae80$@com> <EDC0A1AE77C57744B664A310A0B23AE211EF89D1@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com>
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Date: Fri, 16 Apr 2010 18:08:07 -0400
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Ok.. done.

Paul

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of DRAGE, Keith (Keith)
> Sent: Friday, April 16, 2010 7:37 AM
> To: Paul E. Jones; 'Christer Holmberg'; 'Paul Kyzivat'
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
> 
> Being pedantic, I would note that the second "MAY" (as in "it MAY send
> a subsequent OPTIONS messages") can apparently be replaced with "can"
> as all this sentence is doing is the possibility of repeating the
> option expressed by the "MAY" in the first sentence. It is not
> expressing a new requirement.
> 
> regards
> 
> Keith
> 
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul E. Jones
> > Sent: Friday, April 16, 2010 1:34 AM
> > To: 'Christer Holmberg'; 'Paul Kyzivat'
> > Cc: sipcore@ietf.org
> > Subject: Re: [sipcore] FW: I-D
> > Action:draft-jones-sip-options-ping-00.txt
> >
> > Capturing everyone's comments, I changed the paragraph as follows:
> >
> >    If a requesting entity fails to receive a response to an
> >    OPTIONS message, it MAY retransmit that message following
> >    the procedures defined in RFC 3261.  If a requesting entity
> >    receives a 486 or 503 response code, it MAY send a subsequent
> >    OPTIONS messages in order to detect a change in operational
> >    status, but it SHOULD, as per RFC 3261, honor the Retry-After
> >    header field received in the previous response.
> >
> > Pail
> > > -----Original Message-----
> > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > Sent: Thursday, April 15, 2010 12:04 PM
> > > To: Paul Kyzivat; Paul E. Jones
> > > Cc: sipcore@ietf.org
> > > Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> > > 00.txt
> > >
> > > Hi,
> > >
> > > >I'm ok with the SHOULD since that is what is in 3261. What I would
> > > >prefer to see is some enumeration of the cases when its
> > appropriate
> > > >to violate the SHOULD.
> > >
> > > If it is a reference to 3261 it would be good to say:
> > >
> > > "As specified in RFC 3261, blah blah blah..."
> > >
> > > Regards,
> > >
> > > Christer
> > >
> > >
> > >
> > >
> > > Paul E. Jones wrote:
> > > >> I think that's OK here, since what Brett was suggesting
> > was that it
> > > >> is
> > > >>> already a SHOULD in effect in 3261.  In any case, the
> > worst thing
> > > >> that
> > > >>> happens here is that a device ignores the Retry-After header
> and
> > > >> claims
> > > >>> compliance.  Apparently, some feel there are plenty of good
> > > >>> reasons
> > > >> to do
> > > >>> so.
> > > >> I don't think that lets you off the hook.
> > > >> We've learned from experience that we were too vague in the
> past.
> > > >> So I would expect you to be less vague here - being more
> explicit
> > > about
> > > >> what conditions justify violating the SHOULD.
> > > >
> > > > Shall I put it back to MUST?  Keep in mind, we're not
> > trying to re-
> > > define
> > > > RFC 3261.
> > > >
> > > > Paul
> > > >
> > > >
> > > >
> > > _______________________________________________
> > > 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
> >
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 



From christer.holmberg@ericsson.com  Fri Apr 16 15:30:27 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sat, 17 Apr 2010 00:30:17 +0200
Thread-Topic: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Subject: Re: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Hi Hadriel,

> So, let's look at an example, to make sure I understand you.
> An route set has been established, and ACK is sent:
>
>
>
> UAC ----------- P1 ----------- UAS
>
> ---ACK --------->
> ---Via: UAC;keep
>                 ---ACK --------->
>                 ---Via:P1;keep
>                 ---Via:UAC;keep=3Dyes
>
> Since there is no reply to ACK, how does the UAS tell P1 that it supports
> keep-alives?

As I said in my email: it doesn't.  But that's ok.  SIP isn't broken.


> To make it work, wouldn't we need to require UPDATE or PRACK, so that UAS
> can send back a response with "Via:P1;keep=3Dyes"?

No, we don't need to require anything.  The UAS learned P1 supports keep.  =
If it wants to, it can now send keepalives to P1.  P1 has learned that the =
UAC supports keep.  If it wants to, it can send keepalives to it.

>If the UAC really wants to send keepalive for some reason, it can send a U=
PDATE, re-INVITE, or OPTIONS at its leisure, to find out about P1.  If P1 r=
eally wants to send keepalive for some reason, it can act as a UA of its ow=
n and send an out-of-dialog OPTIONS.

One of the most important use-cases, also documented in the draft, is when =
the non-registered UAC wants to send keep-alives for a dialog, e.g. for an =
emergency call. It then needs to know that P1 supports keep-alives.

>That's it.  It's not *broken* - SIP still works, the dialog still exists, =
etc.  It's just not "optimal" for this specific call-flow scenario, but who=
 cares?!?  Why double the complexity for the common use-cases, just to make=
 it less complex for the rare use-cases??

I can agree that it is rare that the UAS would want to negotiate keep-alive=
s for a dialog - because unless the UAS already sends keep-alives (e.g. ass=
ociated with its registration) the INVITE won't even reach it (assuming the=
 UAS is behind a NAT). So, IF the UAS really wants to negotiate keep-alives=
 for the dialog, IT can send an UPDATE and negotiate the keep-alives with P=
1. Because, eventhough the UAS may know that P1 supports keep-alives, P1 mi=
ght want to be informed that the UAS also support them.

BUT, in the example above, replace the UAS with P2. How does P1 know that P=
2 supports keep-alives? YOU are one of those who wanted to support proxy-to=
-proxy, so I assume you regard that as an important use-case :)

Regards,

Christer=

From HKaplan@acmepacket.com  Fri Apr 16 15:48:23 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 16 Apr 2010 18:48:02 -0400
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Subject: Re: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Friday, April 16, 2010 6:30 PM
>=20
> One of the most important use-cases, also documented in the draft, is whe=
n
> the non-registered UAC wants to send keep-alives for a dialog, e.g. for a=
n
> emergency call. It then needs to know that P1 supports keep-alives.

And the UAC is free to send an UPDATE or re-INVITE at anytime of its choosi=
ng (well, constrained by when it can send such legally).

And really, if it's a UA making an Emergency call, it's not gonna be using =
a non-session-stateful proxy as its next-hop, and besides it can and should=
 use SIP session-timers anyway.

Because as long as the next-hop proxy continues to be the one you sent the =
INVITE to, there's no "issue" with using the Via.  The only case it's not o=
ptimal in is when the next-hop is a stateless or transaction-stateful proxy=
.  If you've got one of those, NAT traversal is going to be tough anyway.


> BUT, in the example above, replace the UAS with P2. How does P1 know that
> P2 supports keep-alives? YOU are one of those who wanted to support proxy=
-
> to-proxy, so I assume you regard that as an important use-case :)

My personal use case isn't for some random next-hop proxy being used only f=
or the ACK and later messages of one dialog - it would be for either the sa=
me proxy that the INVITE went to, or for a longer-lived relationship, or ev=
en provisioned.

But anyway, the P1 proxy can send an OPTIONS-ping to find out.

-hadriel

From christer.holmberg@ericsson.com  Fri Apr 16 16:23:46 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sat, 17 Apr 2010 01:23:34 +0200
Thread-Topic: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Subject: Re: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Hi,

>>One of the most important use-cases, also documented in the draft, is whe=
n
>>the non-registered UAC wants to send keep-alives for a dialog, e.g. for a=
n
>>emergency call. It then needs to know that P1 supports keep-alives.
>
>And the UAC is free to send an UPDATE or re-INVITE at anytime of its choos=
ing (well, constrained by when it can send such legally).

I think people would have problems to require the usage of UPDATE and re-IN=
VITE for emergency calls.=20

Also, you cannot assume SIP session-timers will be required for emergency c=
alls. However, that is a separate issue.

>And really, if it's a UA making an Emergency call, it's not gonna be using=
 a non-session-stateful proxy as its next-hop, and besides it can and shoul=
d use SIP session-timers anyway.

The current assumption is that the next hop is an edge proxy, which IS stat=
eful. So, in that case it will be possible to do the keep-alive negotiation=
 as part of the initial INVITE.

(John indicated that the next hop could also be the registrar, so that woul=
d also have to be stateful)

>Because as long as the next-hop proxy continues to be the one you sent the=
 INVITE to, there's no "issue" with using the Via.  The only case it's not =
optimal in is when the=20
>next-hop is a stateless or transaction-stateful proxy.  If you've got one =
of those, NAT traversal is going to be tough anyway.

I agree.=20

So, I am happy to keep the assumption that the next hop is stateful.

>>BUT, in the example above, replace the UAS with P2. How does P1 know that
>>P2 supports keep-alives? YOU are one of those who wanted to support proxy=
-
>>to-proxy, so I assume you regard that as an important use-case :)
>
>My personal use case isn't for some random next-hop proxy being used only =
for the ACK and later messages of one dialog - it would be for either the s=
ame proxy that the INVITE went to, or for a longer-lived relationship, or e=
ven provisioned.

For proxy-to-proxy I don't think we can make the assumption that the all th=
e proxies that the INVITE traverse will record-route. And, if that is provi=
sioned, you can as well provision the usage of keep-alives - there is no ne=
ed for a protocol level negotiation...

>But anyway, the P1 proxy can send an OPTIONS-ping to find out.

But, in that case, why not *only* use OPTIONS-ping, then, and leave proxy-t=
o-proxy out of the scope of the keep-draft? I don't think a solution which =
requires a proxy to send OPTIONS is simple...

Regards,

Christer=

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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Thread-Topic: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Subject: Re: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Friday, April 16, 2010 7:24 PM
>=20
> >My personal use case isn't for some random next-hop proxy being used onl=
y
> >for the ACK and later messages of one dialog - it would be for either th=
e
> >same proxy that the INVITE went to, or for a longer-lived relationship,
> >or even provisioned.
>=20
> For proxy-to-proxy I don't think we can make the assumption that the all
> the proxies that the INVITE traverse will record-route.=20

Just to be clear, we're not making "assumptions" which affect anything crit=
ically.  All we're doing is making assumptions for what the common use-case=
s of this mechanism are, so we can have a simple mechanism for those cases,=
 while for other cases the devices might need to do more, or else they may =
not be able to leverage the benefits of this mechanism.  But that's ok.  Th=
is mechanism is not fixing something that doesn't work already today.  All =
it's doing is providing an optimization.  It's a valuable/worth-while optim=
ization, but an optimization nonetheless, and thus we don't need to address=
 every possible scenario under the sun.
Do you agree with that characterization?


> >But anyway, the P1 proxy can send an OPTIONS-ping to find out.
>=20
> But, in that case, why not *only* use OPTIONS-ping, then, and leave proxy=
-
> to-proxy out of the scope of the keep-draft? I don't think a solution
> which requires a proxy to send OPTIONS is simple...

The solution does NOT "require" sending an OPTIONS, in the sense that there=
 should be no "MUST" statement to that affect.  It's just something the pro=
xy can do if it wants to learn about the keepalive ability of another proxy=
.  The reason why a proxy might not want to only use an OPTIONS-ping is per=
formance, that's all.

If you prefer, I'm fine with making the proxy-to-proxy case just a side not=
e of this draft, by saying something like "A Proxy MAY use the keepalive in=
dication of its previous-hop or next-hop to detect that hop supports [outbo=
und] keepalives, and MAY send keepalives to that hop."  Or something like t=
hat.  And just keep the focus of the draft the UA case.

-hadriel

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To: Christer Holmberg <christer.holmberg@ericsson.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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Subject: Re: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Friday, April 16, 2010 7:24 PM
>=20
> >>One of the most important use-cases, also documented in the draft, is
> when
> >>the non-registered UAC wants to send keep-alives for a dialog, e.g. for
> an
> >>emergency call. It then needs to know that P1 supports keep-alives.
> >
> >And the UAC is free to send an UPDATE or re-INVITE at anytime of its
> choosing (well, constrained by when it can send such legally).
>=20
> I think people would have problems to require the usage of UPDATE and re-
> INVITE for emergency calls.

I think people would have problems with requiring all UAs and all Proxies t=
o support draft-keep.  I'd be fine if that could be done, I just don't see =
how it reasonably can.  It's a very useful optimization, nothing more.


> Also, you cannot assume SIP session-timers will be required for emergency
> calls. However, that is a separate issue.

And you cannot assume draft-keep will be required or supported for emergenc=
y calls either.  There is no easy answer for emergency calls, if the starti=
ng premise is any UA in existence making an emergency call through any Prox=
y in existence, without Registering and without any requirements for what t=
hey all must support. :)

If there were some global standard and authority to enforce what every UA m=
ust support to make an Emergency call, I would think the ability to send a =
re-INVITE or UPDATE at periodic intervals during the call should be pretty =
high on the list.  It's the most likely thing to work. (note that sending r=
e-INVITES or UPDATEs does not actually require support for session-timers i=
n anything)

-hadriel

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Hadriel Kaplan'" <HKaplan@acmepacket.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hadriel,
 
> The real question is if you SHOULD or SHOULD NOT have this Retry-After
> statement to begin with. ;)

I think we should at least have SHOULD here, since that's consistent with
3261.
 
> If your goal is interoperability from the customer's perspective, then
> I think explicitly specifying what the Retry-After really does/means
> *is* important.  Unfortunately, if you specify that it means no
> subsequent requests (other than OPTIONS) are sent, some of us will have
> to submit another Informational which contradicts this one because for
> a lot of customers it's the wrong behavior.

I don't want this document to change the rules as defined in 3261, unless we
want to also change 3261 in the process.

All we were hoping to do with the draft is to take a practice that is
common, but not consistently implemented, legitimize it, and specify what
devices should do.

In the process, I don't want to create contradictory statements about that
client/server interactions should be.  I do want to encourage a client to
observe a Retry-After header if it gets it, but I do not want to mandate
that servers send that header nor do I want to change anything about the
processing rules as specified in 3261.

Given that these are widely implemented by a variety of devices, I'd
personally like to see this put forward as a Standards Track RFC.  We're not
trying to define procedures to keep firewall pinholes open nor define a way
to manage server overload with this: it has a very limit, but useful scope.

Paul



From HKaplan@acmepacket.com  Sat Apr 17 18:50:27 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Paul E. Jones" <paulej@packetizer.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Sat, 17 Apr 2010 21:50:15 -0400
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Saturday, April 17, 2010 5:12 PM
>=20
> I don't want this document to change the rules as defined in 3261, unless
> we
> want to also change 3261 in the process.

It's not - 3261 says "SHOULD NOT", not "MUST NOT".  Per RFC 2119, it means:
4. SHOULD NOT   This phrase, or the phrase "NOT RECOMMENDED" mean that
   there may exist valid reasons in particular circumstances when the
   particular behavior is acceptable or even useful, but the full
   implications should be understood and the case carefully weighed
   before implementing any behavior described with this label.
=20
So your draft can simply be an Informational document, describing such circ=
umstances.  :)


> All we were hoping to do with the draft is to take a practice that is
> common, but not consistently implemented, legitimize it, and specify what
> devices should do.

Right, and I think it's bad form to recommend something that is knowingly a=
 bad idea in certain, common circumstances.

If in most circumstances people have to configure systems to NOT follow thi=
s draft's recommendations, then what good is the draft??  Isn't it in fact =
harmful?

=20
> Given that these are widely implemented by a variety of devices, I'd
> personally like to see this put forward as a Standards Track RFC.  We're
> not
> trying to define procedures to keep firewall pinholes open nor define a
> way
> to manage server overload with this: it has a very limit, but useful scop=
e.

My concern isn't about keeping firewall pinholes open.  My concern is that =
if a server becomes overloaded and sends a 503, that not only causes a comp=
lete swing to 0% utilization, but more importantly the BYE's don't get sent=
.  Likewise, some people use it for admin-down state, to take a server out =
of service gracefully - and thus still expect in-dialog requests.

-hadriel

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Hadriel Kaplan'" <HKaplan@acmepacket.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hadriel,

> > I don't want this document to change the rules as defined in 3261,
> unless
> > we
> > want to also change 3261 in the process.
> 
> It's not - 3261 says "SHOULD NOT", not "MUST NOT".  Per RFC 2119, it
> means:
> 4. SHOULD NOT   This phrase, or the phrase "NOT RECOMMENDED" mean that
>    there may exist valid reasons in particular circumstances when the
>    particular behavior is acceptable or even useful, but the full
>    implications should be understood and the case carefully weighed
>    before implementing any behavior described with this label.

The sentence in question in my draft is does not have NOT in it.  It is
this:
"... it can send a subsequent OPTIONS messages in order to detect a change
in operational status, but it SHOULD, as per RFC 3261, honor the Retry-After
header field received in the previous response."
(The above was revised as per list discussion.)

> So your draft can simply be an Informational document, describing such
> circumstances.  :)

We could do that, though I think it would be better to make it standards
track in order to encourage consistent behavior and to legitimize a practice
that is in wide use.
 
> > All we were hoping to do with the draft is to take a practice that is
> > common, but not consistently implemented, legitimize it, and specify
> what
> > devices should do.
> 
> Right, and I think it's bad form to recommend something that is
> knowingly a bad idea in certain, common circumstances.

What recommendation is in the draft that's "bad"?  Anything can be bad in
some circumstances, but I don't think an OPTIONS "ping" is a bad idea.

> If in most circumstances people have to configure systems to NOT follow
> this draft's recommendations, then what good is the draft??  Isn't it
> in fact harmful?

Potentially, but that's why I want a set of procedures people can agree to.
 
> > Given that these are widely implemented by a variety of devices, I'd
> > personally like to see this put forward as a Standards Track RFC.
> We're
> > not
> > trying to define procedures to keep firewall pinholes open nor define
> a
> > way
> > to manage server overload with this: it has a very limit, but useful
> scope.
> 
> My concern isn't about keeping firewall pinholes open.  My concern is
> that if a server becomes overloaded and sends a 503, that not only
> causes a complete swing to 0% utilization, but more importantly the
> BYE's don't get sent.  Likewise, some people use it for admin-down
> state, to take a server out of service gracefully - and thus still
> expect in-dialog requests.

Understood, but we're not proposing that a server send that lightly.  It
should be done under the same circumstances as if it received an INVITE and
decided to send a 503.  Again, this draft is not trying to change the
behavior of 3261.  We're just trying to define a consistent usage of OPTIONS
as an application-level ping mechanism.

Paul



From HKaplan@acmepacket.com  Sat Apr 17 19:27:46 2010
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To: "Paul E. Jones" <paulej@packetizer.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Sat, 17 Apr 2010 22:27:37 -0400
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Saturday, April 17, 2010 10:13 PM
>=20
> The sentence in question in my draft is does not have NOT in it.  It is
> this:
> "... it can send a subsequent OPTIONS messages in order to detect a chang=
e
> in operational status, but it SHOULD, as per RFC 3261, honor the Retry-
> After
> header field received in the previous response."
> (The above was revised as per list discussion.)

Sorry, I should have been clearer - I meant the statement in section 21.5.4=
 of 3261 about not sending subsequent requests:
   A client (proxy or UAC) receiving a 503 (Service Unavailable) SHOULD
   attempt to forward the request to an alternate server.  It SHOULD NOT
   forward any other requests to that server for the duration specified
   in the Retry-After header field, if present.

I thought that's what you and other were concerned about "changing" or "ove=
rriding".

> > So your draft can simply be an Informational document, describing such
> > circumstances.  :)
>=20
> We could do that, though I think it would be better to make it standards
> track in order to encourage consistent behavior and to legitimize a
> practice
> that is in wide use.

Sure - I just thought you wanted to keep it Informational.


> What recommendation is in the draft that's "bad"?  Anything can be bad in
> some circumstances, but I don't think an OPTIONS "ping" is a bad idea.

I meant keeping with the 3261 language I noted above, if it's not qualified=
 further.

-hadriel


From christer.holmberg@ericsson.com  Sun Apr 18 09:51:17 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sun, 18 Apr 2010 18:51:04 +0200
Thread-Topic: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Subject: Re: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Hi,

>>I think people would have problems to require the usage of UPDATE and re-=
INVITE for emergency calls.
>
>I think people would have problems with requiring all UAs and all Proxies =
to support draft-keep.  I'd be fine if that could be done, I just don't see=
 how it reasonably can.  It's a very useful optimization, nothing more.
>
>>Also, you cannot assume SIP session-timers will be required for emergency=
 calls. However, that is a separate issue.
>
>And you cannot assume draft-keep will be required or supported for emergen=
cy calls either.  There is no easy answer for emergency calls, if the start=
ing premise is any UA in existence making an emergency call through any Pro=
xy in existence, without Registering and=20
>without any requirements for what they all must support. :)

That's not what I was saying. I was saying that IF a UA supports and uses d=
raft-keep, I don't think they should have to send UPDATE/re-INVITEs for the=
 negotiation of keep-alives. Then, if they send periodic UPDATEs/re-INVITEs=
 in order to keep the session alive is a separate issue. The puropse of kee=
p-alives from UAs is to maintain NAT bindings.

And, for whatever it's worth, at least IMS UEs are required to support draf=
t-keep :)

Regards,

Christer=

From christer.holmberg@ericsson.com  Sun Apr 18 10:12:04 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sun, 18 Apr 2010 19:11:21 +0200
Thread-Topic: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Subject: Re: [sipcore] Keeping alive connections on a dialog path (was RE: Draft new version: draft-ietf-sipcore-keep-02)
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Hi,

>>>My personal use case isn't for some random next-hop proxy being used onl=
y
>>>for the ACK and later messages of one dialog - it would be for either th=
e
>>>same proxy that the INVITE went to, or for a longer-lived relationship,
>>>or even provisioned.
>>
>>For proxy-to-proxy I don't think we can make the assumption that the all
>>the proxies that the INVITE traverse will record-route.
>
>Just to be clear, we're not making "assumptions" which affect anything cri=
tically.  All we're doing is making assumptions for what the common use-cas=
es of this mechanism are, so we can have a simple mechanism for those cases=
, while for other cases the devices might need=20
>o do more, or else they may not be able to leverage the benefits of this m=
echanism.  But that's ok.  This mechanism is not fixing something that does=
n't work already today.  All it's doing is providing an optimization.  It's=
 a valuable/worth-while optimization, but an=20
>optimization nonetheless, and thus we don't need to address every possible=
 scenario under the sun.
>Do you agree with that characterization?

Are we assuming that there will be no proxy between the UA and edge proxy? =
I believe Outbound makes that assumption.

IF we make that assumption, the UA and edge proxy will be able to negotiate=
 the keep-alive as part of the initial INVITE. No need for UPDATEs/re-INVIT=
E.

IF we do NOT make that assumption, the UA will have to send UPDATE/re-INVIT=
Es, because there MAY be another proxy between the UA and edge proxy.

I think we both agree that the most common use-case will be when there is n=
o other proxy between the UA and edge proxy. But, we don't optimize the sol=
ution for that use-case if the UA needs to send UPDATE/re-INVITE in order t=
o support the proxy-in-between use-case...

>>>But anyway, the P1 proxy can send an OPTIONS-ping to find out.
>>
>>But, in that case, why not *only* use OPTIONS-ping, then, and leave proxy=
-
>>to-proxy out of the scope of the keep-draft? I don't think a solution
>>which requires a proxy to send OPTIONS is simple...
>
>The solution does NOT "require" sending an OPTIONS, in the sense that ther=
e should be no "MUST" statement to that affect.  It's just something the pr=
oxy can do if it wants to learn about the keepalive ability of another prox=
y.  The reason why a proxy might not want to=20
>only use an OPTIONS-ping is performance, that's all.
>
>If you prefer, I'm fine with making the proxy-to-proxy case just a side no=
te of this draft, by saying something like "A Proxy MAY use the keepalive i=
ndication of its previous-hop or next-hop to detect that hop supports [outb=
ound] keepalives, and MAY send keepalives to=20
>that hop."

Yes. But, then we need to describe that the proxies will not be able to neg=
otiate the keep-alives during the initial INVITE, and that they need to wai=
t for a UA to send UPDATE/re-INVITE before they can negotiate keep-alives.

>Or something like that.  And just keep the focus of the draft the UA case.

Also keep in mind that keep-alives negotiated for a dialog will only be sen=
t within that dialog. So, the question is whether people are interested in =
such proxy-to-proxy functionality in the first place? From what I've heard =
people are looking for a more "static" ping, not associated with a dialog -=
 something which option-ping provides.

Regards,

Christer=

From paulej@packetizer.com  Sun Apr 18 13:54:47 2010
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Hadriel Kaplan'" <HKaplan@acmepacket.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
References: <01a201cad2a5$ae1096c0$0a31c440$@com>	<430FC6BDED356B4C8498F634416644A91A79E9327E@mail>	<003201cad3a5$6f263540$4d729fc0$@com>	<747A6506A991724FB09B129B79D5FEB61480BBFACB@EXMBXCLUS01.citservers.local>	<002901cad90a$6e6fee10$4b4fca30$@com>	<747A6506A991724FB09B129B79D5FEB61480D10784@EXMBXCLUS01.citservers.local>	<00d701cad9c0$394a0160$abde0420$@com> <4BC3273D.7090902@cisco.com>	<003501cada4a$44b220c0$ce166240$@com> <4BC338F9.80604@cisco.com> <430FC6BDED356B4C8498F634416644A91A79F3D7D3@mail> <747A6506A991724FB09B129B79D5FEB61480D10C16@EXMBXCLUS01.citservers.local> <4BC3B9C9.7090108@cisco.com> <01a401cadc64$6ab7a120$4026e360$@com> <4BC7179F.9020604@cisco.com> <430FC6BDED356B4C8498F634416644A91A79FCEFAD@mail> <006d01cade72$ab824600$0286d200$@com> <430FC6BDED356B4C8498F634416644A91A79FCF110@mail> <009501cade9c$a6e4aff0$f4ae0fd0$@com> <430FC6BDED356B4C8498F634416644A91A79FCF111@mail>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hadriel,

Ah, OK... I think we're on the same page, then.  The changes from the -00
draft to the current draft have been extremely minimal thus far.  You can
see the current change here:
http://hive.packetizer.com/users/paulej/internet-drafts/pre-draft-jones-sip-
options-ping-01.pdf

I believe I've reflected everyone's comments and, I hope, the text does not
suggest a different direction than what is specified in 3261.

Now the next question, what shall we do with this?  As I said, I think it
ought to be a WG document.  It is, after all, behavior that is widely
implemented, but not implemented consistently.  Further, 3261 does not
specify the usage of OPTIONS for a "ping", so it needs documenting.

Paul

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> Sent: Saturday, April 17, 2010 10:28 PM
> To: Paul E. Jones; 'Paul Kyzivat'
> Cc: 'Brett Tate'; sipcore@ietf.org
> Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
> 
> 
> 
> > -----Original Message-----
> > From: Paul E. Jones [mailto:paulej@packetizer.com]
> > Sent: Saturday, April 17, 2010 10:13 PM
> >
> > The sentence in question in my draft is does not have NOT in it.  It
> is
> > this:
> > "... it can send a subsequent OPTIONS messages in order to detect a
> change
> > in operational status, but it SHOULD, as per RFC 3261, honor the
> Retry-
> > After
> > header field received in the previous response."
> > (The above was revised as per list discussion.)
> 
> Sorry, I should have been clearer - I meant the statement in section
> 21.5.4 of 3261 about not sending subsequent requests:
>    A client (proxy or UAC) receiving a 503 (Service Unavailable) SHOULD
>    attempt to forward the request to an alternate server.  It SHOULD
> NOT
>    forward any other requests to that server for the duration specified
>    in the Retry-After header field, if present.
> 
> I thought that's what you and other were concerned about "changing" or
> "overriding".
> 
> > > So your draft can simply be an Informational document, describing
> such
> > > circumstances.  :)
> >
> > We could do that, though I think it would be better to make it
> standards
> > track in order to encourage consistent behavior and to legitimize a
> > practice
> > that is in wide use.
> 
> Sure - I just thought you wanted to keep it Informational.
> 
> 
> > What recommendation is in the draft that's "bad"?  Anything can be
> bad in
> > some circumstances, but I don't think an OPTIONS "ping" is a bad
> idea.
> 
> I meant keeping with the 3261 language I noted above, if it's not
> qualified further.
> 
> -hadriel
> 



From pkyzivat@cisco.com  Sun Apr 18 16:19:04 2010
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Cc: sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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[as co-chair]

Paul E. Jones wrote:

> Now the next question, what shall we do with this?  As I said, I think it
> ought to be a WG document.  It is, after all, behavior that is widely
> implemented, but not implemented consistently.  Further, 3261 does not
> specify the usage of OPTIONS for a "ping", so it needs documenting.

Well, we already had another proposal for work on OPTIONS, but have not 
yet decided whether the WG will take this up or not.

*This* work might be considered an aspect of that, or it might be 
considered and entirely separate thing. Either way we will need to see 
if there is enough interest from the WG to do this.

So far there has been considerable activity from two people, with a 
couple of others chiming in from time to time.

It would be easier to take this up if there were broader interest.
Or potentially we could pursue advancing it as an individual draft.

I'd like to hear from some other people on this. That would help us to 
understand the extent to which this is "best practice".

	Thanks,
	Paul

From christer.holmberg@ericsson.com  Sun Apr 18 21:14:28 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Mon, 19 Apr 2010 06:14:15 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,=20

>>Now the next question, what shall we do with this?  As I said, I think=20
>>it ought to be a WG document.  It is, after all, behavior that is=20
>>widely implemented, but not implemented consistently.  Further, 3261=20
>>does not specify the usage of OPTIONS for a "ping", so it needs documenti=
ng.
>
>Well, we already had another proposal for work on OPTIONS, but have not ye=
t decided whether the WG will take this up or not.
>
>*This* work might be considered an aspect of that, or it might be consider=
ed and entirely separate thing. Either way we will need to see if there is =
enough interest from the WG to do this.
>
>So far there has been considerable activity from two people, with a couple=
 of others chiming in from time to time.
>
>It would be easier to take this up if there were broader interest.
>Or potentially we could pursue advancing it as an individual draft.
>
>I'd like to hear from some other people on this. That would help us to und=
erstand the extent to which this is "best practice".

Note that we already have a previous decission to work on a proxy-to-proxy =
ping/heartbeat MECHANISM as such, using the keep-draft.

So, I think the question is whether we want to use the keep-draft, options-=
ping, or both to implement that mechanism.

Regards,

Christer


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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Christer Holmberg wrote:
> Hi, 
> 
>>> Now the next question, what shall we do with this?  As I said, I think 
>>> it ought to be a WG document.  It is, after all, behavior that is 
>>> widely implemented, but not implemented consistently.  Further, 3261 
>>> does not specify the usage of OPTIONS for a "ping", so it needs documenting.
>> Well, we already had another proposal for work on OPTIONS, but have not yet decided whether the WG will take this up or not.
>>
>> *This* work might be considered an aspect of that, or it might be considered and entirely separate thing. Either way we will need to see if there is enough interest from the WG to do this.
>>
>> So far there has been considerable activity from two people, with a couple of others chiming in from time to time.
>>
>> It would be easier to take this up if there were broader interest.
>> Or potentially we could pursue advancing it as an individual draft.
>>
>> I'd like to hear from some other people on this. That would help us to understand the extent to which this is "best practice".
> 
> Note that we already have a previous decission to work on a proxy-to-proxy ping/heartbeat MECHANISM as such, using the keep-draft.
> 
> So, I think the question is whether we want to use the keep-draft, options-ping, or both to implement that mechanism.

People can correct me if I'm wrong - my impression is that the *intent* 
of the keep stuff and the options ping stuff is different, at least in 
emphasis. I am thinking the focus of keep is on keeping nat pinholes 
open, while the focus of options ping is on determination of which 
neighbors are up/down.

But each may be suitable for the other purpose itself. So there is 
overlap, at least in the eyes of some.

And of course there is also some perceived overlap between detecting 
neighbors that are down and overload control.

So to an extent its a matter of whether to characterize this ping work 
as part of one of those other charters, or as something independent of 
them. What is your take on that?

	Thanks,
	Paul

From christer.holmberg@ericsson.com  Mon Apr 19 06:02:30 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 19 Apr 2010 15:02:15 +0200
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Hi,=20

>>>>Now the next question, what shall we do with this?  As I said, I=20
>>>>think it ought to be a WG document.  It is, after all,=20
>>>>behavior that is widely implemented, but not implemented consistently. =
=20
>>>>Further, 3261 does not specify the usage of OPTIONS for a "ping",=20
>>>>so it needs documenting.
>>>Well, we already had another proposal for work on OPTIONS,=20
>>>but have not yet decided whether the WG will take this up or not.
>>>
>>>*This* work might be considered an aspect of that, or it=20
>>>might be considered and entirely separate thing. Either way=20
>>>we will need to see if there is enough interest from the WG=20
>>>to do this.
>>>
>>>So far there has been considerable activity from two=20
>>>people, with a couple of others chiming in from time to time.
>>>
>>>It would be easier to take this up if there were broader interest.
>>>Or potentially we could pursue advancing it as an individual draft.
>>>
>>>I'd like to hear from some other people on this. That=20
>>>would help us to understand the extent to which this is "best=20
>>>practice".
>>=20
>>Note that we already have a previous decission to work on a=20
>>proxy-to-proxy ping/heartbeat MECHANISM as such, using the keep-draft.
>>=20
>>So, I think the question is whether we want to use the=20
>>keep-draft, options-ping, or both to implement that mechanism.
>=20
>People can correct me if I'm wrong - my impression is that=20
>the *intent* of the keep stuff and the options ping stuff is=20
>different, at least in emphasis. I am thinking the focus of=20
>keep is on keeping nat pinholes open, while the focus of=20
>options ping is on determination of which neighbors are up/down.

It is correct that keeping nat pinholes open, between a UA and its edge pro=
xy/registrar, is the main focus and scope of the document.

However, at some point I was asked to also cover proxy-to-proxy heartbeat i=
n the document, and nobody objected.
=20
>But each may be suitable for the other purpose itself. So=20
>there is overlap, at least in the eyes of some.
>=20
>And of course there is also some perceived overlap between=20
>detecting neighbors that are down and overload control.
>=20
>So to an extent its a matter of whether to characterize this=20
>ping work as part of one of those other charters, or as=20
>something independent of them. What is your take on that?

Personally I don't mind removing proxy-to-proxy hearbeat from the keep-draf=
t - assuming we would adopt options-ping for that purpose. I do believe the=
 functionality is needed.

And, as I said earlier, it seems like people want to have something persist=
ant, not associated with a dialog, and if that is the case then options-pin=
g would probably be better.=20

So, it really depends on what people want between proxies. Something which =
is per dialog, or something which is more persistant - or both?

Regards,

Christer=

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>
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Date: Mon, 19 Apr 2010 12:47:35 -0400
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Cc: sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul,
 
> People can correct me if I'm wrong - my impression is that the *intent*
> of the keep stuff and the options ping stuff is different, at least in
> emphasis. I am thinking the focus of keep is on keeping nat pinholes
> open, while the focus of options ping is on determination of which
> neighbors are up/down.

That's also my understanding.
 
> But each may be suitable for the other purpose itself. So there is
> overlap, at least in the eyes of some.
> 
> And of course there is also some perceived overlap between detecting
> neighbors that are down and overload control.
> 
> So to an extent its a matter of whether to characterize this ping work
> as part of one of those other charters, or as something independent of
> them. What is your take on that?

I fully agree that the OPTIONS "ping" can be viewed as overlapping with the
other two drafts.

As you know, it was not the intent of the "ping" mechanism to serve as an
overload mechanism, nor was it intended as a means of keeping pinholes open.

I think both the keep and overload work is needed, as they work to solve
specific problems.

Likewise, I think we need the "ping" mechanism in places where the other two
would not provide the desired device status information.  We would not need
keep or overload between some devices.

Paul



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Cc: sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul E. Jones wrote:
> Paul,
>  
>> People can correct me if I'm wrong - my impression is that the *intent*
>> of the keep stuff and the options ping stuff is different, at least in
>> emphasis. I am thinking the focus of keep is on keeping nat pinholes
>> open, while the focus of options ping is on determination of which
>> neighbors are up/down.
> 
> That's also my understanding.
>  
>> But each may be suitable for the other purpose itself. So there is
>> overlap, at least in the eyes of some.
>>
>> And of course there is also some perceived overlap between detecting
>> neighbors that are down and overload control.
>>
>> So to an extent its a matter of whether to characterize this ping work
>> as part of one of those other charters, or as something independent of
>> them. What is your take on that?
> 
> I fully agree that the OPTIONS "ping" can be viewed as overlapping with the
> other two drafts.
> 
> As you know, it was not the intent of the "ping" mechanism to serve as an
> overload mechanism, nor was it intended as a means of keeping pinholes open.
> 
> I think both the keep and overload work is needed, as they work to solve
> specific problems.
> 
> Likewise, I think we need the "ping" mechanism in places where the other two
> would not provide the desired device status information.  We would not need
> keep or overload between some devices.

I don't *in principle* have a problem with treating these as separate 
things. But then we must ensure that they don't have unintended 
(negative) interactions. That should be managed by careful scoping and 
by ensuring that they take account of one another.

Christer indicated that there was a previous decision to include 
proxy-proxy keep-alives in the -keep draft. Christer - do you know where 
that decision might be documented? The existing goal list is very vague 
on this.

If we were to piggyback options-ping onto that goal, it would make it 
even later than it already is. Otherwise it will need a new goal to 
become a WG draft. We will need some support from the ADs for that. But 
before asking them, I would like to see a broader level of interest on 
this mailing list.

	THanks
	Paul

From christer.holmberg@ericsson.com  Mon Apr 19 14:14:12 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Mon, 19 Apr 2010 23:13:58 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,=20

>>> People can correct me if I'm wrong - my impression is that the=20
>>> *intent* of the keep stuff and the options ping stuff is different,=20
>>> at least in emphasis. I am thinking the focus of keep is on keeping=20
>>> nat pinholes open, while the focus of options ping is on=20
>>> determination of which neighbors are up/down.
>>=20
>> That's also my understanding.
>> =20
>>> But each may be suitable for the other purpose itself. So there is=20
>>> overlap, at least in the eyes of some.
>>>
>>> And of course there is also some perceived overlap between detecting=20
>>> neighbors that are down and overload control.
>>>
>>> So to an extent its a matter of whether to characterize this ping=20
>>> work as part of one of those other charters, or as something=20
>>> independent of them. What is your take on that?
>>=20
>> I fully agree that the OPTIONS "ping" can be viewed as overlapping=20
>> with the other two drafts.
>>=20
>> As you know, it was not the intent of the "ping" mechanism to serve as=20
>> an overload mechanism, nor was it intended as a means of keeping pinhole=
s open.
>>=20
>> I think both the keep and overload work is needed, as they work to=20
>> solve specific problems.
>>=20
>> Likewise, I think we need the "ping" mechanism in places where the=20
>> other two would not provide the desired device status information.  We=20
>> would not need keep or overload between some devices.
>
>I don't *in principle* have a problem with treating these as separate thin=
gs. But then we must ensure that they don't have unintended
>(negative) interactions. That should be managed by careful scoping and by =
ensuring that they take account of one another.
>
>Christer indicated that there was a previous decision to include proxy-pro=
xy keep-alives in the -keep draft. Christer - do you know where that decisi=
on might be documented? The existing goal list is very=20
>vague on this.

I haven't checked the meeting minutes, but the use-case was in the slides I=
 presented at the Dublin meeting.

Regards,

Christer


From rjsparks@nostrum.com  Tue Apr 20 12:53:29 2010
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Subject: [sipcore] Fwd: Registration closes in 10 days (Re: [RAI] SIPit26 registration is open)
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fyi

Begin forwarded message:

> From: Robert Sparks <rjsparks@nostrum.com>
> Date: April 20, 2010 2:52:43 PM CDT
> To: rai@ietf.org
> Subject: Registration closes in 10 days (Re: [RAI] SIPit26 =
registration is open)
>=20
> Registration closes May 1. If you are planning to attend and have not =
yet registered, please do so soon.
>=20
> RjS
>=20
> On Mar 16, 2010, at 11:15 AM, Robert Sparks wrote:
>=20
>> SIPit 26 will be held in Kista - the telecom valley outside =
Stockholm, Sweden on May 17-21, 2010. The event is hosted by Edvina and =
TANDBERG and sponsored by Ingate, Intertex and .se.
>>=20
>> See <http://www.sipit.net> and <http://sipit.edvina.se> for details =
and to register.
>>=20
>> RjS
>> _______________________________________________
>> RAI mailing list
>> RAI@ietf.org
>> https://www.ietf.org/mailman/listinfo/rai
>=20


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">fyi<br><div><br><div>Begin forwarded message:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>From: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;">Robert Sparks =
&lt;<a =
href=3D"mailto:rjsparks@nostrum.com">rjsparks@nostrum.com</a>&gt;<br></spa=
n></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>Date: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;">April 20, 2010 =
2:52:43 PM CDT<br></span></div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1);"><b>To: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;"><a =
href=3D"mailto:rai@ietf.org">rai@ietf.org</a><br></span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>Subject: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;"><b>Registration =
closes in 10 days (Re: [RAI] SIPit26 registration is =
open)</b><br></span></div><br><div>Registration closes May 1. If you are =
planning to attend and have not yet registered, please do so =
soon.<br><br>RjS<br><br>On Mar 16, 2010, at 11:15 AM, Robert Sparks =
wrote:<br><br><blockquote type=3D"cite">SIPit 26 will be held in Kista - =
the telecom valley outside Stockholm, Sweden on May 17-21, 2010. The =
event is hosted by Edvina and TANDBERG and sponsored by Ingate, Intertex =
and .se.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">See &lt;<a =
href=3D"http://www.sipit.net">http://www.sipit.net</a>&gt; and &lt;<a =
href=3D"http://sipit.edvina.se">http://sipit.edvina.se</a>&gt; for =
details and to register.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite">RjS<br></blockquote><blockquote =
type=3D"cite">_______________________________________________<br></blockqu=
ote><blockquote type=3D"cite">RAI mailing =
list<br></blockquote><blockquote type=3D"cite"><a =
href=3D"mailto:RAI@ietf.org">RAI@ietf.org</a><br></blockquote><blockquote =
type=3D"cite"><a =
href=3D"https://www.ietf.org/mailman/listinfo/rai">https://www.ietf.org/ma=
ilman/listinfo/rai</a><br></blockquote><br></div></blockquote></div><br></=
body></html>=

--Apple-Mail-13--719262017--

From mlm.michael.miller@gmail.com  Wed Apr 21 07:38:09 2010
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From: Michael Miller <mlm.michael.miller@gmail.com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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--0015174c1b362691ac0484bfb838
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> Hi,
>
> >>> People can correct me if I'm wrong - my impression is that the
> >>> *intent* of the keep stuff and the options ping stuff is different,
> >>> at least in emphasis. I am thinking the focus of keep is on keeping
> >>> nat pinholes open, while the focus of options ping is on
> >>> determination of which neighbors are up/down.
> >>
> >> That's also my understanding.
> >>
> >>> But each may be suitable for the other purpose itself. So there is
> >>> overlap, at least in the eyes of some.
> >>>
> >>> And of course there is also some perceived overlap between detecting
> >>> neighbors that are down and overload control.
> >>>
> >>> So to an extent its a matter of whether to characterize this ping
> >>> work as part of one of those other charters, or as something
> >>> independent of them. What is your take on that?
> >>
> >> I fully agree that the OPTIONS "ping" can be viewed as overlapping
> >> with the other two drafts.
> >>
> >> As you know, it was not the intent of the "ping" mechanism to serve as
> >> an overload mechanism, nor was it intended as a means of keeping
> pinholes open.
> >>
> >> I think both the keep and overload work is needed, as they work to
> >> solve specific problems.
> >>
> >> Likewise, I think we need the "ping" mechanism in places where the
> >> other two would not provide the desired device status information.  We
> >> would not need keep or overload between some devices.
> >
> >I don't *in principle* have a problem with treating these as separate
> things. But then we must ensure that they don't have unintended
> >(negative) interactions. That should be managed by careful scoping and by
> ensuring that they take account of one another.
> >
> >Christer indicated that there was a previous decision to include
> proxy-proxy keep-alives in the -keep draft. Christer - do you know where
> that decision might be documented? The existing goal list is very
> >vague on this.
>
> I haven't checked the meeting minutes, but the use-case was in the slides I
> presented at the Dublin meeting.
>
> Regards,
>
> Christer
>

Hi, I'm new to this process, so please be patience with me.

The draft looks good and I just want to make sure of a couple points:
1-We have endpoints that require Notify or Invite to be used for this ping.
I think by saying use the method OPTIONS I am hoping that is the only way to
perform this function.
2-Should we come up with a recommend (or default value) to be used on how
often we should send.  Concerns is that some devices may not reply back if
they get so many in a 30 second period (for an example).

Michael Miller
Global Crossing.


>
>

--0015174c1b362691ac0484bfb838
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<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"b=
order-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; paddin=
g-left: 1ex;">
Hi,<br>
<br>
&gt;&gt;&gt; People can correct me if I&#39;m wrong - my impression is that=
 the<br>
&gt;&gt;&gt; *intent* of the keep stuff and the options ping stuff is diffe=
rent,<br>
&gt;&gt;&gt; at least in emphasis. I am thinking the focus of keep is on ke=
eping<br>
&gt;&gt;&gt; nat pinholes open, while the focus of options ping is on<br>
&gt;&gt;&gt; determination of which neighbors are up/down.<br>
&gt;&gt;<br>
&gt;&gt; That&#39;s also my understanding.<br>
&gt;&gt;<br>
&gt;&gt;&gt; But each may be suitable for the other purpose itself. So ther=
e is<br>
&gt;&gt;&gt; overlap, at least in the eyes of some.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; And of course there is also some perceived overlap between det=
ecting<br>
&gt;&gt;&gt; neighbors that are down and overload control.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; So to an extent its a matter of whether to characterize this p=
ing<br>
&gt;&gt;&gt; work as part of one of those other charters, or as something<b=
r>
&gt;&gt;&gt; independent of them. What is your take on that?<br>
&gt;&gt;<br>
&gt;&gt; I fully agree that the OPTIONS &quot;ping&quot; can be viewed as o=
verlapping<br>
&gt;&gt; with the other two drafts.<br>
&gt;&gt;<br>
&gt;&gt; As you know, it was not the intent of the &quot;ping&quot; mechani=
sm to serve as<br>
&gt;&gt; an overload mechanism, nor was it intended as a means of keeping p=
inholes open.<br>
&gt;&gt;<br>
&gt;&gt; I think both the keep and overload work is needed, as they work to=
<br>
&gt;&gt; solve specific problems.<br>
&gt;&gt;<br>
&gt;&gt; Likewise, I think we need the &quot;ping&quot; mechanism in places=
 where the<br>
&gt;&gt; other two would not provide the desired device status information.=
 =A0We<br>
&gt;&gt; would not need keep or overload between some devices.<br>
&gt;<br>
&gt;I don&#39;t *in principle* have a problem with treating these as separa=
te things. But then we must ensure that they don&#39;t have unintended<br>
&gt;(negative) interactions. That should be managed by careful scoping and =
by ensuring that they take account of one another.<br>
&gt;<br>
&gt;Christer indicated that there was a previous decision to include proxy-=
proxy keep-alives in the -keep draft. Christer - do you know where that dec=
ision might be documented? The existing goal list is very<br>
&gt;vague on this.<br>
<br>
I haven&#39;t checked the meeting minutes, but the use-case was in the slid=
es I presented at the Dublin meeting.<br>
<br>
Regards,<br>
<br>
Christer<br></blockquote><div><br>Hi, I&#39;m new to this process, so pleas=
e be patience with me.<br><br>The draft looks good and I just want to make =
sure of a couple points:<br>1-We have endpoints that require Notify or Invi=
te to be used for this ping.=A0 I think by saying use the method OPTIONS I =
am hoping that is the only way to perform this function.<br>
2-Should we come up with a recommend (or default value) to be used on how o=
ften we should send.=A0 Concerns is that some devices may not reply back if=
 they get so many in a 30 second period (for an example).<br><br>Michael Mi=
ller<br>
Global Crossing.<br>=A0</div><blockquote class=3D"gmail_quote" style=3D"bor=
der-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-=
left: 1ex;">
<br>
</blockquote></div><br>

--0015174c1b362691ac0484bfb838--

From rbarnes@bbn.com  Thu Apr 22 12:24:51 2010
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Subject: [sipcore] Announcement: Emergency Services Workshop 7
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Dear colleagues,

We're pleased to announce that registration is open for the seventh  
Emergency Services Workshop, to be held 11-13 May 2010, at the  
University of Maryland, in College Park, MD, USA.

A registration link, as well as agenda and travel information, can be  
found on the ESW website:
<http://www.emergency-services-coordination.info/esw7.html>

We invite talks of 30-60 minutes on topics related to IP emergency  
services, especially around the following topics:
-- Standards development activities related to emergency communications
-- Prototypes of standards-based IP emergency services
-- Internet-based authority-to-citizen alerts and early warning
-- Automatic IP geolocation technologies and deployment considerations
-- Implementation and deployment of IP-based emergency calling systems
Please send proposals for presentations to <esw- 
submit@googlegroups.com>.

Background:
The Emergency Services Workshop series is an ongoing effort in the
emergency services community to coordinate global standards and
technologies for emergency calling and emergency notification. The
primary focus of the workshop series is foster coordination among the
many standards development organizations (SDOs) involved in emergency
services as they all work toward a global solution for emergency
communications using Internet technologies. In addition, the workshops
try to bring in operational and regulatory perspectives on emergency
services, so that these experiences and requirements can be
incorporated into ongoing technical development processes.
Participation is open  all stakeholders in the emergency
communications system, including industry (e.g., equipment vendors or
telecommunications companies) as well as government (e.g., regulatory
bodies or emergency response  organizations).

Thanks for your interest,
The ESW Coordination Team

From paulej@packetizer.com  Fri Apr 23 08:49:06 2010
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Michael Miller'" <mlm.michael.miller@gmail.com>, <sipcore@ietf.org>
References: <p2j53f87ca11004210707td7869937s83ec155fdd0f9ffc@mail.gmail.com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Michael,

 

It was our intent to recommend use of OPTIONS only for "ping".  The very
fact you've got a couple of implementations that use different messages for
the same purpose is evidence that we need to settle on one way.  Having
multiple ways to do this is clearly not in the best interest of customers.

 

As for the frequency of sending messages, I'm torn on this one.  I've heard
numbers that vary considerably.  Since this is something done between
devices that have some kind of relationship, I'd prefer to let the frequency
be determined by administrators.  That is what we put in Section 7.
However, if there are strong preferences to recommend an interval, I'll
change it.  But, I think we could only recommend something, not require it.

 

Paul

 

From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf
Of Michael Miller
Sent: Wednesday, April 21, 2010 10:08 AM
To: sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt

 

 

Hi,

>>> People can correct me if I'm wrong - my impression is that the
>>> *intent* of the keep stuff and the options ping stuff is different,
>>> at least in emphasis. I am thinking the focus of keep is on keeping
>>> nat pinholes open, while the focus of options ping is on
>>> determination of which neighbors are up/down.
>>
>> That's also my understanding.
>>
>>> But each may be suitable for the other purpose itself. So there is
>>> overlap, at least in the eyes of some.
>>>
>>> And of course there is also some perceived overlap between detecting
>>> neighbors that are down and overload control.
>>>
>>> So to an extent its a matter of whether to characterize this ping
>>> work as part of one of those other charters, or as something
>>> independent of them. What is your take on that?
>>
>> I fully agree that the OPTIONS "ping" can be viewed as overlapping
>> with the other two drafts.
>>
>> As you know, it was not the intent of the "ping" mechanism to serve as
>> an overload mechanism, nor was it intended as a means of keeping pinholes
open.
>>
>> I think both the keep and overload work is needed, as they work to
>> solve specific problems.
>>
>> Likewise, I think we need the "ping" mechanism in places where the
>> other two would not provide the desired device status information.  We
>> would not need keep or overload between some devices.
>
>I don't *in principle* have a problem with treating these as separate
things. But then we must ensure that they don't have unintended
>(negative) interactions. That should be managed by careful scoping and by
ensuring that they take account of one another.
>
>Christer indicated that there was a previous decision to include
proxy-proxy keep-alives in the -keep draft. Christer - do you know where
that decision might be documented? The existing goal list is very
>vague on this.

I haven't checked the meeting minutes, but the use-case was in the slides I
presented at the Dublin meeting.

Regards,

Christer


Hi, I'm new to this process, so please be patience with me.

The draft looks good and I just want to make sure of a couple points:
1-We have endpoints that require Notify or Invite to be used for this ping.
I think by saying use the method OPTIONS I am hoping that is the only way to
perform this function.
2-Should we come up with a recommend (or default value) to be used on how
often we should send.  Concerns is that some devices may not reply back if
they get so many in a 30 second period (for an example).

Michael Miller
Global Crossing.
 

 

 


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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Michael,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>It was our intent to recommend use of OPTIONS only for =
&#8220;ping&#8221;.&nbsp;
The very fact you&#8217;ve got a couple of implementations that use =
different
messages for the same purpose is evidence that we need to settle on one
way.&nbsp; Having multiple ways to do this is clearly not in the best =
interest
of customers.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>As for the frequency of sending messages, I&#8217;m torn =
on this
one. &nbsp;I&#8217;ve heard numbers that vary considerably.&nbsp; Since =
this is
something done between devices that have some kind of relationship, =
I&#8217;d
prefer to let the frequency be determined by administrators.&nbsp; That =
is what
we put in Section 7.&nbsp; However, if there are strong preferences to
recommend an interval, I&#8217;ll change it.&nbsp; But, I think we could =
only
recommend something, not require it.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Paul<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
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color:#1F497D'><o:p>&nbsp;</o:p></span></p>

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0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] <b>On Behalf =
Of </b>Michael
Miller<br>
<b>Sent:</b> Wednesday, April 21, 2010 10:08 AM<br>
<b>To:</b> sipcore@ietf.org<br>
<b>Subject:</b> Re: [sipcore] FW: I-D =
Action:draft-jones-sip-options-ping-00.txt<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<blockquote style=3D'border:none;border-left:solid #CCCCCC =
1.0pt;padding:0in 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>

<p class=3DMsoNormal>Hi,<br>
<br>
&gt;&gt;&gt; People can correct me if I'm wrong - my impression is that =
the<br>
&gt;&gt;&gt; *intent* of the keep stuff and the options ping stuff is
different,<br>
&gt;&gt;&gt; at least in emphasis. I am thinking the focus of keep is on
keeping<br>
&gt;&gt;&gt; nat pinholes open, while the focus of options ping is =
on<br>
&gt;&gt;&gt; determination of which neighbors are up/down.<br>
&gt;&gt;<br>
&gt;&gt; That's also my understanding.<br>
&gt;&gt;<br>
&gt;&gt;&gt; But each may be suitable for the other purpose itself. So =
there is<br>
&gt;&gt;&gt; overlap, at least in the eyes of some.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; And of course there is also some perceived overlap between
detecting<br>
&gt;&gt;&gt; neighbors that are down and overload control.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; So to an extent its a matter of whether to characterize =
this ping<br>
&gt;&gt;&gt; work as part of one of those other charters, or as =
something<br>
&gt;&gt;&gt; independent of them. What is your take on that?<br>
&gt;&gt;<br>
&gt;&gt; I fully agree that the OPTIONS &quot;ping&quot; can be viewed =
as
overlapping<br>
&gt;&gt; with the other two drafts.<br>
&gt;&gt;<br>
&gt;&gt; As you know, it was not the intent of the &quot;ping&quot; =
mechanism
to serve as<br>
&gt;&gt; an overload mechanism, nor was it intended as a means of =
keeping
pinholes open.<br>
&gt;&gt;<br>
&gt;&gt; I think both the keep and overload work is needed, as they work =
to<br>
&gt;&gt; solve specific problems.<br>
&gt;&gt;<br>
&gt;&gt; Likewise, I think we need the &quot;ping&quot; mechanism in =
places
where the<br>
&gt;&gt; other two would not provide the desired device status =
information.
&nbsp;We<br>
&gt;&gt; would not need keep or overload between some devices.<br>
&gt;<br>
&gt;I don't *in principle* have a problem with treating these as =
separate
things. But then we must ensure that they don't have unintended<br>
&gt;(negative) interactions. That should be managed by careful scoping =
and by
ensuring that they take account of one another.<br>
&gt;<br>
&gt;Christer indicated that there was a previous decision to include
proxy-proxy keep-alives in the -keep draft. Christer - do you know where =
that
decision might be documented? The existing goal list is very<br>
&gt;vague on this.<br>
<br>
I haven't checked the meeting minutes, but the use-case was in the =
slides I
presented at the Dublin meeting.<br>
<br>
Regards,<br>
<br>
Christer<o:p></o:p></p>

</blockquote>

<div>

<p class=3DMsoNormal><br>
Hi, I'm new to this process, so please be patience with me.<br>
<br>
The draft looks good and I just want to make sure of a couple =
points:<br>
1-We have endpoints that require Notify or Invite to be used for this
ping.&nbsp; I think by saying use the method OPTIONS I am hoping that is =
the
only way to perform this function.<br>
2-Should we come up with a recommend (or default value) to be used on =
how often
we should send.&nbsp; Concerns is that some devices may not reply back =
if they
get so many in a 30 second period (for an example).<br>
<br>
Michael Miller<br>
Global Crossing.<br>
&nbsp;<o:p></o:p></p>

</div>

<blockquote style=3D'border:none;border-left:solid #CCCCCC =
1.0pt;padding:0in 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</blockquote>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</div>

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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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On Apr 23, 2010, at 10:48 AM, Paul E. Jones wrote:

> Michael,
>
> It was our intent to recommend use of OPTIONS only for =93ping=94.  =
The =20
> very fact you=92ve got a couple of implementations that use different =20=

> messages for the same purpose is evidence that we need to settle on =20=

> one way.  Having multiple ways to do this is clearly not in the best =20=

> interest of customers.

There's probably been some discussion of this that has just slipped =20
out of my head, but we may wish to consider that OPTIONS gets =20
"heavier" with every additional SIP extension. Eventually, it'll get =20
too big for UDP, and become fairly useless for pinging.

Factor in that some people seem to be implementing OPTIONS to provide =20=

a different answer based on "who's asking", and we add the overhead of =20=

(optionally) authenticating the request,  making an options-selection =20=

based on what the requester is authorized to see, and then formulating =20=

the specific reply.


Two alternatives suggest themselves to me:

1) Defining a new method, like PING, specifically constructed to have =20=

no semantic other than "Are you there?"

2) Holding open a long-lived dialog with the node you are "pinging" =20
and using keepalive (and or session-timer) to make sure it's still =20
there. At least this lets the AAA process happen only once (for some =20
values of authentication) and lets us reuse a long-lived TCP (or TLS) =20=

connection if appropriate.

I somewhat prefer the second; it has better backward-compatibility, =20
and helps discomfit all those people I've been telling not to bill =20
based on INVITE/BYE for so long.

--
Dean=

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Dean,

> Factor in that some people seem to be implementing OPTIONS to provide
> a different answer based on "who's asking", and we add the overhead of
> (optionally) authenticating the request,  making an options-selection
> based on what the requester is authorized to see, and then formulating
> the specific reply.

Your concerns are valid, but devices implementing a "ping" know they're
doing that, and would not have to fatten the OPTIONS message.  They also
would not need to authenticate the request, either.  At a minimal, they
respond in all cases, and as one step toward "security" (if the "ping"
needed it), one could employ an ACL.

> 1) Defining a new method, like PING, specifically constructed to have
> no semantic other than "Are you there?"

>From an architectural purity perspective, I like this.  The problem is that
it requires changes to every stack in the field.  It also means we have one
more way to do "ping", since I suspect people will continue to use OPTIONS.

There was also a suggestion to have a tag to indicate that the OPTIONS is
for "ping", though I think the Request URI could serve the same purpose. One
should not ping sip:paulej@packetizer.com, while one might ping
sip:192.168.1.10 or sip:ping@192.168.1.10 and the responding entity should
formulate a reasonable (non-heavy) reply.
 
> 2) Holding open a long-lived dialog with the node you are "pinging"
> and using keepalive (and or session-timer) to make sure it's still
> there. At least this lets the AAA process happen only once (for some
> values of authentication) and lets us reuse a long-lived TCP (or TLS)
> connection if appropriate.

This likewise requires a bit of change on most devices to understand that
such sessions are "ping" sessions.  It also requires a bit more state
management on both devices.  It's not horrible, but not nearly as
light-weight as sending an OPTIONS "ping" every now and then.  Further,
consider that sending a "ping" is not really necessary for devices that know
(through other means) that the peer device is alive (e.g., active dialogs
starting/stopping).  This approach would not benefit from that. 
 
> I somewhat prefer the second; it has better backward-compatibility,
> and helps discomfit all those people I've been telling not to bill
> based on INVITE/BYE for so long.

Billing?  We don't need no stinkin' bills ;-)

Obviously, I'm partial to the OOD OPTIONS "ping" method.  It's fairly widely
(albeit inconsistently) implemented.  Other options seem to require more
work in terms of specification and implementation.

Paul



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Date: Fri, 23 Apr 2010 19:26:11 +0200
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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Dumb question: if nodes A and B were exchanging traffic over both UDP
and TCP, should they exchange "PING" messages for/over every
transport?

-Victor

On Fri, Apr 23, 2010 at 7:05 PM, Hadriel Kaplan <HKaplan@acmepacket.com> wr=
ote:
> I think if we just agree on a common explicit target username, then all t=
hose issues are avoided. =C2=A0We'd just specify the user identifies a logi=
cal UA in the proxy/whatever, which is only there for pinging. =C2=A0The pu=
rpose of the request would thus be explicit.
>
> And there's no point in authenticating, since the 401 itself is a respons=
e, and besides the information returned is benign.
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>



--=20
Victor Pascual =C3=81vila

From pkyzivat@cisco.com  Fri Apr 23 10:28:05 2010
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[ as individual ]

Paul E. Jones wrote:
> Dean,
> 
>> Factor in that some people seem to be implementing OPTIONS to provide
>> a different answer based on "who's asking", and we add the overhead of
>> (optionally) authenticating the request,  making an options-selection
>> based on what the requester is authorized to see, and then formulating
>> the specific reply.
> 
> Your concerns are valid, but devices implementing a "ping" know they're
> doing that, and would not have to fatten the OPTIONS message.  They also
> would not need to authenticate the request, either.  At a minimal, they
> respond in all cases, and as one step toward "security" (if the "ping"
> needed it), one could employ an ACL.
...
> There was also a suggestion to have a tag to indicate that the OPTIONS is
> for "ping", though I think the Request URI could serve the same purpose. One
> should not ping sip:paulej@packetizer.com, while one might ping
> sip:192.168.1.10 or sip:ping@192.168.1.10 and the responding entity should
> formulate a reasonable (non-heavy) reply.

I'm bothered by this - it makes too many assumptions about usage.

The *sender* knows when the options is a ping and when it is not, but 
the recipient doesn't.

Making a decision based on the form of the R-URI presumes that non-ping 
users of options never use an ip address as the R-URI. Requiring the 
R-URI to be ping@ip reserves a user part, and also is just a sleazy way 
of adding another indicator to the message to indicate its intended use.

One possibility would be to use an option tag: Required:ping. That would 
ensure the other party knows this is a ping. (And of course if that 
fails you could remove the tag, and realize you were not getting a 
specialized ping implementation.)

But that itself requires parsing the message enough to recognize the 
option tag. I know of at least one implementation that optimizes OPTIONS 
deep in the stack - replying immediately without ever passing it to the 
applications layer, and without parsing much of the message. But it 
depends on knowing its in an environment where that is acceptable.

I'd certainly like to see the pros/cons debated thoroughly.

	Thanks,
	Paul

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Victor Pascual Avila'" <victor.pascual.avila@gmail.com>, "'Hadriel Kaplan'" <HKaplan@acmepacket.com>
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Victor,

I suppose they could, but I wouldn't.  I'd configure the use of whatever =
transport made sense to ping the peer device, which might include =
Henry's infamous carrier pigeon.

Paul

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Victor Pascual Avila
> Sent: Friday, April 23, 2010 1:26 PM
> To: Hadriel Kaplan
> Cc: Michael Miller; Paul E. Jones; sipcore@ietf.org
> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
>=20
> Dumb question: if nodes A and B were exchanging traffic over both UDP
> and TCP, should they exchange "PING" messages for/over every
> transport?
>=20
> -Victor
>=20
> On Fri, Apr 23, 2010 at 7:05 PM, Hadriel Kaplan
> <HKaplan@acmepacket.com> wrote:
> > I think if we just agree on a common explicit target username, then
> all those issues are avoided.  We'd just specify the user identifies a
> logical UA in the proxy/whatever, which is only there for pinging.  =
The
> purpose of the request would thus be explicit.
> >
> > And there's no point in authenticating, since the 401 itself is a
> response, and besides the information returned is benign.
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
>=20
>=20
>=20
> --
> Victor Pascual =C3=81vila
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From dean.willis@softarmor.com  Fri Apr 23 11:55:27 2010
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On Apr 23, 2010, at 12:27 PM, Paul Kyzivat wrote:

>
> One possibility would be to use an option tag: Required:ping. That  
> would ensure the other party knows this is a ping. (And of course if  
> that fails you could remove the tag, and realize you were not  
> getting a specialized ping implementation.)
>


Nah, that just requires somebody to support the ping extension; it  
doesn't tell them to apply that extension to this OPTIONS request.

> But that itself requires parsing the message enough to recognize the  
> option tag. I know of at least one implementation that optimizes  
> OPTIONS deep in the stack - replying immediately without ever  
> passing it to the applications layer, and without parsing much of  
> the message. But it depends on knowing its in an environment where  
> that is acceptable.

How does the implementation know which options are supported at the  
application level? It probably doesn't, and is sending a arong options- 
set in the response.

--
Dean


From dean.willis@softarmor.com  Fri Apr 23 11:59:59 2010
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Subject: [sipcore] Discussion on Table 2 relative to RFC3329 errata (2170)
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On Apr 23, 2010, at 8:27 AM, RFC Errata System wrote:
>
>
> Original Text
> -------------
> The second INVITE (4) and the ACK (8) contain a Security-Verify
>   header field that mirrors the Security-Server header field received
>   in the 421.
>
> Corrected Text
> --------------
> The second INVITE (4) contains a Security-Verify
>   header field that mirrors the Security-Server header field received
>   in the 421.
>
> Notes
> -----
> RFC 3329 Section 2.6, Table 1: Summary of Header Usage. indicates  
> that Security-Client, Security-Server, Security-Verify are "Not  
> applicable" to the SIP ACK request.
>
> RFC 3261 says "Not applicable" means that the header field MUST NOT  
> be present in a request.  If one is placed in a request by mistake,  
> it MUST be ignored by the UAS receiving the request."
>


We've been discussing the role of Table 2. Here, said table (as  
modified by the RFC in question) is being viewed as authoritative on  
the inclusion of Security-Verify in the examples used by that RFC.

Clearly, we have a mismatch between the specification and its table 2  
modification.

Is Table 2 here helping us catch spec errors, or is it just confusing  
the specification writer?

--
dean



From pkyzivat@cisco.com  Fri Apr 23 12:25:51 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Dean,

OK, you caught me falling into that trap too. :-(

In any case, once you are going to mandate something new, it doesn't 
matter much whether it is an option tag or a method. Either way you have 
to implement something new.

Similarly, if you mandate that a particular R-URI format is to be 
treated differently, that also is something new to implement.

The *advantage* of OPTIONS is that it can be used as a ping without the 
recipient having any knowledge that is the purpose. But if it is 
unsuitable for use with those that don't understand ping as something 
special, then it has no advantage.

I guess the argument remains that if you use some special form of ping 
that will still work for the clueless, but which can be optimized by the 
clueful, then you have the best of both worlds.

For that to work well, the discriminator should be something that is 
certain not to cause trouble for the clueless, and that doesn't limit 
the clueful who also use OPTIONS for non-ping purposes. I think that 
rules out R-URI values. But it could include new header fields. Note 
that you have to implement *something* new in the UAS if you want to 
optimize pings. So implementing a new header should not be a big deal.

	Thanks,
	Paul

Dean Willis wrote:
> 
> On Apr 23, 2010, at 12:27 PM, Paul Kyzivat wrote:
> 
>>
>> One possibility would be to use an option tag: Required:ping. That 
>> would ensure the other party knows this is a ping. (And of course if 
>> that fails you could remove the tag, and realize you were not getting 
>> a specialized ping implementation.)
>>
> 
> 
> Nah, that just requires somebody to support the ping extension; it 
> doesn't tell them to apply that extension to this OPTIONS request.
> 
>> But that itself requires parsing the message enough to recognize the 
>> option tag. I know of at least one implementation that optimizes 
>> OPTIONS deep in the stack - replying immediately without ever passing 
>> it to the applications layer, and without parsing much of the message. 
>> But it depends on knowing its in an environment where that is acceptable.
> 
> How does the implementation know which options are supported at the 
> application level? It probably doesn't, and is sending a arong 
> options-set in the response.
> 
> -- 
> Dean
> 
> 

From christer.holmberg@ericsson.com  Fri Apr 23 14:56:39 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Fri, 23 Apr 2010 23:56:24 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

>> One possibility would be to use an option tag: Required:ping. That
>> would ensure the other party knows this is a ping. (And of course if
>> that fails you could remove the tag, and realize you were not
>> getting a specialized ping implementation.)
>
>
>
>Nah, that just requires somebody to support the ping extension; it
>doesn't tell them to apply that extension to this OPTIONS request.

I don't think you can make such general assumption. It needs to be specifie=
d per option tag.

For example, if you receive an INVITE with Require:100rel, you must not onl=
y support the extension - you must also use it when sending provisional res=
ponses (see RFC3262, section 3, paragraph 2).

If you don't think is correct we'd need to update RFC3262.

In addition, the scope of an option-tag can be a single transaction, or wid=
er.=20

For example, just because I require 100rel in the initial INVITE, it doesn'=
t mean (at least that is my understanding) I have implicitly also required =
it in each subsequent re-INVITE.

The scope of Require:sec-agree, on the other hand, is for the whole session=
.=20

Is there ANY option-tag defined, where we specify that Require only means m=
ust-support-but-is-not-required-to-use?

Anyway, if we move forward with options-ping, I DO think there should be a =
way to identify the OPTIONS as a ping, so that the receiver e.g. doens't ne=
ed to include all supported capabilities etc in the response - or expect th=
em in the request.

Regards,

Christer=

From dean.willis@softarmor.com  Fri Apr 23 15:11:13 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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On Apr 23, 2010, at 4:56 PM, Christer Holmberg wrote:

> Hi,
>
>>> One possibility would be to use an option tag: Required:ping. That
>>> would ensure the other party knows this is a ping. (And of course if
>>> that fails you could remove the tag, and realize you were not
>>> getting a specialized ping implementation.)
>>
>>
>>
>> Nah, that just requires somebody to support the ping extension; it
>> doesn't tell them to apply that extension to this OPTIONS request.
>
> I don't think you can make such general assumption. It needs to be  
> specified per option tag.
>
> For example, if you receive an INVITE with Require:100rel, you must  
> not only support the extension - you must also use it when sending  
> provisional responses (see RFC3262, section 3, paragraph 2).
>

We've had this discussion before: 100rel was badly written in this  
respect.

For example, what do you do if you receive a request that has Require:  
100rel but not Supported: 100rel?


> If you don't think is correct we'd need to update RFC3262.

It's on my list of things to do Someday When I Have The Time.

>
> In addition, the scope of an option-tag can be a single transaction,  
> or wider.
>
> For example, just because I require 100rel in the initial INVITE, it  
> doesn't mean (at least that is my understanding) I have implicitly  
> also required it in each subsequent re-INVITE.
>
> The scope of Require:sec-agree, on the other hand, is for the whole  
> session.
>
> Is there ANY option-tag defined, where we specify that Require only  
> means must-support-but-is-not-required-to-use?

Path and Service-Route and answer-mode come to mind. I edited them . . .

>
> Anyway, if we move forward with options-ping, I DO think there  
> should be a way to identify the OPTIONS as a ping, so that the  
> receiver e.g. doens't need to include all supported capabilities etc  
> in the response - or expect them in the request.

Yes, that would be useful.

--
Dean

From HKaplan@acmepacket.com  Fri Apr 23 15:15:15 2010
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To: Paul Kyzivat <pkyzivat@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>
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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Paul Kyzivat
> Sent: Friday, April 23, 2010 1:28 PM
>=20
> Making a decision based on the form of the R-URI presumes that non-ping
> users of options never use an ip address as the R-URI. Requiring the
> R-URI to be ping@ip reserves a user part, and also is just a sleazy way
> of adding another indicator to the message to indicate its intended use.

Actually in a weird way I think doing a specific username may be the right =
thing to do.  Think about it from an email perspective.  When you check on =
the status and settings of an email list using SMTP, do you (a) send a new =
different SMTP message type, or (b) address a normal message to the actual =
logical administrative code for the list?  You send a normal message to use=
rlist-request@emailserver.com. =20

And logically the thing being "pinged" is a higher layer, because the respo=
nse provides an administrative status as well. (even though in code one wou=
ld probably do this at a low level)=20

Put it another way: in theory you could send a SUBSCRIBE to this logical ad=
min-UA in that proxy so that it Notifies you when it goes admin up/down.  T=
hat logical UA is a resource in the proxy, and could be addressed as such.=
=20

-hadriel

From pkyzivat@cisco.com  Fri Apr 23 15:45:01 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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I will grant you that having per-user policies for what to include in 
OPTIONS would be possible.

But using that approach here means reserving a specific username. Which 
one are we allowed to pick that won't impact somebody? (I expect there 
are people out there named "ping" who might not like having their user 
name preempted.)

AFAIK we have never preempted a user name, except for Anonymous, which 
is only preempted in the anonymous.invalid domain. I'd much rather see a 
special uri parameter to designate this behavior.

	Thanks,
	Paul

Hadriel Kaplan wrote:
> 
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf
>> Of Paul Kyzivat
>> Sent: Friday, April 23, 2010 1:28 PM
>>
>> Making a decision based on the form of the R-URI presumes that non-ping
>> users of options never use an ip address as the R-URI. Requiring the
>> R-URI to be ping@ip reserves a user part, and also is just a sleazy way
>> of adding another indicator to the message to indicate its intended use.
> 
> Actually in a weird way I think doing a specific username may be the right thing to do.  Think about it from an email perspective.  When you check on the status and settings of an email list using SMTP, do you (a) send a new different SMTP message type, or (b) address a normal message to the actual logical administrative code for the list?  You send a normal message to userlist-request@emailserver.com.  
> 
> And logically the thing being "pinged" is a higher layer, because the response provides an administrative status as well. (even though in code one would probably do this at a low level) 
> 
> Put it another way: in theory you could send a SUBSCRIBE to this logical admin-UA in that proxy so that it Notifies you when it goes admin up/down.  That logical UA is a resource in the proxy, and could be addressed as such. 
> 
> -hadriel
> 

From christer.holmberg@ericsson.com  Fri Apr 23 15:54:21 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Sat, 24 Apr 2010 00:54:07 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>, 'Michael, Miller' <mlm.michael.miller@gmail.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

>>>> One possibility would be to use an option tag: Required:ping. That
>>>> would ensure the other party knows this is a ping. (And of course if
>>>> that fails you could remove the tag, and realize you were not
>>>> getting a specialized ping implementation.)
>>>
>>>
>>>
>>> Nah, that just requires somebody to support the ping extension; it
>>> doesn't tell them to apply that extension to this OPTIONS request.
>>
>> I don't think you can make such general assumption. It needs to be
>> specified per option tag.
>>
>> For example, if you receive an INVITE with Require:100rel, you must
>> not only support the extension - you must also use it when sending
>> provisional responses (see RFC3262, section 3, paragraph 2).
>>
>
>We've had this discussion before: 100rel was badly written in this respect=
.
>
>For example, what do you do if you receive a request that has Require:100r=
el but not Supported: 100rel?

Assuming I support 100rel I would send my 18x responses reliably, because I=
 would assume, based on the text in 3262, that Require:100 implicitly indic=
ates that the UAC also supports it.

>>If you don't think is correct we'd need to update RFC3262.
>
>It's on my list of things to do Someday When I Have The Time.

So, you want to say that a UAS must not send reliable responses in case the=
 INVITE only contains Require:100rel, but not Supported:100rel? In that cas=
e I think "Someday" may have been a long time ago :)

And, I also think you have the same issue with the "preconditions" option-t=
ag. Are you going to update 3312/4032 also?

And, what about sec-agree?

>> In addition, the scope of an option-tag can be a single transaction,
>> or wider.
>>
>> For example, just because I require 100rel in the initial INVITE, it
>> doesn't mean (at least that is my understanding) I have implicitly
>> also required it in each subsequent re-INVITE.
>>
>> The scope of Require:sec-agree, on the other hand, is for the whole
>> session.
>>
>> Is there ANY option-tag defined, where we specify that Require only
>> means must-support-but-is-not-required-to-use?
>
>Path and Service-Route and answer-mode come to mind. I edited them . . .

I don't think we have an option-tag for Service-Route, do we?

And, as far as I remember, RFC 3327 doesn't say anything about the usage of=
 the "path" option-tag in Require.

Regards,

Christer=

From dean.willis@softarmor.com  Fri Apr 23 15:54:44 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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On Apr 23, 2010, at 5:44 PM, Paul Kyzivat wrote:
>
> I'd much rather see a special uri parameter to designate this  
> behavior.
>

That would be so cool!

If only we had a way to pass URI parameters across proxies that do RFC  
3261 contact-routing...

Which of corse if why I've been standing on that particular soapbox  
for ten years or so.

The best Idea I've heard along these lines is to stop REGISTERing  
Contact and start PUBLISHing a Route, as suggested by Adam early in  
MARTINI's life cycle. Perhaps we'll get there someday.

--
Dean


From HKaplan@acmepacket.com  Fri Apr 23 16:00:58 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Fri, 23 Apr 2010 19:00:44 -0400
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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"root" obviously.  ;)
But seriously, picking a crazy username that won't collide is not that hard=
.  The question is if it's the right thing to do.

Paul Kyzivat <pkyzivat@cisco.com> wrote:


I will grant you that having per-user policies for what to include in
OPTIONS would be possible.

But using that approach here means reserving a specific username. Which
one are we allowed to pick that won't impact somebody? (I expect there
are people out there named "ping" who might not like having their user
name preempted.)

AFAIK we have never preempted a user name, except for Anonymous, which
is only preempted in the anonymous.invalid domain. I'd much rather see a
special uri parameter to designate this behavior.

        Thanks,
        Paul

Hadriel Kaplan wrote:
>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Beha=
lf
>> Of Paul Kyzivat
>> Sent: Friday, April 23, 2010 1:28 PM
>>
>> Making a decision based on the form of the R-URI presumes that non-ping
>> users of options never use an ip address as the R-URI. Requiring the
>> R-URI to be ping@ip reserves a user part, and also is just a sleazy way
>> of adding another indicator to the message to indicate its intended use.
>
> Actually in a weird way I think doing a specific username may be the righ=
t thing to do.  Think about it from an email perspective.  When you check o=
n the status and settings of an email list using SMTP, do you (a) send a ne=
w different SMTP message type, or (b) address a normal message to the actua=
l logical administrative code for the list?  You send a normal message to u=
serlist-request@emailserver.com.
>
> And logically the thing being "pinged" is a higher layer, because the res=
ponse provides an administrative status as well. (even though in code one w=
ould probably do this at a low level)
>
> Put it another way: in theory you could send a SUBSCRIBE to this logical =
admin-UA in that proxy so that it Notifies you when it goes admin up/down. =
 That logical UA is a resource in the proxy, and could be addressed as such=
.
>
> -hadriel
>

From christer.holmberg@ericsson.com  Sat Apr 24 00:30:59 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Sat, 24 Apr 2010 09:28:28 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

>>One possibility would be to use an option tag: Required:ping. That
>>would ensure the other party knows this is a ping. (And of course if
>>that fails you could remove the tag, and realize you were not
>>getting a specialized ping implementation.)
>
>
>
>Nah, that just requires somebody to support the ping extension; it
>doesn't tell them to apply that extension to this OPTIONS request.

In this specific case, does it really matter?

As long as you only want SOME response, does it really matter? You will eit=
her get a 200 or a 420 response.

Regards,

Christer=

From sanjsinh@cisco.com  Sat Apr 24 11:23:02 2010
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From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, "Dean Willis" <dean.willis@softarmor.com>, "Paul Kyzivat (pkyzivat)" <pkyzivat@cisco.com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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It matters in the sense what the UAS returns in 200 OK response to
OPTIONS. If it supports that extension, it will return a success
response without it's full capabilities, as will be required with normal
OPTIONS processing.

Sanjay

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
Behalf Of
> Christer Holmberg
> Sent: Saturday, April 24, 2010 12:58 PM
> To: Dean Willis; Paul Kyzivat (pkyzivat)
> Cc: Paul E. Jones; sipcore@ietf.org; 'Michael Miller'
> Subject: Re: [sipcore] FW: I-D
Action:draft-jones-sip-options-ping-00.txt
>=20
> Hi,
>=20
> >>One possibility would be to use an option tag: Required:ping. That
> >>would ensure the other party knows this is a ping. (And of course if
> >>that fails you could remove the tag, and realize you were not
> >>getting a specialized ping implementation.)
> >
> >
> >
> >Nah, that just requires somebody to support the ping extension; it
> >doesn't tell them to apply that extension to this OPTIONS request.
>=20
> In this specific case, does it really matter?
>=20
> As long as you only want SOME response, does it really matter? You
will either get a
> 200 or a 420 response.
>=20
> Regards,
>=20
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From dean.willis@softarmor.com  Sat Apr 24 13:43:41 2010
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On Apr 23, 2010, at 5:54 PM, Christer Holmberg wrote:
>>
>> For example, what do you do if you receive a request that has  
>> Require:100rel but not Supported: 100rel?
>
> Assuming I support 100rel I would send my 18x responses reliably,  
> because I would assume, based on the text in 3262, that Require:100  
> implicitly indicates that the UAC also supports it.
>

So, 3262 has a bug in it. Putting an option tag in Require clearly  
doesn't men you support the feature, only that you require the UAS to  
do so.


>>> If you don't think is correct we'd need to update RFC3262.
>>
>> It's on my list of things to do Someday When I Have The Time.
>
> So, you want to say that a UAS must not send reliable responses in  
> case the INVITE only contains Require:100rel, but not Supported: 
> 100rel? In that case I think "Someday" may have been a long time  
> ago :)

I think at a minimum that 3262 should say that a UAC putting a  
Require: 100rel MUST also put Supported: 100rel


>
> And, I also think you have the same issue with the "preconditions"  
> option-tag. Are you going to update 3312/4032 also?
>

I suspect it needs it.

> And, what about sec-agree?

It's clearly got bugs.

>
>>> In addition, the scope of an option-tag can be a single transaction,
>>> or wider.
>>>
>>> For example, just because I require 100rel in the initial INVITE, it
>>> doesn't mean (at least that is my understanding) I have implicitly
>>> also required it in each subsequent re-INVITE.
>>>
>>> The scope of Require:sec-agree, on the other hand, is for the whole
>>> session.
>>>
>>> Is there ANY option-tag defined, where we specify that Require only
>>> means must-support-but-is-not-required-to-use?
>>
>> Path and Service-Route and answer-mode come to mind. I edited  
>> them . . .
>
> I don't think we have an option-tag for Service-Route, do we?
>
> And, as far as I remember, RFC 3327 doesn't say anything about the  
> usage of the "path" option-tag in Require.

Right. The UAS is free to use path if the UAC supports it. It doesn't  
decide whether or not to use the extension based on a Require.

--
Dean

From christer.holmberg@ericsson.com  Sun Apr 25 02:57:59 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>, Dean Willis <dean.willis@softarmor.com>, "Paul Kyzivat (pkyzivat)" <pkyzivat@cisco.com>
Date: Sun, 25 Apr 2010 11:55:12 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

>It matters in the sense what the UAS returns in 200 OK response to
>OPTIONS. If it supports that extension, it will return a success
>response without it's full capabilities, as will be required with normal
>OPTIONS processing.

But, from a ping perspective, why does it matter what the response code it?=
 Why does it matter whether the receiver supports the functionality or not?

Of course, if you also expect the receiver to send pings towards you, then =
you may need to know whether it supports the mechanism. But, if you are the=
 only one sending pings, all you want is A response, isn't it?

Regards,

Christer


> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
Behalf Of
> Christer Holmberg
> Sent: Saturday, April 24, 2010 12:58 PM
> To: Dean Willis; Paul Kyzivat (pkyzivat)
> Cc: Paul E. Jones; sipcore@ietf.org; 'Michael Miller'
> Subject: Re: [sipcore] FW: I-D
Action:draft-jones-sip-options-ping-00.txt
>
> Hi,
>
> >>One possibility would be to use an option tag: Required:ping. That
> >>would ensure the other party knows this is a ping. (And of course if
> >>that fails you could remove the tag, and realize you were not
> >>getting a specialized ping implementation.)
> >
> >
> >
> >Nah, that just requires somebody to support the ping extension; it
> >doesn't tell them to apply that extension to this OPTIONS request.
>
> In this specific case, does it really matter?
>
> As long as you only want SOME response, does it really matter? You
will either get a
> 200 or a 420 response.
>
> Regards,
>
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore=

From christer.holmberg@ericsson.com  Sun Apr 25 12:25:27 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Sun, 25 Apr 2010 21:25:10 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

>>> For example, what do you do if you receive a request that has
>>> Require:100rel but not Supported: 100rel?
>>
>> Assuming I support 100rel I would send my 18x responses reliably,
>> because I would assume, based on the text in 3262, that Require:100
>> implicitly indicates that the UAC also supports it.
>>
>
>So, 3262 has a bug in it. Putting an option tag in Require clearly doesn't=
 men you support the feature, only that you require the UAS to do so.

Upon what text do you base "clearly"?=20

If it's so clear, why have we ended up with a number of RFCs which would be=
 incorrect?

>>>> If you don't think is correct we'd need to update RFC3262.
>>>
>>>It's on my list of things to do Someday When I Have The Time.
>>
>>So, you want to say that a UAS must not send reliable responses in
>>case the INVITE only contains Require:100rel, but not Supported:
>>100rel? In that case I think "Someday" may have been a long time
>>ago :)
>
>I think at a minimum that 3262 should say that a UAC putting a Require: 10=
0rel MUST also put Supported: 100rel

If we would be doing 3262 today I would not necessarily disagree with you. =
Today, I just think you're going to end up with backward compability issues=
.

>>>And, I also think you have the same issue with the "preconditions" optio=
n-tag. Are you going to update 3312/4032 also?
>>
>>I suspect it needs it.
>>
>>And, what about sec-agree?
>>
>It's clearly got bugs.

I love this protocol...

Also, in some cases the Require header in a *response* seems to have differ=
ent meanings.=20

For example, "Require: outbound" in a response seems to indicate that the r=
egistrar has done somthing, while "timer" seems to require the UAC to do so=
mething (ie not only require to support some extension).

So, if Require only means "you must support", why would someone send 'Requi=
re: x' in a response if the associated request already contained 'Supported=
: x'? The UAC has already indicated support, so there is no need to say you=
-must-support...

Regards,

Christer=

From sanjsinh@cisco.com  Sun Apr 25 21:00:38 2010
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From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, "Dean Willis" <dean.willis@softarmor.com>, "Paul Kyzivat (pkyzivat)" <pkyzivat@cisco.com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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It is true that all the receiver wants is some response. And for that,
the receiving node does not even have to support OPTIONS. If it does
not, then it will send 405 response. That is enough for the sender to
know that the ping was successful.
But I think that we are talking about case where 200 OK is considered as
successful ping. For that case, Dean was suggesting an options tag for
the receiver to sends a ping 200 OK response as compared to 200 OK
response for an OPTIONS, which is to query capabilities.

Maybe all confusion is due to overloading of OPTIONS.

Thanks
Sanjay

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Sunday, April 25, 2010 3:25 PM
> To: Sanjay Sinha (sanjsinh); Dean Willis; Paul Kyzivat (pkyzivat)
> Cc: Paul E. Jones; sipcore@ietf.org; Michael Miller
> Subject: RE: [sipcore] FW: I-D
Action:draft-jones-sip-options-ping-00.txt
>=20
> Hi,
>=20
> >It matters in the sense what the UAS returns in 200 OK response to
> >OPTIONS. If it supports that extension, it will return a success
> >response without it's full capabilities, as will be required with
normal
> >OPTIONS processing.
>=20
> But, from a ping perspective, why does it matter what the response
code it? Why does
> it matter whether the receiver supports the functionality or not?
>=20
> Of course, if you also expect the receiver to send pings towards you,
then you may
> need to know whether it supports the mechanism. But, if you are the
only one sending
> pings, all you want is A response, isn't it?
>
>=20
> Regards,
>=20
> Christer
>=20
>=20
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of
> > Christer Holmberg
> > Sent: Saturday, April 24, 2010 12:58 PM
> > To: Dean Willis; Paul Kyzivat (pkyzivat)
> > Cc: Paul E. Jones; sipcore@ietf.org; 'Michael Miller'
> > Subject: Re: [sipcore] FW: I-D
> Action:draft-jones-sip-options-ping-00.txt
> >
> > Hi,
> >
> > >>One possibility would be to use an option tag: Required:ping. That
> > >>would ensure the other party knows this is a ping. (And of course
if
> > >>that fails you could remove the tag, and realize you were not
> > >>getting a specialized ping implementation.)
> > >
> > >
> > >
> > >Nah, that just requires somebody to support the ping extension; it
> > >doesn't tell them to apply that extension to this OPTIONS request.
> >
> > In this specific case, does it really matter?
> >
> > As long as you only want SOME response, does it really matter? You
> will either get a
> > 200 or a 420 response.
> >
> > Regards,
> >
> > Christer
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore

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Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, "Dean Willis" <dean.willis@softarmor.com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Without "Require" in the response, how does the UAC know that the UAS
supports that extension, if the UAC had added a Supported header in the
request?

Thanks
Sanjay

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
Behalf Of
> Christer Holmberg
> Sent: Monday, April 26, 2010 12:55 AM
> To: Dean Willis
> Cc: Paul E. Jones; sipcore@ietf.org; Michael@core3.amsl.com; Miller'
> Subject: Re: [sipcore] FW: I-D
Action:draft-jones-sip-options-ping-00.txt
>=20
> Hi,
>=20
> >>> For example, what do you do if you receive a request that has
> >>> Require:100rel but not Supported: 100rel?
> >>
> >> Assuming I support 100rel I would send my 18x responses reliably,
> >> because I would assume, based on the text in 3262, that Require:100
> >> implicitly indicates that the UAC also supports it.
> >>
> >
> >So, 3262 has a bug in it. Putting an option tag in Require clearly
doesn't men you
> support the feature, only that you require the UAS to do so.
>=20
> Upon what text do you base "clearly"?
>=20
> If it's so clear, why have we ended up with a number of RFCs which
would be
> incorrect?
>=20
> >>>> If you don't think is correct we'd need to update RFC3262.
> >>>
> >>>It's on my list of things to do Someday When I Have The Time.
> >>
> >>So, you want to say that a UAS must not send reliable responses in
> >>case the INVITE only contains Require:100rel, but not Supported:
> >>100rel? In that case I think "Someday" may have been a long time
> >>ago :)
> >
> >I think at a minimum that 3262 should say that a UAC putting a
Require: 100rel
> MUST also put Supported: 100rel
>=20
> If we would be doing 3262 today I would not necessarily disagree with
you. Today, I
> just think you're going to end up with backward compability issues.
>=20
> >>>And, I also think you have the same issue with the "preconditions"
option-tag. Are
> you going to update 3312/4032 also?
> >>
> >>I suspect it needs it.
> >>
> >>And, what about sec-agree?
> >>
> >It's clearly got bugs.
>=20
> I love this protocol...
>=20
> Also, in some cases the Require header in a *response* seems to have
different
> meanings.
>=20
> For example, "Require: outbound" in a response seems to indicate that
the registrar has
> done somthing, while "timer" seems to require the UAC to do something
(ie not only
> require to support some extension).
>=20
> So, if Require only means "you must support", why would someone send
'Require: x'
> in a response if the associated request already contained 'Supported:
x'? The UAC has
> already indicated support, so there is no need to say
you-must-support...
>=20
> Regards,
>=20
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From christer.holmberg@ericsson.com  Mon Apr 26 00:02:23 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>, Dean Willis <dean.willis@softarmor.com>
Date: Mon, 26 Apr 2010 09:02:05 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,=20

>Without "Require" in the response, how does the UAC know that the UAS supp=
orts that extension, if the UAC had added a=20
>Supported header in the request?

According to Dean "Require" doesn't say anything about what the UAS support=
s, or doesn't support. It only says what the receiver must support.

Regards,

Christer


> > -----Original Message-----
> > From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of
> > Christer Holmberg
> > Sent: Monday, April 26, 2010 12:55 AM
> > To: Dean Willis
> > Cc: Paul E. Jones; sipcore@ietf.org; Michael@core3.amsl.com; Miller'
> > Subject: Re: [sipcore] FW: I-D
> Action:draft-jones-sip-options-ping-00.txt
> >=20
> > Hi,
> >=20
> > >>> For example, what do you do if you receive a request that has=20
> > >>> Require:100rel but not Supported: 100rel?
> > >>
> > >> Assuming I support 100rel I would send my 18x responses=20
> reliably,=20
> > >> because I would assume, based on the text in 3262, that=20
> Require:100=20
> > >> implicitly indicates that the UAC also supports it.
> > >>
> > >
> > >So, 3262 has a bug in it. Putting an option tag in Require clearly
> doesn't men you
> > support the feature, only that you require the UAS to do so.
> >=20
> > Upon what text do you base "clearly"?
> >=20
> > If it's so clear, why have we ended up with a number of RFCs which
> would be
> > incorrect?
> >=20
> > >>>> If you don't think is correct we'd need to update RFC3262.
> > >>>
> > >>>It's on my list of things to do Someday When I Have The Time.
> > >>
> > >>So, you want to say that a UAS must not send reliable=20
> responses in=20
> > >>case the INVITE only contains Require:100rel, but not Supported:
> > >>100rel? In that case I think "Someday" may have been a=20
> long time ago=20
> > >>:)
> > >
> > >I think at a minimum that 3262 should say that a UAC putting a
> Require: 100rel
> > MUST also put Supported: 100rel
> >=20
> > If we would be doing 3262 today I would not necessarily=20
> disagree with
> you. Today, I
> > just think you're going to end up with backward compability issues.
> >=20
> > >>>And, I also think you have the same issue with the=20
> "preconditions"
> option-tag. Are
> > you going to update 3312/4032 also?
> > >>
> > >>I suspect it needs it.
> > >>
> > >>And, what about sec-agree?
> > >>
> > >It's clearly got bugs.
> >=20
> > I love this protocol...
> >=20
> > Also, in some cases the Require header in a *response* seems to have
> different
> > meanings.
> >=20
> > For example, "Require: outbound" in a response seems to=20
> indicate that
> the registrar has
> > done somthing, while "timer" seems to require the UAC to do=20
> something
> (ie not only
> > require to support some extension).
> >=20
> > So, if Require only means "you must support", why would someone send
> 'Require: x'
> > in a response if the associated request already contained=20
> 'Supported:
> x'? The UAC has
> > already indicated support, so there is no need to say
> you-must-support...
> >=20
> > Regards,
> >=20
> > Christer
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> =

From dean.willis@softarmor.com  Mon Apr 26 02:13:47 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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On Apr 25, 2010, at 4:55 AM, Christer Holmberg wrote:

> Hi,
>
>> It matters in the sense what the UAS returns in 200 OK response to
>> OPTIONS. If it supports that extension, it will return a success
>> response without it's full capabilities, as will be required with  
>> normal
>> OPTIONS processing.
>
> But, from a ping perspective, why does it matter what the response  
> code it? Why does it matter whether the receiver supports the  
> functionality or not?
>
> Of course, if you also expect the receiver to send pings towards  
> you, then you may need to know whether it supports the mechanism.  
> But, if you are the only one sending pings, all you want is A  
> response, isn't it?
>

Then you don't need an option tag and don't need to be able to Require  
the extension. It's just an optimization parameter; if the UAS sees a  
recognizable "ping" flag, then it can return an optimized options-free  
response. If it doesn't understand the flag, it returns a conventional  
response.

Sounds like a great candidate for a new header field. Note that I've  
generally believed that header fields should be easy to register,  
given that they're our only really way to pass parameters since we  
broke URI-parameters with contact-routing.

Maybe we can define an "OPtions-Mode" header field, and register the  
value "ping" here -- and add additional Options-Mode values here in  
the future if needed.

--
Dean

From dean.willis@softarmor.com  Mon Apr 26 02:34:06 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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On Apr 25, 2010, at 2:25 PM, Christer Holmberg wrote:

> Hi,
>
>>>> For example, what do you do if you receive a request that has
>>>> Require:100rel but not Supported: 100rel?
>>>
>>> Assuming I support 100rel I would send my 18x responses reliably,
>>> because I would assume, based on the text in 3262, that Require:100
>>> implicitly indicates that the UAC also supports it.
>>>
>>
>> So, 3262 has a bug in it. Putting an option tag in Require clearly  
>> doesn't men you support the feature, only that you require the UAS  
>> to do so.
>
> Upon what text do you base "clearly"?

RFC 3261: 20.32 Require says (verbatim):
"The Require header field is used by UACs to tell UASs about options  
that the UAC expects the UAS to support in order to process the  
request. Although an optional header field, the Require MUST NOT be  
ignored if it is present. The Require header field contains a list of  
option tags, described in Section 19.2. Each option tag defines a SIP  
extension that MUST be understood to process the request. Frequently,  
this is used to indicate that a specific set of extension header  
fields need to be understood. A UAC compliant to this specification  
MUST only include option tags corresponding to standards-track RFCs."

The above clearly indicates that the usual use of the option-tag in  
Require is to indicate that there is other data in the request,  
typically header fields, which the UAS MUST be able to understand in  
order to process the request. It does NOT say that the presence of the  
Require changes the behavior of the UAS in any other way; it's the  
presence of those header fields (or other extension flags) in the  
request that changes the behavior. All the Require does is cause the  
UAS to reject the request if it doesn't understand that other data.  
This changes the default behavior of a UAS, which is to silently  
ignore "other data" that it does not understand. Note that having a  
Require for a new request type is rather useless, as we're less  
liberal-in-what-we -accept with request types and reject unknowns  
anyhow.
In other words, un-Required extensions appearing in a known request  
type are ignored if not understood, whereas Required extensions that  
are not understood cause the request to be rejected.
Note also that the prohibition of non-standards-track option-tags is  
spelled out in RFC 3261.
>
> If it's so clear, why have we ended up with a number of RFCs which  
> would be incorrect?
>

It's an overly complicated design for our simple minds? I didn't say I  
LIKE the way this works in RFC 3261, just that I mostly (ok, sortof)  
understand it.

>>>
>> It's clearly got bugs.
>
> I love this protocol...
>

Yes, it gives us so much to work on. Much better than knitting or  
jigsaw puzzles on a cold winter night!

> Also, in some cases the Require header in a *response* seems to have  
> different meanings.

Very true. At the moment, I can't say where the meaning of Require in  
a response comes from. RFC 3261 does not, AFAIK, say anything about  
it. We seem to have invented it on the fly without formal  
specification, using an error in RFC 3261's table 2/3 to justify the  
approach. Perhaps a more formal specification would be a good idea.

>
> For example, "Require: outbound" in a response seems to indicate  
> that the registrar has done somthing, while "timer" seems to require  
> the UAC to do something (ie not only require to support some  
> extension).
>
> So, if Require only means "you must support", why would someone send  
> 'Require: x' in a response if the associated request already  
> contained 'Supported: x'? The UAC has already indicated support, so  
> there is no need to say you-must-support...

I dunno. Maybe somebody wrote a specification based on not  
understanding Require in the request and we were too tired of arguing  
about it to stop the publication?

--
Dean

From dean.willis@softarmor.com  Mon Apr 26 02:40:26 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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On Apr 26, 2010, at 2:02 AM, Christer Holmberg wrote:

>
> Hi,
>
>> Without "Require" in the response, how does the UAC know that the  
>> UAS supports that extension, if the UAC had added a
>> Supported header in the request?
>
> According to Dean "Require" doesn't say anything about what the UAS  
> supports, or doesn't support. It only says what the receiver must  
> support.

Not quite. The UAS is (by definition) the receiver of any request, and  
the Require says that if the UAS does not support the required option,  
then it must reject the request.

Other than that, the Require doesn't invoke any specific behavior in  
the UAS.

So for example a INVITE with a "Require: explodemyphone" doesn't make  
the phone blow up, but would cause the phone to reject the request if  
it doesn't know how to blow up. It'd be some other header field, like  
"Explodemyphone:withsmokeandfire". In the absence of a Require, a  
phone that didn't know how to explode would just ignore that part of  
the request and alert in the usual fashion.

Also, an INVITE with a "Require:explodemyphone" but that lacked an  
"Explodemyphone" header field would be accepted by a phone that knows  
how to explode, but said phone probably wouldn't explode as a result  
(especially not with both smoke and fire!)

--
Dean


From christer.holmberg@ericsson.com  Mon Apr 26 03:43:14 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Mon, 26 Apr 2010 12:42:55 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,=20

>>>So, 3262 has a bug in it. Putting an option tag in Require clearly=20
>>>doesn't men you support the feature, only that you require=20
>>>the UAS to do so.
>>
>>Upon what text do you base "clearly"?
>=20
>RFC 3261: 20.32 Require says (verbatim):
>"The Require header field is used by UACs to tell UASs about=20
>options that the UAC expects the UAS to support in order to=20
>process the request. Although an optional header field, the=20
>Require MUST NOT be ignored if it is present. The Require=20
>header field contains a list of option tags, described in=20
>Section 19.2. Each option tag defines a SIP extension that=20
>MUST be understood to process the request. Frequently, this=20
>is used to indicate that a specific set of extension header=20
>fields need to be understood. A UAC compliant to this=20
>specification MUST only include option tags corresponding to=20
>standards-track RFCs."
>=20
>The above clearly indicates that the usual use of the=20
>option-tag in Require is to indicate that there is other data=20
>in the request, typically header fields, which the UAS MUST=20
>be able to understand in order to process the request. It=20
>does NOT say that the presence of the Require changes the=20
>behavior of the UAS in any other way; it's the presence of=20
>those header fields (or other extension flags) in the request=20
>that changes the behavior. All the Require does is cause the=20
>UAS to reject the request if it doesn't understand that other data. =20
>This changes the default behavior of a UAS, which is to=20
>silently ignore "other data" that it does not understand.=20
>Note that having a Require for a new request type is rather=20
>useless, as we're less liberal-in-what-we -accept with=20
>request types and reject unknowns anyhow.
>In other words, un-Required extensions appearing in a known=20
>request type are ignored if not understood, whereas Required=20
>extensions that are not understood cause the request to be rejected.
>Note also that the prohibition of non-standards-track=20
>option-tags is spelled out in RFC 3261.

Section 8.1.1.9 says:

"If the UAC wishes to insist that a UAS understand an extension that
the UAC will apply to the request in order to process the request, it
MUST insert a Require header field into the request listing the
option tag for that extension."

I think the "that the UAC will apply" could be read as indicating that the =
UAC also supports the extension.

>>Also, in some cases the Require header in a *response* seems to have =20
>>different meanings.
>=20
>Very true. At the moment, I can't say where the meaning of=20
>Require in a response comes from. RFC 3261 does not, AFAIK, say anything a=
bout =20
>it. We seem to have invented it on the fly without formal =20
>specification, using an error in RFC 3261's table 2/3 to justify the =20
>approach. Perhaps a more formal specification would be a good idea.
>=20
>>For example, "Require: outbound" in a response seems to indicate =20
>>that the registrar has done somthing, while "timer" seems=20
>>to require the UAC to do something (ie not only require to support some =
=20
>>extension).
>>=20
>>So, if Require only means "you must support", why would someone send =20
>>'Require: x' in a response if the associated request already =20
>>contained 'Supported: x'? The UAC has already indicated support, so =20
>>there is no need to say you-must-support...
>=20
>I dunno. Maybe somebody wrote a specification based on not =20
>understanding Require in the request and we were too tired of=20
>arguing about it to stop the publication?

Well, it's not only one specification...

Regards,

Christer

From christer.holmberg@ericsson.com  Mon Apr 26 03:45:34 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Mon, 26 Apr 2010 12:45:18 +0200
Thread-Topic: Meaning of Require in response [was: I-D Action:draft-jones-sip-options-ping-00.txt]
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Subject: [sipcore] Meaning of Require in response [was: I-D Action:draft-jones-sip-options-ping-00.txt]
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Hi,=20

>>>Without "Require" in the response, how does the UAC know=20
>>>that the UAS supports that extension, if the UAC had added a Supported=20
>>>header in the request?
>>
>>According to Dean "Require" doesn't say anything about what the UAS=20
>>supports, or doesn't support. It only says what the receiver must=20
>>support.
>=20
>Not quite. The UAS is (by definition) the receiver of any=20
>request, and the Require says that if the UAS does not=20
>support the required option, then it must reject the request.

Note that the text above was about Require in a response (in which case the=
 UAC is the receiver).

Regards,

Christer

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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: [sipcore] The semantics of Require.
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If this discussion is to continue, can we at least have a relevant 
subject rather than hijacking the discussion options ping?

I agree in principle with Dean's point. But from a practical perspective 
I think we just acknowledge that the definition of certain options added 
on additional semantics. I think we should just grandfather that but 
refrain from making the same mistake in the future.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi, 
> 
>>>> So, 3262 has a bug in it. Putting an option tag in Require clearly 
>>>> doesn't men you support the feature, only that you require 
>>>> the UAS to do so.
>>> Upon what text do you base "clearly"?
>> RFC 3261: 20.32 Require says (verbatim):
>> "The Require header field is used by UACs to tell UASs about 
>> options that the UAC expects the UAS to support in order to 
>> process the request. Although an optional header field, the 
>> Require MUST NOT be ignored if it is present. The Require 
>> header field contains a list of option tags, described in 
>> Section 19.2. Each option tag defines a SIP extension that 
>> MUST be understood to process the request. Frequently, this 
>> is used to indicate that a specific set of extension header 
>> fields need to be understood. A UAC compliant to this 
>> specification MUST only include option tags corresponding to 
>> standards-track RFCs."
>>
>> The above clearly indicates that the usual use of the 
>> option-tag in Require is to indicate that there is other data 
>> in the request, typically header fields, which the UAS MUST 
>> be able to understand in order to process the request. It 
>> does NOT say that the presence of the Require changes the 
>> behavior of the UAS in any other way; it's the presence of 
>> those header fields (or other extension flags) in the request 
>> that changes the behavior. All the Require does is cause the 
>> UAS to reject the request if it doesn't understand that other data.  
>> This changes the default behavior of a UAS, which is to 
>> silently ignore "other data" that it does not understand. 
>> Note that having a Require for a new request type is rather 
>> useless, as we're less liberal-in-what-we -accept with 
>> request types and reject unknowns anyhow.
>> In other words, un-Required extensions appearing in a known 
>> request type are ignored if not understood, whereas Required 
>> extensions that are not understood cause the request to be rejected.
>> Note also that the prohibition of non-standards-track 
>> option-tags is spelled out in RFC 3261.
> 
> Section 8.1.1.9 says:
> 
> "If the UAC wishes to insist that a UAS understand an extension that
> the UAC will apply to the request in order to process the request, it
> MUST insert a Require header field into the request listing the
> option tag for that extension."
> 
> I think the "that the UAC will apply" could be read as indicating that the UAC also supports the extension.
> 
>>> Also, in some cases the Require header in a *response* seems to have  
>>> different meanings.
>> Very true. At the moment, I can't say where the meaning of 
>> Require in a response comes from. RFC 3261 does not, AFAIK, say anything about  
>> it. We seem to have invented it on the fly without formal  
>> specification, using an error in RFC 3261's table 2/3 to justify the  
>> approach. Perhaps a more formal specification would be a good idea.
>>
>>> For example, "Require: outbound" in a response seems to indicate  
>>> that the registrar has done somthing, while "timer" seems 
>>> to require the UAC to do something (ie not only require to support some  
>>> extension).
>>>
>>> So, if Require only means "you must support", why would someone send  
>>> 'Require: x' in a response if the associated request already  
>>> contained 'Supported: x'? The UAC has already indicated support, so  
>>> there is no need to say you-must-support...
>> I dunno. Maybe somebody wrote a specification based on not  
>> understanding Require in the request and we were too tired of 
>> arguing about it to stop the publication?
> 
> Well, it's not only one specification...
> 
> Regards,
> 
> Christer
> 

From pkyzivat@cisco.com  Mon Apr 26 05:52:35 2010
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>, 'Michael Miller' <mlm.michael.miller@gmail.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Dean,

The loss of parameters on retargeting is a valid issue in general, and 
may be a reason not to use a parameter in this case. But it isn't a 
problem for the use case contemplated because that case doesn't expect 
there to be retargeting.

But haven't we decided to solve the uri parameter problem with H-I?

	Thanks,
	Paul

Dean Willis wrote:
> 
> On Apr 23, 2010, at 5:44 PM, Paul Kyzivat wrote:
>>
>> I'd much rather see a special uri parameter to designate this behavior.
>>
> 
> That would be so cool!
> 
> If only we had a way to pass URI parameters across proxies that do RFC 
> 3261 contact-routing...
> 
> Which of corse if why I've been standing on that particular soapbox for 
> ten years or so.
> 
> The best Idea I've heard along these lines is to stop REGISTERing 
> Contact and start PUBLISHing a Route, as suggested by Adam early in 
> MARTINI's life cycle. Perhaps we'll get there someday.
> 
> -- 
> Dean
> 
> 

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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hadriel Kaplan wrote:
> "root" obviously.  ;)
> But seriously, picking a crazy username that won't collide is not that hard.  The question is if it's the right thing to do.

I don't believe in reserving names from other people's namespaces after 
the fact, on general principles. You cannot know how they use their 
namespace. You can reduce the probability of collision by making the 
value very obscure, but then it is very obscure.

So IMO it is the wrong thing to do.

	Thanks,
	Paul

> Paul Kyzivat <pkyzivat@cisco.com> wrote:
> 
> 
> I will grant you that having per-user policies for what to include in
> OPTIONS would be possible.
> 
> But using that approach here means reserving a specific username. Which
> one are we allowed to pick that won't impact somebody? (I expect there
> are people out there named "ping" who might not like having their user
> name preempted.)
> 
> AFAIK we have never preempted a user name, except for Anonymous, which
> is only preempted in the anonymous.invalid domain. I'd much rather see a
> special uri parameter to designate this behavior.
> 
>         Thanks,
>         Paul
> 
> Hadriel Kaplan wrote:
>>> -----Original Message-----
>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf
>>> Of Paul Kyzivat
>>> Sent: Friday, April 23, 2010 1:28 PM
>>>
>>> Making a decision based on the form of the R-URI presumes that non-ping
>>> users of options never use an ip address as the R-URI. Requiring the
>>> R-URI to be ping@ip reserves a user part, and also is just a sleazy way
>>> of adding another indicator to the message to indicate its intended use.
>> Actually in a weird way I think doing a specific username may be the right thing to do.  Think about it from an email perspective.  When you check on the status and settings of an email list using SMTP, do you (a) send a new different SMTP message type, or (b) address a normal message to the actual logical administrative code for the list?  You send a normal message to userlist-request@emailserver.com.
>>
>> And logically the thing being "pinged" is a higher layer, because the response provides an administrative status as well. (even though in code one would probably do this at a low level)
>>
>> Put it another way: in theory you could send a SUBSCRIBE to this logical admin-UA in that proxy so that it Notifies you when it goes admin up/down.  That logical UA is a resource in the proxy, and could be addressed as such.
>>
>> -hadriel
>>
> 

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To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 26 Apr 2010 12:59:39 -0400
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Monday, April 26, 2010 8:56 AM
>=20
> Hadriel Kaplan wrote:
> > "root" obviously.  ;)
> > But seriously, picking a crazy username that won't collide is not that
> hard.  The question is if it's the right thing to do.
>=20
> I don't believe in reserving names from other people's namespaces after
> the fact, on general principles. You cannot know how they use their
> namespace. You can reduce the probability of collision by making the
> value very obscure, but then it is very obscure.
>=20
> So IMO it is the wrong thing to do.

So you think it's the wrong thing to do NOT because indicating the specific=
 resource being targeted is the wrong thing, but rather that we can't safel=
y pick a username??=20

There are solutions to that, for example:
sip:80737871;phone-context=3Diana.org@proxy.com;user=3Dphone
or this:
urn:admin:ping

But we could also bite the bullet and define a namespace concept for these =
types of things, because ping is not the first or last time we'll need it. =
 For example a SUBSCRIBE to overload info, if we go that way for overload, =
could use such a target resource identifier concept.  As could draft-ietf-s=
ipping-policy-package.

-hadriel

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Dean Willis <dean.willis@softarmor.com>
Date: Mon, 26 Apr 2010 13:19:37 -0400
Thread-Topic: Semantics of option-tags (was:draft-jones-sip-options-ping-00.txt)
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Subject: [sipcore] Semantics of option-tags (was:draft-jones-sip-options-ping-00.txt)
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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Christer Holmberg
> Sent: Monday, April 26, 2010 6:43 AM
>=20
> Section 8.1.1.9 says:
>=20
> "If the UAC wishes to insist that a UAS understand an extension that
> the UAC will apply to the request in order to process the request, it
> MUST insert a Require header field into the request listing the
> option tag for that extension."
>=20
> I think the "that the UAC will apply" could be read as indicating that th=
e
> UAC also supports the extension.

Dean's point is that the Require does indeed mean the UA must support the e=
xtension - but it does not mean it is *using* the extension nor even demand=
ing the extension be used, for this particular transaction/dialog/whatever.=
  It's the header/param which indicates the actual usage, and the option-ta=
g just says "you must understand the header or param".  That way the proces=
sing of the Require/Supported header fields can be handled separately from =
whatever logic/layer handles the extension logic itself.

We've had this debate before. :)
We all agreed there were RFCs which used it to mean both "I understand" and=
 "I am using" the extension, and we wouldn't deprecate/change them.  But I'=
m not sure we ever agreed on what we'd do going forward.

-hadriel

From pkyzivat@cisco.com  Mon Apr 26 10:21:21 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hadriel Kaplan wrote:
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Monday, April 26, 2010 8:56 AM
>>
>> Hadriel Kaplan wrote:
>>> "root" obviously.  ;)
>>> But seriously, picking a crazy username that won't collide is not that
>> hard.  The question is if it's the right thing to do.
>>
>> I don't believe in reserving names from other people's namespaces after
>> the fact, on general principles. You cannot know how they use their
>> namespace. You can reduce the probability of collision by making the
>> value very obscure, but then it is very obscure.
>>
>> So IMO it is the wrong thing to do.
> 
> So you think it's the wrong thing to do NOT because indicating the specific resource being targeted is the wrong thing, but rather that we can't safely pick a username?? 

Yes. I have no problem, in principle, with having a class of "standard" 
resource names, as long as we came up with an acceptable way to define 
them without treading on anyone's toes.

> There are solutions to that, for example:
> sip:80737871;phone-context=iana.org@proxy.com;user=phone

Yeah, that could work, but its really ugly.

> or this:
> urn:admin:ping

That could work too, but then you couldn't use the R-URI for routing, so 
you would have to include a Route header to get it where you want it.
Still, that might work.

It could even fit within the existing service URN scheme.

> But we could also bite the bullet and define a namespace concept for these types of things, because ping is not the first or last time we'll need it.  For example a SUBSCRIBE to overload info, if we go that way for overload, could use such a target resource identifier concept.  As could draft-ietf-sipping-policy-package.



What did you have in mind? another URI parameter? E.g.

	sip:ping@bar.com;namespace=sipstuff

That raises all sorts of ugly issues. I like the URN better. The URN 
mechanism defines namespaces. It won't collide with anything, and those 
that don't support it will (hopefully) just return 404 or 416. Then you 
can just be happy with the error, or simply fall back to a "normal" OPTIONS.

But there are no doubt many other ways to skin this cat.

	Thanks,
	Paul

From dean.willis@softarmor.com  Mon Apr 26 10:23:21 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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On Apr 26, 2010, at 7:52 AM, Paul Kyzivat wrote:

> Dean,
>
> The loss of parameters on retargeting is a valid issue in general,  
> and may be a reason not to use a parameter in this case. But it  
> isn't a problem for the use case contemplated because that case  
> doesn't expect there to be retargeting.

How does it know?

>
> But haven't we decided to solve the uri parameter problem with H-I?

We keep saying that. I haven't seen it work yet. I retain the opinion  
that it's a bad solution to the problem.

--
Dean


From pkyzivat@cisco.com  Mon Apr 26 10:51:43 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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chair hat off:

Dean Willis wrote:
> 
> On Apr 26, 2010, at 7:52 AM, Paul Kyzivat wrote:
> 
>> Dean,
>>
>> The loss of parameters on retargeting is a valid issue in general, and 
>> may be a reason not to use a parameter in this case. But it isn't a 
>> problem for the use case contemplated because that case doesn't expect 
>> there to be retargeting.
> 
> How does it know?

I should let him answer.

My impression is either:
- Max-Forwards:1
or
- R-URI is ip address with no user part, and an *expectation* that
   this will not be redirected, or that if it is, then it isn't a
   use case of interest.

>> But haven't we decided to solve the uri parameter problem with H-I?
> 
> We keep saying that. I haven't seen it work yet. I retain the opinion 
> that it's a bad solution to the problem.

Well, I don't like it either. Yet that *seems* to be the consensus.

	Thanks,
	Paul

From christer.holmberg@ericsson.com  Mon Apr 26 10:53:37 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Hadriel Kaplan' <HKaplan@acmepacket.com>, Dean Willis <dean.willis@softarmor.com>
Date: Mon, 26 Apr 2010 19:53:11 +0200
Thread-Topic: Semantics of option-tags (was:draft-jones-sip-options-ping-00.txt)
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Cc: Miller' <mlm.michael.miller@gmail.com>, "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>, "'Michael@core3.amsl.com" <'Michael@core3.amsl.com>
Subject: Re: [sipcore] Semantics of option-tags (was:draft-jones-sip-options-ping-00.txt)
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Hi,=20

>>Section 8.1.1.9 says:
>>=20
>>"If the UAC wishes to insist that a UAS understand an extension that=20
>>the UAC will apply to the request in order to process the request, it=20
>>MUST insert a Require header field into the request listing the option=20
>>tag for that extension."
>>=20
>>I think the "that the UAC will apply" could be read as indicating that=20
>>the UAC also supports the extension.
>
>Dean's point is that the Require does indeed mean the UA must support the =
extension - but it does not mean it is *using* the extension nor even deman=
ding the extension be used, for this particular=20
>transaction/dialog/whatever.=20

Yes, I know.

>It's the header/param which indicates the actual usage, and the option-tag=
 just says "you must understand the header or param".  That way the process=
ing of the Require/Supported header fields can be=20
>handled separately from whatever logic/layer handles the extension logic i=
tself.

So, how does this work for 100rel? The INVITE request does not contain any =
of the extension specific header fields (RSeq and RAck). They will be used =
when the UAS starts using the extension.

Yes, I know, you say we don't deprcate/change 100rel, but what if we have s=
imilar extensions in the future, where the option-tag is the only thing you=
 initially send?

>We've had this debate before. :)
>
>We all agreed there were RFCs which used it to mean both "I understand" an=
d "I am using" the extension, and we wouldn't deprecate/change them.
>
>But I'm not sure we ever agreed on what we'd do going forward.

It would be really good if we could document the outcome of such debates, r=
ather than leaving them in an e-mail somewhere...

Regards,

Christer

From HKaplan@acmepacket.com  Mon Apr 26 11:03:03 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Date: Mon, 26 Apr 2010 14:02:48 -0400
Thread-Topic: Semantics of option-tags (was:draft-jones-sip-options-ping-00.txt)
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Subject: Re: [sipcore] Semantics of option-tags (was:draft-jones-sip-options-ping-00.txt)
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Monday, April 26, 2010 1:53 PM
>=20
> So, how does this work for 100rel? The INVITE request does not contain an=
y
> of the extension specific header fields (RSeq and RAck). They will be use=
d
> when the UAS starts using the extension.
>=20
> Yes, I know, you say we don't deprcate/change 100rel, but what if we have
> similar extensions in the future, where the option-tag is the only thing
> you initially send?

We would make sure they don't just send an option-tag only.  In some cases =
this would just mean a placebo header gets defined and used, like for RFC 4=
488 norefersub where a "Refer-Sub: false" is used.

-hadriel

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Cc: Miller' <mlm.michael.miller@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>, "'Michael@core3.amsl.com" <'Michael@core3.amsl.com>, "Paul E. Jones" <paulej@packetizer.com>
Subject: Re: [sipcore] Semantics of option-tags
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Christer Holmberg wrote:
> Hi, 
> 
>>> Section 8.1.1.9 says:
>>>
>>> "If the UAC wishes to insist that a UAS understand an extension that 
>>> the UAC will apply to the request in order to process the request, it 
>>> MUST insert a Require header field into the request listing the option 
>>> tag for that extension."
>>>
>>> I think the "that the UAC will apply" could be read as indicating that 
>>> the UAC also supports the extension.
>> Dean's point is that the Require does indeed mean the UA must support the extension - but it does not mean it is *using* the extension nor even demanding the extension be used, for this particular 
>> transaction/dialog/whatever. 
> 
> Yes, I know.
> 
>> It's the header/param which indicates the actual usage, and the option-tag just says "you must understand the header or param".  That way the processing of the Require/Supported header fields can be 
>> handled separately from whatever logic/layer handles the extension logic itself.
> 
> So, how does this work for 100rel? The INVITE request does not contain any of the extension specific header fields (RSeq and RAck). They will be used when the UAS starts using the extension.

100rel botched it, so now its grandfathered.

> Yes, I know, you say we don't deprcate/change 100rel, but what if we have similar extensions in the future, where the option-tag is the only thing you initially send?

If we have similar things in the future, then shame on us for approving 
them.

>> We've had this debate before. :)
>>
>> We all agreed there were RFCs which used it to mean both "I understand" and "I am using" the extension, and we wouldn't deprecate/change them.
>>
>> But I'm not sure we ever agreed on what we'd do going forward.
> 
> It would be really good if we could document the outcome of such debates, rather than leaving them in an e-mail somewhere...

Yes, it would be good. Where would you propose to put it?
Is it a problem we need to solve? Or do we already have it under control 
for future options.

	Thanks,
	Paul

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

From gao.yang2@zte.com.cn  Mon Apr 26 21:07:59 2010
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Subject: Re: [sipcore] The semantics of Require.
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Hi Paul,

Yes. Under semantic analysis, Require(from UAC) only means the demand of 
UAS's capability.

To expressing the semantics of demanding UAS's specific behavior (in 
specific conditions) needs something new. Relating, if UAS want the UAC's 
specific behavior (in specific conditions) should use the same *new 
header*, Require is not proper. So, Require 100rel from both UAC and UAS 
is misapplication.

And it seams that we should clarify that we haven't had tool with the 
semantics of demanding the peer's specific behavior (in specific 
conditions). But I prefer there's no correction for RFC3262 just for this 
reason.

Thanks,

Gao 

> If this discussion is to continue, can we at least have a relevant 
> subject rather than hijacking the discussion options ping?
> 
> I agree in principle with Dean's point. But from a practical perspective 

> I think we just acknowledge that the definition of certain options added 

> on additional semantics. I think we should just grandfather that but 
> refrain from making the same mistake in the future.
> 
>    Thanks,
>    Paul
> 

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<br><tt><font size=2>Hi Paul,</font></tt>
<br>
<br><tt><font size=2>Yes. Under semantic analysis, Require(from UAC) only
means the demand of UAS's capability.</font></tt>
<br>
<br><tt><font size=2>To expressing the semantics of demanding UAS's specific
behavior (in specific conditions) needs something new. Relating, if UAS
want the UAC's specific behavior (in specific conditions) should use the
same *new header*, Require is not proper. So, Require 100rel from both
UAC and UAS is misapplication.</font></tt>
<br>
<br><tt><font size=2>And it seams that we should clarify that we haven't
had tool with <b>the semantics of demanding the peer's specific behavior
(in specific conditions)</b>. But I prefer there's no correction for RFC3262
just for this reason.</font></tt>
<br>
<br><tt><font size=2>Thanks,</font></tt>
<br>
<br><tt><font size=2>Gao &nbsp;<br>
<br>
&gt; If this discussion is to continue, can we at least have a relevant
<br>
&gt; subject rather than hijacking the discussion options ping?<br>
&gt; <br>
&gt; I agree in principle with Dean's point. But from a practical perspective
<br>
&gt; I think we just acknowledge that the definition of certain options
added <br>
&gt; on additional semantics. I think we should just grandfather that but
<br>
&gt; refrain from making the same mistake in the future.<br>
&gt; <br>
&gt; &nbsp; &nbsp;Thanks,<br>
&gt; &nbsp; &nbsp;Paul<br>
&gt; </font></tt><br><pre>
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From christer.holmberg@ericsson.com  Tue Apr 27 04:54:16 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
Date: Tue, 27 Apr 2010 13:53:45 +0200
Thread-Topic: Semantics of option-tags (was:draft-jones-sip-options-ping-00.txt)
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References: <p2j53f87ca11004210707td7869937s83ec155fdd0f9ffc@mail.gmail.com> <02a301cae2fc$639733d0$2ac59b70$@com> <32A7449E-341E-4B0D-B0C9-5F95731C2F2B@softarmor.com> <02c001cae306$45e49e40$d1addac0$@com> <4BD1D895.8030900@cisco.com>, <50D76842-FE81-4AEC-ADD7-D3039D3A491F@softarmor.com> <FF84A09F50A6DC48ACB6714F4666CC745E21B30AF1@ESESSCMS0354.eemea.ericsson.se>, <C03F0E81-4442-48CD-B8AF-902967411E21@softarmor.com> <FF84A09F50A6DC48ACB6714F4666CC745E21B30AF3@ESESSCMS0354.eemea.ericsson.se>, <20F20E85-AB18-40CB-9AB8-76F19D849A8E@softarmor.com> <FF84A09F50A6DC48ACB6714F4666CC74632A8942D9@ESESSCMS0354.eemea.ericsson.se> <F87F84FE-FA3E-4DFA-AE63-E10E2628F3A7@softarmor.com> <FF84A09F50A6DC48ACB6714F4666CC74632A837746@ESESSCMS0354.eemea.ericsson.se> <430FC6BDED356B4C8498F634416644A91A7A06C058@mail> <FF84A09F50A6DC48ACB6714F4666CC74632A9E759D@ESESSCMS0354.eemea.ericsson.se>, <430FC6BDED356B4C8498F634416644A91A7A06C088@mail>
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Subject: Re: [sipcore] Semantics of option-tags (was:draft-jones-sip-options-ping-00.txt)
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Hi,

>> So, how does this work for 100rel? The INVITE request does not contain a=
ny
>> of the extension specific header fields (RSeq and RAck). They will be us=
ed
>> when the UAS starts using the extension.
>>
>> Yes, I know, you say we don't deprcate/change 100rel, but what if we hav=
e
>> similar extensions in the future, where the option-tag is the only thing
>> you initially send?
>
>We would make sure they don't just send an option-tag only.  In some cases=
 this would just mean a placebo header gets defined and used, like for RFC =
4488 norefersub where a "Refer-Sub: false" is used.

In many cases the extension does have associated information elements (head=
er fields, SDP parameters etc), but they still don't say anything about whe=
ther the receiver should use the required extension or not. They simply pro=
vide additional information associated with the extension. Preconditions is=
 a good example.

I just wonder why a UA would require support of some extension, if it doesn=
't also require the usage of it... Exactly HOW it is then used can of cours=
e depend on header field values etc.

Regards,

Christer=

From christer.holmberg@ericsson.com  Tue Apr 27 05:03:49 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 27 Apr 2010 14:03:31 +0200
Thread-Topic: [sipcore] Semantics of option-tags
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Cc: Miller' <mlm.michael.miller@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>, "'Michael@core3.amsl.com" <'Michael@core3.amsl.com>, "Paul E. Jones" <paulej@packetizer.com>
Subject: Re: [sipcore] Semantics of option-tags
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Hi,

>>Yes, I know, you say we don't deprcate/change 100rel, but what if we have=
 similar extensions in the future, where the option-tag is the only thing y=
ou initially send?
>
>If we have similar things in the future, then shame on us for approving th=
em.

Yes, but "we" (referring to the people active in the WG at this point) will=
 not be here forever. This should be clear also to people defining SIP exte=
nsion in 100 years from now :)

In addition, even if we are not going to change existing extensions, I am n=
ot sure all of them are very clear on what Require really means. Maybe some=
 of the assume that Require also indciates UAC support, but it is not state=
d anywhere. So, I guess we would need to check that.

>>>We've had this debate before. :)
>>>
>>>We all agreed there were RFCs which used it to mean both "I understand" =
and "I am using" the extension, and we wouldn't deprecate/change them.
>>>
>>>But I'm not sure we ever agreed on what we'd do going forward.
>>
>>It would be really good if we could document the outcome of such debates,=
 rather than leaving them in an e-mail somewhere...
>
>Yes, it would be good. Where would you propose to put it?
>
>Is it a problem we need to solve? Or do we already have it under control f=
or future options.

There are two things we need to consider:

1. Even if "we" have it under control - is it clearly enough specified?

2. As described above, even if we won't change the meaning of Require in ex=
isting extensions, is it always clear what Require actually means in those =
extensions?

Regards,

Christer=

From pkyzivat@cisco.com  Tue Apr 27 06:46:05 2010
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] Semantics of option-tags
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Christer Holmberg wrote:
> Hi,
> 
>>> So, how does this work for 100rel? The INVITE request does not contain any
>>> of the extension specific header fields (RSeq and RAck). They will be used
>>> when the UAS starts using the extension.
>>>
>>> Yes, I know, you say we don't deprcate/change 100rel, but what if we have
>>> similar extensions in the future, where the option-tag is the only thing
>>> you initially send?
>> We would make sure they don't just send an option-tag only.  In some cases this would just mean a placebo header gets defined and used, like for RFC 4488 norefersub where a "Refer-Sub: false" is used.
> 
> In many cases the extension does have associated information elements (header fields, SDP parameters etc), but they still don't say anything about whether the receiver should use the required extension or not. They simply provide additional information associated with the extension. Preconditions is a good example.
> 
> I just wonder why a UA would require support of some extension, if it doesn't also require the usage of it... Exactly HOW it is then used can of course depend on header field values etc.

Well, the simplest case of this is when "support" means 
ability/willingness to receive and act upon something in a message. In 
that case the UAC might Require that because it *might* need to send 
something that requires it. OTOH the circumstance where it would send 
that might never come up, and so the required support may never be employed.

Another case might be a conflict between capability and policy.
The UAS might be capable of supporting some option, but unwilling to do 
so, in some circumstances, due to policy. It thus may have to wait until 
the circumstances where the option would be used to evaluate the policy 
and decide what to do.

	Thanks,
	Paul

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Cc: Miller' <mlm.michael.miller@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>, "'Michael@core3.amsl.com" <'Michael@core3.amsl.com>, "Paul E. Jones" <paulej@packetizer.com>
Subject: Re: [sipcore] Semantics of option-tags
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Christer,

I expect you know by now that my inclination is to carefully "dot every 
'i' and cross every 't'", so philosophically I am entirely with you.

But from a practical perspective I see this as a very low priority 
issue. If we happen to be doing something else that we could reasonably 
piggy back this on, then I would go for it. But I can't get very worked 
up over initiating an effort specifically to do this.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi,
> 
>>> Yes, I know, you say we don't deprcate/change 100rel, but what if we have similar extensions in the future, where the option-tag is the only thing you initially send?
>> If we have similar things in the future, then shame on us for approving them.
> 
> Yes, but "we" (referring to the people active in the WG at this point) will not be here forever. This should be clear also to people defining SIP extension in 100 years from now :)
> 
> In addition, even if we are not going to change existing extensions, I am not sure all of them are very clear on what Require really means. Maybe some of the assume that Require also indciates UAC support, but it is not stated anywhere. So, I guess we would need to check that.
> 
>>>> We've had this debate before. :)
>>>>
>>>> We all agreed there were RFCs which used it to mean both "I understand" and "I am using" the extension, and we wouldn't deprecate/change them.
>>>>
>>>> But I'm not sure we ever agreed on what we'd do going forward.
>>> It would be really good if we could document the outcome of such debates, rather than leaving them in an e-mail somewhere...
>> Yes, it would be good. Where would you propose to put it?
>>
>> Is it a problem we need to solve? Or do we already have it under control for future options.
> 
> There are two things we need to consider:
> 
> 1. Even if "we" have it under control - is it clearly enough specified?
> 
> 2. As described above, even if we won't change the meaning of Require in existing extensions, is it always clear what Require actually means in those extensions?
> 
> Regards,
> 
> Christer

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To: "'Paul Kyzivat'" <pkyzivat@cisco.com>, "'Hadriel Kaplan'" <HKaplan@acmepacket.com>
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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I don't think we should reserve a name.  Use of options "ping" should be
between two devices that have some kind of established relationship and the
"ping" address should be known only locally.

Paul

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Monday, April 26, 2010 8:56 AM
> To: Hadriel Kaplan
> Cc: Paul E. Jones; 'Michael Miller'; sipcore@ietf.org
> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
> 
> 
> 
> Hadriel Kaplan wrote:
> > "root" obviously.  ;)
> > But seriously, picking a crazy username that won't collide is not
> that hard.  The question is if it's the right thing to do.
> 
> I don't believe in reserving names from other people's namespaces after
> the fact, on general principles. You cannot know how they use their
> namespace. You can reduce the probability of collision by making the
> value very obscure, but then it is very obscure.
> 
> So IMO it is the wrong thing to do.
> 
> 	Thanks,
> 	Paul
> 
> > Paul Kyzivat <pkyzivat@cisco.com> wrote:
> >
> >
> > I will grant you that having per-user policies for what to include in
> > OPTIONS would be possible.
> >
> > But using that approach here means reserving a specific username.
> Which
> > one are we allowed to pick that won't impact somebody? (I expect
> there
> > are people out there named "ping" who might not like having their
> user
> > name preempted.)
> >
> > AFAIK we have never preempted a user name, except for Anonymous,
> which
> > is only preempted in the anonymous.invalid domain. I'd much rather
> see a
> > special uri parameter to designate this behavior.
> >
> >         Thanks,
> >         Paul
> >
> > Hadriel Kaplan wrote:
> >>> -----Original Message-----
> >>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf
> >>> Of Paul Kyzivat
> >>> Sent: Friday, April 23, 2010 1:28 PM
> >>>
> >>> Making a decision based on the form of the R-URI presumes that non-
> ping
> >>> users of options never use an ip address as the R-URI. Requiring
> the
> >>> R-URI to be ping@ip reserves a user part, and also is just a sleazy
> way
> >>> of adding another indicator to the message to indicate its intended
> use.
> >> Actually in a weird way I think doing a specific username may be the
> right thing to do.  Think about it from an email perspective.  When you
> check on the status and settings of an email list using SMTP, do you
> (a) send a new different SMTP message type, or (b) address a normal
> message to the actual logical administrative code for the list?  You
> send a normal message to userlist-request@emailserver.com.
> >>
> >> And logically the thing being "pinged" is a higher layer, because
> the response provides an administrative status as well. (even though in
> code one would probably do this at a low level)
> >>
> >> Put it another way: in theory you could send a SUBSCRIBE to this
> logical admin-UA in that proxy so that it Notifies you when it goes
> admin up/down.  That logical UA is a resource in the proxy, and could
> be addressed as such.
> >>
> >> -hadriel
> >>
> >
> 



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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul E. Jones wrote:
> I don't think we should reserve a name.  Use of options "ping" should be
> between two devices that have some kind of established relationship and the
> "ping" address should be known only locally.

IIUC, you are saying that any specific options-ping behavior should be 
the result of *configuration* of the UAC and UAS. And then can I infer 
that recognition that in incoming OPIONS is a ping would be based on 
source address and/or use of a "special" address that is configured into 
the UAC rather than reserved.

Is that right?

	Thanks,
	Paul (K)

> Paul
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Monday, April 26, 2010 8:56 AM
>> To: Hadriel Kaplan
>> Cc: Paul E. Jones; 'Michael Miller'; sipcore@ietf.org
>> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
>> 00.txt
>>
>>
>>
>> Hadriel Kaplan wrote:
>>> "root" obviously.  ;)
>>> But seriously, picking a crazy username that won't collide is not
>> that hard.  The question is if it's the right thing to do.
>>
>> I don't believe in reserving names from other people's namespaces after
>> the fact, on general principles. You cannot know how they use their
>> namespace. You can reduce the probability of collision by making the
>> value very obscure, but then it is very obscure.
>>
>> So IMO it is the wrong thing to do.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Paul Kyzivat <pkyzivat@cisco.com> wrote:
>>>
>>>
>>> I will grant you that having per-user policies for what to include in
>>> OPTIONS would be possible.
>>>
>>> But using that approach here means reserving a specific username.
>> Which
>>> one are we allowed to pick that won't impact somebody? (I expect
>> there
>>> are people out there named "ping" who might not like having their
>> user
>>> name preempted.)
>>>
>>> AFAIK we have never preempted a user name, except for Anonymous,
>> which
>>> is only preempted in the anonymous.invalid domain. I'd much rather
>> see a
>>> special uri parameter to designate this behavior.
>>>
>>>         Thanks,
>>>         Paul
>>>
>>> Hadriel Kaplan wrote:
>>>>> -----Original Message-----
>>>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>> Behalf
>>>>> Of Paul Kyzivat
>>>>> Sent: Friday, April 23, 2010 1:28 PM
>>>>>
>>>>> Making a decision based on the form of the R-URI presumes that non-
>> ping
>>>>> users of options never use an ip address as the R-URI. Requiring
>> the
>>>>> R-URI to be ping@ip reserves a user part, and also is just a sleazy
>> way
>>>>> of adding another indicator to the message to indicate its intended
>> use.
>>>> Actually in a weird way I think doing a specific username may be the
>> right thing to do.  Think about it from an email perspective.  When you
>> check on the status and settings of an email list using SMTP, do you
>> (a) send a new different SMTP message type, or (b) address a normal
>> message to the actual logical administrative code for the list?  You
>> send a normal message to userlist-request@emailserver.com.
>>>> And logically the thing being "pinged" is a higher layer, because
>> the response provides an administrative status as well. (even though in
>> code one would probably do this at a low level)
>>>> Put it another way: in theory you could send a SUBSCRIBE to this
>> logical admin-UA in that proxy so that it Notifies you when it goes
>> admin up/down.  That logical UA is a resource in the proxy, and could
>> be addressed as such.
>>>> -hadriel
>>>>
> 
> 
> 

From christer.holmberg@ericsson.com  Tue Apr 27 13:00:13 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Tue, 27 Apr 2010 21:59:53 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,=20

>> I don't think we should reserve a name.  Use of options "ping" should=20
>> be between two devices that have some kind of established relationship=20
>> and the "ping" address should be known only locally.
>
>IIUC, you are saying that any specific options-ping behavior should be the=
 result of *configuration* of the UAC and UAS. And then can I infer that re=
cognition that in incoming OPIONS is a ping would be=20
>based on source address and/or use of a "special" address that is configur=
ed into the UAC rather than reserved.

Or a URI parameter. I assume the pings will use a pre-defined route, and wo=
uld not be re-targeted etc. Maybe we can also assume that they will be used=
 between adjacent nodes.

Regards,

Christer



=20
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Monday, April 26, 2010 8:56 AM
>> To: Hadriel Kaplan
>> Cc: Paul E. Jones; 'Michael Miller'; sipcore@ietf.org
>> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
>> 00.txt
>>
>>
>>
>> Hadriel Kaplan wrote:
>>> "root" obviously.  ;)
>>> But seriously, picking a crazy username that won't collide is not
>> that hard.  The question is if it's the right thing to do.
>>
>> I don't believe in reserving names from other people's namespaces=20
>> after the fact, on general principles. You cannot know how they use=20
>> their namespace. You can reduce the probability of collision by=20
>> making the value very obscure, but then it is very obscure.
>>
>> So IMO it is the wrong thing to do.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Paul Kyzivat <pkyzivat@cisco.com> wrote:
>>>
>>>
>>> I will grant you that having per-user policies for what to include=20
>>> in OPTIONS would be possible.
>>>
>>> But using that approach here means reserving a specific username.
>> Which
>>> one are we allowed to pick that won't impact somebody? (I expect
>> there
>>> are people out there named "ping" who might not like having their
>> user
>>> name preempted.)
>>>
>>> AFAIK we have never preempted a user name, except for Anonymous,
>> which
>>> is only preempted in the anonymous.invalid domain. I'd much rather
>> see a
>>> special uri parameter to designate this behavior.
>>>
>>>         Thanks,
>>>         Paul
>>>
>>> Hadriel Kaplan wrote:
>>>>> -----Original Message-----
>>>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org]=20
>>>>> On
>> Behalf
>>>>> Of Paul Kyzivat
>>>>> Sent: Friday, April 23, 2010 1:28 PM
>>>>>
>>>>> Making a decision based on the form of the R-URI presumes that=20
>>>>> non-
>> ping
>>>>> users of options never use an ip address as the R-URI. Requiring
>> the
>>>>> R-URI to be ping@ip reserves a user part, and also is just a=20
>>>>> sleazy
>> way
>>>>> of adding another indicator to the message to indicate its=20
>>>>> intended
>> use.
>>>> Actually in a weird way I think doing a specific username may be=20
>>>> the
>> right thing to do.  Think about it from an email perspective.  When=20
>> you check on the status and settings of an email list using SMTP, do=20
>> you
>> (a) send a new different SMTP message type, or (b) address a normal=20
>> message to the actual logical administrative code for the list?  You=20
>> send a normal message to userlist-request@emailserver.com.
>>>> And logically the thing being "pinged" is a higher layer, because
>> the response provides an administrative status as well. (even though=20
>> in code one would probably do this at a low level)
>>>> Put it another way: in theory you could send a SUBSCRIBE to this
>> logical admin-UA in that proxy so that it Notifies you when it goes=20
>> admin up/down.  That logical UA is a resource in the proxy, and could=20
>> be addressed as such.
>>>> -hadriel
>>>>
>=20
>=20
>=20
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul,

> Paul E. Jones wrote:
> > I don't think we should reserve a name.  Use of options "ping" should
> be
> > between two devices that have some kind of established relationship
> and the
> > "ping" address should be known only locally.
> 
> IIUC, you are saying that any specific options-ping behavior should be
> the result of *configuration* of the UAC and UAS. And then can I infer
> that recognition that in incoming OPIONS is a ping would be based on
> source address and/or use of a "special" address that is configured
> into
> the UAC rather than reserved.
> 
> Is that right?

I don't think behavior of OPTIONS "ping" should be unspecified (as it is
today).  I'm only saying that we should not reserve in a specification a
particular address (e.g., "sip:ping@host") to which SIP OPTIONS should be
directed.  A vendor might want to process OPTIONS "ping" messages addressed
as sip:192.168.1.10 or sip:ping@example.com, and I see no reason to specify
what that is.  We might specify that either form may be used.

What I had written in the draft originally was that OPTIONS should be
addressed only as sip:hostport, but I removed that paragraph from my working
draft since I can see some value in allowing a user part in the URI.

Paul




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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> Or a URI parameter. I assume the pings will use a pre-defined route,
> and would not be re-targeted etc. Maybe we can also assume that they
> will be used between adjacent nodes.

What we wrote in the draft in that regard is that SIP messages must be sent
directly to the IP address of the peer.  The intent is that these messages
are not going to be sent to any intermediaries.

Paul



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Paul E. Jones wrote:
> Paul,
> 
>> Paul E. Jones wrote:
>>> I don't think we should reserve a name.  Use of options "ping" should
>> be
>>> between two devices that have some kind of established relationship
>> and the
>>> "ping" address should be known only locally.
>> IIUC, you are saying that any specific options-ping behavior should be
>> the result of *configuration* of the UAC and UAS. And then can I infer
>> that recognition that in incoming OPIONS is a ping would be based on
>> source address and/or use of a "special" address that is configured
>> into
>> the UAC rather than reserved.
>>
>> Is that right?
> 
> I don't think behavior of OPTIONS "ping" should be unspecified (as it is
> today).  I'm only saying that we should not reserve in a specification a
> particular address (e.g., "sip:ping@host") to which SIP OPTIONS should be
> directed.  A vendor might want to process OPTIONS "ping" messages addressed
> as sip:192.168.1.10 or sip:ping@example.com, and I see no reason to specify
> what that is.  We might specify that either form may be used.
> 
> What I had written in the draft originally was that OPTIONS should be
> addressed only as sip:hostport, but I removed that paragraph from my working
> draft since I can see some value in allowing a user part in the URI.

I think I understand, but I still not certain I do.

I think you are saying that:

- there are potentially two behaviors in response to OPTIONS:
   . the "ping" behavior, which is potentially simpler
     and more optimized
   . the "normal" (non ping) behavior, which may have extra
     detail and be bigger and/or more expensive to produce.

- that there is *some* feature of the incoming OPTIONS message
   that the recipient uses to distinguish which behavior to apply.

- that the particular feature used to make that distinction
   may vary from one UAS to another. Its the responsibility
   of the UAC (e.g. via configuration) to know what it is that
   the UAS will used to decide, and then to construct the OPTIONS
   appropriately depending on which behavior it wants.

Is that right?

	Thanks,
	Paul

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Paul E. Jones" <paulej@packetizer.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Tue, 27 Apr 2010 22:10:11 -0400
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Tuesday, April 27, 2010 4:17 PM
> To: 'Paul Kyzivat'
>=20
> I don't think behavior of OPTIONS "ping" should be unspecified (as it is
> today).  I'm only saying that we should not reserve in a specification a
> particular address (e.g., "sip:ping@host") to which SIP OPTIONS should be
> directed.  A vendor might want to process OPTIONS "ping" messages
> addressed
> as sip:192.168.1.10 or sip:ping@example.com, and I see no reason to
> specify
> what that is.  We might specify that either form may be used.

The rub with that is it doesn't remove the per-hop interop configuration th=
at a draft such as this one is trying to remove/avoid.  I'm not saying that=
 makes the draft useless, because at least it would avoid the response-code=
 configuration portion - but just wondering why we can't avoid the intended=
 target issue as well.

BTW, defining the syntax for the intent of the request can be done multiple=
 ways, and I was only suggesting that agreeing on a specific user name woul=
d be one way.  (it's always seemed to me to be a hack that requests intende=
d for a Proxy to process as a logical UA are not addressed to a specific lo=
gical UA/resource, but instead just inferred by the Proxy as being for it)

But there are other ways of skinning that cat.  It would be nice to agree o=
n one way, though, to avoid more config.

-hadriel

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Subject: [sipcore] OPTIONS, except not (was Re: [RAI] Option-tag registration)
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[as participant]

Moving this conversation from RAI to SIPCORE, since it seems more 
appropriate over here now.

On 4/27/10 21:30, Apr 27, Hadriel Kaplan wrote:
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: Tuesday, April 27, 2010 10:01 PM
>> To: Dean Willis
>>
>> This is really, really starting to sound like a new method.
>>      
> Why?  Creating a new method doesn't avoid the issue of intended target routing. (a legacy proxy would forward it like any other unknown method)
>    

It's not just the target routing. Taken in aggregate, this thread has 
basically taken the position that it "works just like OPTIONS, except 
for everything." The removal of OPTIONS' actual raison d'être -- the 
querying of capabilities -- is clearly a last straw.

I know that OPTIONS has the benefit of being one of the Fourteen True 
Methods handed down to Moses on Mount Sinai, but if we're really looking 
at something intended as a liveness check -- as opposed to a 
capabilities query -- let's go ahead and do it correctly.

This is a new thing, let's make it a new thing.

/a

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Adam Roach <adam@nostrum.com>
Date: Tue, 27 Apr 2010 23:49:45 -0400
Thread-Topic: OPTIONS, except not (was Re: [RAI] Option-tag registration)
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Subject: Re: [sipcore] OPTIONS, except not (was Re: [RAI] Option-tag registration)
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> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Tuesday, April 27, 2010 11:21 PM
> To: Hadriel Kaplan
>=20
> It's not just the target routing. Taken in aggregate, this thread has
> basically taken the position that it "works just like OPTIONS, except
> for everything." The removal of OPTIONS' actual raison d'=EAtre -- the
> querying of capabilities -- is clearly a last straw.

But really the capabilities piece is the "everything" part of a "except for=
 everything".  I mean the rest of it (status codes, max-forwards, etc.) was=
 actually stuff rfc3261 talked about applying to OPTIONS.  So except for th=
e capabilities piece, Paul's draft is just trying to be more explicit about=
 what OPTIONS already legitimately does I think.  No?

But I do think it should query capabilities.  It just may be the capabiliti=
es of the target are very, very limited. :)

-hadriel

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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul,

> I think you are saying that:
> 
> - there are potentially two behaviors in response to OPTIONS:
>    . the "ping" behavior, which is potentially simpler
>      and more optimized
>    . the "normal" (non ping) behavior, which may have extra
>      detail and be bigger and/or more expensive to produce.

I would expect that if a device claims support for the draft we've
submitted, then it will always provide an "optimized" implementation
(whatever that means).  Likely, it will return a 200 with some fairly static
payload or another code that makes sense.

The "normal" reply would suggest the device does not support the draft,
right?  That's a case I'm not interested in what such a device would do.

I guess the question here is how we know whether an OPTIONS is for "ping" or
to query for OPTIONS.  What I suggested is that manufacturers would design
their boxes to support the draft -- and the draft tries to align with 3261.
There should be some address to which ping messages are directed, which my
preference would be the IP address of the device.  The reason for my
suggesting that is that, generally, one does not establish a session with
the IP address of an SBC :-)  That said, I can see where it might be useful
to have a ping@example.com type address for such things as IP phones or even
SBCs.  In any case, the "ping" address should be an address that the
manufacturer discloses and the peer entity should be configured to use.

> - that there is *some* feature of the incoming OPTIONS message
>    that the recipient uses to distinguish which behavior to apply.

I'm not opposed to it, but it seems like overkill.  If one were to build an
SBC that expects to receive an OPTIONS "ping" at "sip:ping@192.168.1.10" and
the feature tag (or whatever) for "ping" is not there, do you really think
the box would behave differently?

> - that the particular feature used to make that distinction
>    may vary from one UAS to another. Its the responsibility
>    of the UAC (e.g. via configuration) to know what it is that
>    the UAS will used to decide, and then to construct the OPTIONS
>    appropriately depending on which behavior it wants.
> 
> Is that right?

If we feel that some kind of tag is necessary to identify a request as a
"ping", I would certainly not want it to vary by implementation.  However,
I'm not yet convinced we even need such a feature tag, as I think the
Request URI would suffice.

Paul



From dean.willis@softarmor.com  Tue Apr 27 21:56:06 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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On Apr 27, 2010, at 11:30 PM, Paul E. Jones wrote:

>
> I guess the question here is how we know whether an OPTIONS is for  
> "ping" or
> to query for OPTIONS.  What I suggested is that manufacturers would  
> design
> their boxes to support the draft -- and the draft tries to align  
> with 3261.
> There should be some address to which ping messages are directed,  
> which my
> preference would be the IP address of the device.  The reason for my
> suggesting that is that, generally, one does not establish a session  
> with
> the IP address of an SBC :-)  That said, I can see where it might be  
> useful
> to have a ping@example.com type address for such things as IP phones  
> or even
> SBCs.  In any case, the "ping" address should be an address that the
> manufacturer discloses and the peer entity should be configured to  
> use.

In my experience, everybody uses IP addresses for almost everything.  
It's a mistake to assume DNS in a real-world deployment outside of the  
occasional user-configured proxy.

So if we don't want any of the behavior of OPTIONS why not just do a  
coaxial STUN request on the SIP port, as one would do for outbound?  
It's MUCH lighter weight and doesn't even need text parsing.


>
>> - that there is *some* feature of the incoming OPTIONS message
>>   that the recipient uses to distinguish which behavior to apply.
>
> I'm not opposed to it, but it seems like overkill.  If one were to  
> build an
> SBC that expects to receive an OPTIONS "ping" at "sip:ping@192.168.1.10 
> " and
> the feature tag (or whatever) for "ping" is not there, do you really  
> think
> the box would behave differently?

Yes. Because if it's got a reasonable SIP stack and I send it an  
OPTIONS request, I expect to see a legitimate OPTIONS response.



>> - that the particular feature used to make that distinction
>>   may vary from one UAS to another. Its the responsibility
>>   of the UAC (e.g. via configuration) to know what it is that
>>   the UAS will used to decide, and then to construct the OPTIONS
>>   appropriately depending on which behavior it wants.
>>
>> Is that right?
>
> If we feel that some kind of tag is necessary to identify a request  
> as a
> "ping", I would certainly not want it to vary by implementation.   
> However,
> I'm not yet convinced we even need such a feature tag, as I think the
> Request URI would suffice.

The tag could be encoded in the R-URI, in the user-name, a parameter  
on the user-name, as a request-URI parameter, or as part of the target  
host name.

Or it could be a new header field inside the body of the request.  
That's got the best backward compatibility of any of the options  
discussed.

--
dean


From dean.willis@softarmor.com  Tue Apr 27 22:01:01 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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On Apr 27, 2010, at 2:59 PM, Christer Holmberg wrote:

>
> Hi,
>
>>> I don't think we should reserve a name.  Use of options "ping"  
>>> should
>>> be between two devices that have some kind of established  
>>> relationship
>>> and the "ping" address should be known only locally.
>>
>> IIUC, you are saying that any specific options-ping behavior should  
>> be the result of *configuration* of the UAC and UAS. And then can I  
>> infer that recognition that in incoming OPIONS is a ping would be
>> based on source address and/or use of a "special" address that is  
>> configured into the UAC rather than reserved.
>
> Or a URI parameter. I assume the pings will use a pre-defined route,  
> and would not be re-targeted etc. Maybe we can also assume that they  
> will be used between adjacent nodes.

But we always have to consider (and document) that which happens if  
they are not used as expected. What happens when you ping a non- 
adjacent node? What happens when the ping is contact-routed or loose- 
routed by an intermediate proxy? I'm not saying that we need to have a  
perfect solution, but at least we need to consider and document the  
likely failures and oddities.

For example, saying "The node being pinged MUST be an adjacent node"  
is insufficient. This is a "naked requirement" -- for it to be  
respected, we need to put clothes on it. Perhaps we could add   
"because if the node is not adjacent, then the following is likely...."

--
Dean

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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hadriel,

> > I don't think behavior of OPTIONS "ping" should be unspecified (as it
> is
> > today).  I'm only saying that we should not reserve in a
> specification a
> > particular address (e.g., "sip:ping@host") to which SIP OPTIONS
> should be
> > directed.  A vendor might want to process OPTIONS "ping" messages
> > addressed
> > as sip:192.168.1.10 or sip:ping@example.com, and I see no reason to
> > specify
> > what that is.  We might specify that either form may be used.
> 
> The rub with that is it doesn't remove the per-hop interop
> configuration that a draft such as this one is trying to remove/avoid.
> I'm not saying that makes the draft useless, because at least it would
> avoid the response-code configuration portion - but just wondering why
> we can't avoid the intended target issue as well.

I assume you're referring to such things as performing a DNS query on a
name, discovering 4 IP addresses, and then needed to know how to address
them (if not by IP address).  In that case, yes, I can see why a defined
username or some type of "ping" tag might be necessary.

A question: why not just use the IP address and port as the "ping" address?
I'm sure there may be a few good reasons, but that seems like a logical
destination (and is the reason I proposed it in the draft).
 
> BTW, defining the syntax for the intent of the request can be done
> multiple ways, and I was only suggesting that agreeing on a specific
> user name would be one way.  (it's always seemed to me to be a hack
> that requests intended for a Proxy to process as a logical UA are not
> addressed to a specific logical UA/resource, but instead just inferred
> by the Proxy as being for it)
> 
> But there are other ways of skinning that cat.  It would be nice to
> agree on one way, though, to avoid more config.

Agree.  This is not a complex problem, so I'd like to just agree on
something and go forward.  Where's that coin?  Or do I need to roll a die?
What options do we want to consider?

Paul




From christer.holmberg@ericsson.com  Wed Apr 28 00:55:27 2010
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To: Dean Willis <dean.willis@softarmor.com>
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Hi,=20

>>>> I don't think we should reserve a name.  Use of options "ping" =20
>>>> should
>>>> be between two devices that have some kind of established=20
>>>> relationship and the "ping" address should be known only locally.
>>>
>>> IIUC, you are saying that any specific options-ping behavior should=20
>>> be the result of *configuration* of the UAC and UAS. And then can I=20
>>> infer that recognition that in incoming OPIONS is a ping would be=20
>>> based on source address and/or use of a "special" address that is=20
>>> configured into the UAC rather than reserved.
>>
>> Or a URI parameter. I assume the pings will use a pre-defined route,=20
>> and would not be re-targeted etc. Maybe we can also assume that they=20
>> will be used between adjacent nodes.
>=20
>But we always have to consider (and document) that which=20
>happens if they are not used as expected. What happens when=20
>you ping a non- adjacent node? What happens when the ping is=20
>contact-routed or loose- routed by an intermediate proxy? I'm=20
>not saying that we need to have a perfect solution, but at=20
>least we need to consider and document the likely failures=20
>and oddities.
>=20
>For example, saying "The node being pinged MUST be an adjacent node" =20
>is insufficient. This is a "naked requirement" -- for it to be =20
>respected, we need to put clothes on it. Perhaps we could add  =20
>"because if the node is not adjacent, then the following is=20
>likely...."

True.

But, you could say that the Max-Forwards value MUST be 1.

Regards,

Christer


From christer.holmberg@ericsson.com  Wed Apr 28 01:25:23 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Wed, 28 Apr 2010 10:25:05 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,=20

>>I guess the question here is how we know whether an OPTIONS is for=20
>>"ping" or to query for OPTIONS.  What I suggested is that=20
>>manufacturers would design their boxes to support the draft=20
>>-- and the=20
>>draft tries to align with 3261.
>>There should be some address to which ping messages are directed,=20
>>which my preference would be the IP address of the device. =20
>>The reason for my suggesting that is that, generally, one does not establ=
ish a=20
>>session with the IP address of an SBC :-)  That said, I can see where=20
>>it might be useful to have a ping@example.com type address for such=20
>>things as IP phones or even SBCs.  In any case, the "ping" address=20
>>should be an address that the manufacturer discloses and the peer=20
>>entity should be configured to use.
>=20
>In my experience, everybody uses IP addresses for almost everything. =20
>It's a mistake to assume DNS in a real-world deployment=20
>outside of the occasional user-configured proxy.
>=20
>So if we don't want any of the behavior of OPTIONS why not=20
>just do a coaxial STUN request on the SIP port, as one would=20
>do for outbound? =20
>It's MUCH lighter weight and doesn't even need text parsing.

The keep-draft uses STUN. However, when you send something in-dialog, or in=
-registration, you have the possibility to negotiate the usage.

However, if you send something "stand alone", you don't know whether the re=
ceiver supports something. Of course, you could send ONE OPTIONS, and see f=
rom the response whether the receiver supports stun, eg using the mechanism=
 in the keep-draft.

If you start sending STUN messages, and the receiver does not support it, i=
t could treat the messages as crap, and in the worse case even put you on s=
ome black list :)

Regards,

Christer




> >> - that the particular feature used to make that distinction
> >>   may vary from one UAS to another. Its the responsibility
> >>   of the UAC (e.g. via configuration) to know what it is that
> >>   the UAS will used to decide, and then to construct the OPTIONS
> >>   appropriately depending on which behavior it wants.
> >>
> >> Is that right?
> >
> > If we feel that some kind of tag is necessary to identify a=20
> request as=20
> > a
> > "ping", I would certainly not want it to vary by implementation.  =20
> > However,
> > I'm not yet convinced we even need such a feature tag, as I=20
> think the=20
> > Request URI would suffice.
>=20
> The tag could be encoded in the R-URI, in the user-name, a=20
> parameter on the user-name, as a request-URI parameter, or as=20
> part of the target host name.
>=20
> Or it could be a new header field inside the body of the request. =20
> That's got the best backward compatibility of any of the=20
> options discussed.
>=20
> --
> dean
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul E. Jones wrote:
> Paul,
> 
>> I think you are saying that:
>>
>> - there are potentially two behaviors in response to OPTIONS:
>>    . the "ping" behavior, which is potentially simpler
>>      and more optimized
>>    . the "normal" (non ping) behavior, which may have extra
>>      detail and be bigger and/or more expensive to produce.
> 
> I would expect that if a device claims support for the draft we've
> submitted, then it will always provide an "optimized" implementation
> (whatever that means).  Likely, it will return a 200 with some fairly static
> payload or another code that makes sense.

I can think of cases where that wouldn't be the case.
Certainly the other OPTIONS discussion is suggesting returning *more* 
information about capabilities of the addressed resource. And that might 
be expected in a box that should also be involved in pinging.

For instance a "registrar/home-proxy" box serving atlanta.com might be 
expected to respond to OPTIONS for sip:alice@atlanta with specifics 
about alice - possibly a 3xx with the registered contacts.

OTOH a neighboring "load balancer" proxy might want to be pinging that 
box, and desiring a quick response with no "capabilities".

Of course in this case the distinction could be made by the address of 
the request. But still there are two behaviors, distinguished by 
something in the request.

> The "normal" reply would suggest the device does not support the draft,
> right?  That's a case I'm not interested in what such a device would do.

This goes to the other discussion we  have been having about Require - 
the distinction about "supporting" a feature and "using" the feature. 
Just because the box "supports" this ping doesn't mean it should use it 
all the time.

> I guess the question here is how we know whether an OPTIONS is for "ping" or
> to query for OPTIONS. 

Yup.

> What I suggested is that manufacturers would design
> their boxes to support the draft -- and the draft tries to align with 3261.

Discussed above.

> There should be some address to which ping messages are directed,

So you *are* suggesting that ping behavior be selected based on the form 
of the address in the OPTIONS request.

> which my
> preference would be the IP address of the device.  The reason for my
> suggesting that is that, generally, one does not establish a session with
> the IP address of an SBC :-) 

There are multiple "features" of that approach that might be the actual 
determiner of the behavior:

- there is no user part to the uri
- the domain part of the URI is an ip address rather than an FQDN
- both of the above

Trouble is: there is plenty of usage where that's a bad choice.
Its my understanding that there are quite a few "phone" type devices 
that use IP address without a user part as a contact address, and so 
receive real requests directly at such an address. An OPTIONS request to 
their registered Contact would eventually arrive to them with just the 
ip address in the R-URI, yet the expectation is that the request is not 
a ping. And at the same time, a proxy upstream from that device might 
indeed want to ping it.

> That said, I can see where it might be useful
> to have a ping@example.com type address for such things as IP phones or even
> SBCs.  In any case, the "ping" address should be an address that the
> manufacturer discloses and the peer entity should be configured to use.

IMO the reason we have standards is to avoid having to configure such 
things. If its worth standardizing the behavior then its worth 
standardizing how its invoked.

>> - that there is *some* feature of the incoming OPTIONS message
>>    that the recipient uses to distinguish which behavior to apply.
> 
> I'm not opposed to it, but it seems like overkill. 

As I note above, I consider a particular address form (such as 
ip-address-only R-URI, or "ping" as the user part of the R-URI) as an 
example of a feature that might be used to select the behavior.

I think there are several alternatives:

1) OPTIONS has only one behavior.

1a) A UAC may use OPTIONS as a ping, ignoring information
     in the response that is irrelevant to it.

1b) There is a new method, different from OPTIONS
     for the ping functionality.

2) OPTIONS has two different standardized behaviors: normal & ping.

2a) A given box always implements one or the other of
     those behaviors, in all cases.

2b) A given box may implement them both. The selection
     of which behavior to use for a specific request
     is determined by some feature of the request.
     (the source address, the R-URI, a special header, ...)
     Precisely what feature causes this is unstandardized.
     The sender who cares about which behavior it gets
     must "know" how to construct the request to get
     that behavior.

2c) A given box may implement them both. The selection
     of which behavior to use for a specific request
     is standardized. (Exactly what that is is TBD.

Based on what you have said, I think you are advocating 2b.
My preference would be for 1a or 1b. I could live with 2c.
I find 2a and 2b unacceptable for reasons previously stated.

Note that 1a and 1b can be used together: a UAC that wants to ping first 
tries sending the special PING request. If that is rejected, it falls 
back to using a "normal" OPTIONS as an alternative.

	Thanks,
	Paul

> If one were to build an
> SBC that expects to receive an OPTIONS "ping" at "sip:ping@192.168.1.10" and
> the feature tag (or whatever) for "ping" is not there, do you really think
> the box would behave differently?
> 
>> - that the particular feature used to make that distinction
>>    may vary from one UAS to another. Its the responsibility
>>    of the UAC (e.g. via configuration) to know what it is that
>>    the UAS will used to decide, and then to construct the OPTIONS
>>    appropriately depending on which behavior it wants.
>>
>> Is that right?
> 
> If we feel that some kind of tag is necessary to identify a request as a
> "ping", I would certainly not want it to vary by implementation.  However,
> I'm not yet convinced we even need such a feature tag, as I think the
> Request URI would suffice.
> 
> Paul
> 
> 
> 

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Registration closes May 1. If you are planning to attend and have not =
yet registered, please do so now.

RjS

On Mar 16, 2010, at 11:15 AM, Robert Sparks wrote:

> SIPit 26 will be held in Kista - the telecom valley outside Stockholm, =
Sweden on May 17-21, 2010. The event is hosted by Edvina and TANDBERG =
and sponsored by Ingate, Intertex and .se.
>=20
> See <http://www.sipit.net> and <http://sipit.edvina.se> for details =
and to register.
>=20
> RjS
>=20


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<html><head></head><body style=3D"word-wrap: break-word; =
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"><div><br><div style=3D"word-wrap: break-word; -webkit-nbsp-mode: =
space; -webkit-line-break: after-white-space; =
"><div><br><div>Registration closes May 1. If you are planning to attend =
and have not yet registered, please do so now.<br><br>RjS<br><br>On Mar =
16, 2010, at 11:15 AM, Robert Sparks wrote:<br><br><blockquote =
type=3D"cite">SIPit 26 will be held in Kista - the telecom valley =
outside Stockholm, Sweden on May 17-21, 2010. The event is hosted by =
Edvina and TANDBERG and sponsored by Ingate, Intertex and =
.se.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">See &lt;<a =
href=3D"http://www.sipit.net/">http://www.sipit.net</a>&gt; and &lt;<a =
href=3D"http://sipit.edvina.se/">http://sipit.edvina.se</a>&gt; for =
details and to register.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Dean,

> In my experience, everybody uses IP addresses for almost everything.

I can see where this might be an issue for sending "ping" messages to
phones.

> It's a mistake to assume DNS in a real-world deployment outside of the
> occasional user-configured proxy.

DNS is just one way of obtaining an address.  One is not precluded from
getting the address in any way whatsoever.
 
> So if we don't want any of the behavior of OPTIONS why not just do a
> coaxial STUN request on the SIP port, as one would do for outbound?
> It's MUCH lighter weight and doesn't even need text parsing.

The purpose for a SIP OPTIONS ping is to get a response from the SIP
process, not another process on the box.  Following your argument, an ICMP
packet is even lighter :-)
 
> >> - that there is *some* feature of the incoming OPTIONS message
> >>   that the recipient uses to distinguish which behavior to apply.
> >
> > I'm not opposed to it, but it seems like overkill.  If one were to
> > build an
> > SBC that expects to receive an OPTIONS "ping" at
> "sip:ping@192.168.1.10
> > " and
> > the feature tag (or whatever) for "ping" is not there, do you really
> > think
> > the box would behave differently?
> 
> Yes. Because if it's got a reasonable SIP stack and I send it an
> OPTIONS request, I expect to see a legitimate OPTIONS response.

Yeah, ok.  Agreed.
 
> >> - that the particular feature used to make that distinction
> >>   may vary from one UAS to another. Its the responsibility
> >>   of the UAC (e.g. via configuration) to know what it is that
> >>   the UAS will used to decide, and then to construct the OPTIONS
> >>   appropriately depending on which behavior it wants.
> >>
> >> Is that right?
> >
> > If we feel that some kind of tag is necessary to identify a request
> > as a
> > "ping", I would certainly not want it to vary by implementation.
> > However,
> > I'm not yet convinced we even need such a feature tag, as I think the
> > Request URI would suffice.
> 
> The tag could be encoded in the R-URI, in the user-name, a parameter
> on the user-name, as a request-URI parameter, or as part of the target
> host name.
> 
> Or it could be a new header field inside the body of the request.
> That's got the best backward compatibility of any of the options
> discussed.

Personally, I'd prefer to make this as simple and unobtrusive as possible.
What is the preference?  We've got too many options here.

Paul


From HKaplan@acmepacket.com  Wed Apr 28 16:58:48 2010
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To: "Paul E. Jones" <paulej@packetizer.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Wed, 28 Apr 2010 19:58:28 -0400
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Wednesday, April 28, 2010 12:31 AM
> To: 'Paul Kyzivat'
>=20
> I guess the question here is how we know whether an OPTIONS is for "ping"
> or
> to query for OPTIONS.  What I suggested is that manufacturers would desig=
n
> their boxes to support the draft -- and the draft tries to align with 326=
1.

Right, but when I design my box to handle it, I need to know the OPTIONS re=
quest I receive from your box expects your draft's specific semantics.  Bec=
ause I still need to support my legacy OPTIONS request handling code, inclu=
ding for OPTIONS which were just routed through your box to me and thus are=
 not expecting the same draft's semantics.


> There should be some address to which ping messages are directed, which m=
y
> preference would be the IP address of the device.  The reason for my
> suggesting that is that, generally, one does not establish a session with
> the IP address of an SBC :-) =20

I beg to differ - for every SBC vendor I've seen, you do establish a sessio=
n (or target an OPTIONS request) to the SBC.  I'm sure there are some SBC v=
endors gear I haven't seen, which may act as some inline bump-in-the-wire A=
LG-thingy, but that's not the model "generally". =20


> That said, I can see where it might be
> useful
> to have a ping@example.com type address for such things as IP phones or
> even
> SBCs.  In any case, the "ping" address should be an address that the
> manufacturer discloses and the peer entity should be configured to use.

I'm getting confused - are you thinking the IP Address of the OPTIONS' requ=
est-URI would be a *different* IP Address from the one other SIP requests a=
re sent to??  I didn't grok that from the draft at all, and it would make t=
hings very complicated.

-hadriel

From dean.willis@softarmor.com  Wed Apr 28 17:17:48 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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On Apr 28, 2010, at 6:55 PM, Paul E. Jones wrote:

>> So if we don't want any of the behavior of OPTIONS why not just do a
>> coaxial STUN request on the SIP port, as one would do for outbound?
>> It's MUCH lighter weight and doesn't even need text parsing.
>
> The purpose for a SIP OPTIONS ping is to get a response from the SIP
> process, not another process on the box.  Following your argument,  
> an ICMP
> packet is even lighter :-)

Right, but the coaxial-STUN is at least driven from the same port- 
listener-dispatcher as the SIP process. Remember, we're running both  
protocols on the same port, so the packets go to the same listener.  
One can envision a multithreaded model with the SIP thread has crashed  
but the listener and STUN responders linger on, but I system a more  
total faliure is likely to be the norm.

>> Or it could be a new header field inside the body of the request.
>> That's got the best backward compatibility of any of the options
>> discussed.
>
> Personally, I'd prefer to make this as simple and unobtrusive as  
> possible.
> What is the preference?  We've got too many options here.


I think the simplest thing is a new header field, coupled with an  
option-tag that could be used to indicate or require support for the  
header field. Yes, it is standards-track work, but it's not "hard"  
standards track; we know how to do these things reasonably well by  
now. An AD could bless it through the individual track using a 1 month  
IETF LC.

So for example, a new header field "Options-mode" with a value of  
"ping" means "If you support this extension, do not proxy this  
request, just send an immediate 200 OK response  and do not bother  
populating the Supported header field of the 200 OK response."  
Somebody who didn't recognize the header field would just send a  
conventional 200 OK response.

And perhaps an option tag of "optionsmodeping". That way we can add  
further options-modes, and option tags for them in the future as (and  
if) needed.

You could basically cut-and-paste the answer-mode draft and be done  
writing in an hour.

--
Dean

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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hadriel,

> > That said, I can see where it might be
> > useful
> > to have a ping@example.com type address for such things as IP phones
> or
> > even
> > SBCs.  In any case, the "ping" address should be an address that the
> > manufacturer discloses and the peer entity should be configured to
> use.
> 
> I'm getting confused - are you thinking the IP Address of the OPTIONS'
> request-URI would be a *different* IP Address from the one other SIP
> requests are sent to??  I didn't grok that from the draft at all, and
> it would make things very complicated.

No, I certainly didn't mean to suggest there would a different IP address.

What I am referring to is the Request URI.  Previously, I was suggesting
that the request URI would be addressed to the IP address if it's a ping,
but would be addressed to a user if it wasn't.  But, you and Dean have
convinced me that that's not always a good way to differentiate an OPTIONS
"ping" from one that's an actual OPTIONS request.

So, I removed the restriction from my local draft that specified what the
Request URI looks like.  What we need to decide is exactly what we want to
have as a differentiator?  A specific request URI, a tag, a header, etc.

Having said that, I'm still favorable to sending an OPTIONS "ping" addressed
to the IP address of the device (e.g., sip:192.168.1.10).  Whether that
contains a "ping" header or tag or other is likely not going to make much
difference to an SBC that does not receive other requests addressed to that
address.  

Paul



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>
References: <aqw0yhx7u936e9xuwleac6av.1272063656876@email.android.com> <4BD58D45.3000709@cisco.com> <004f01cae641$4e690a90$eb3b1fb0$@com> <4BD73FE1.7050105@cisco.com> <005b01cae646$9892b0d0$c9b81270$@com> <4BD76496.50905@cisco.com> <005301cae68b$8f7d8110$ae788330$@com> <1BDEDBAC-7959-4767-85EA-BCF3E9D6C746@softarmor.com> <00da01cae72e$4f5590c0$ee00b240$@com> <8748ED98-3A25-4E30-B9D7-FE0C08ECA33B@softarmor.com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Dean,

> So for example, a new header field "Options-mode" with a value of
> "ping" means "If you support this extension, do not proxy this
> request, just send an immediate 200 OK response  and do not bother
> populating the Supported header field of the 200 OK response."
> Somebody who didn't recognize the header field would just send a
> conventional 200 OK response.

Not objecting to the header, but I think we would have failed in our effort
if the message might possibly be proxied.  The intent was for this to be
peer-to-peer.
 
> And perhaps an option tag of "optionsmodeping". That way we can add
> further options-modes, and option tags for them in the future as (and
> if) needed.

So, you want a head like this:

  Options-Mode: ping

Or this:

  Require: optionsmodeping

> You could basically cut-and-paste the answer-mode draft and be done
> writing in an hour.

Oh, I'll be happy to plagiarize your work ... but not sure exactly what you
want me to copy :-)

Paul



From pkyzivat@cisco.com  Wed Apr 28 19:37:39 2010
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References: <aqw0yhx7u936e9xuwleac6av.1272063656876@email.android.com> <4BD58D45.3000709@cisco.com> <004f01cae641$4e690a90$eb3b1fb0$@com> <4BD73FE1.7050105@cisco.com> <005b01cae646$9892b0d0$c9b81270$@com> <4BD76496.50905@cisco.com> <005301cae68b$8f7d8110$ae788330$@com> <1BDEDBAC-7959-4767-85EA-BCF3E9D6C746@softarmor.com> <00da01cae72e$4f5590c0$ee00b240$@com> <8748ED98-3A25-4E30-B9D7-FE0C08ECA33B@softarmor.com> <010901cae73a$766dfec0$6349fc40$@com>
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Now that we have established the goal is devising some options/headers 
to entirely change the behavior of OPTIONS, I would rather see a new 
method: PING. Given the assumptions being made I can see few if any 
downsides to this, and definite upsides.

IMO the value of using OPTIONS for ping is in those cases where the 
device being pinged doesn't know that it is being pinged.

	Thanks,
	Paul

Paul E. Jones wrote:
> Dean,
> 
>> So for example, a new header field "Options-mode" with a value of
>> "ping" means "If you support this extension, do not proxy this
>> request, just send an immediate 200 OK response  and do not bother
>> populating the Supported header field of the 200 OK response."
>> Somebody who didn't recognize the header field would just send a
>> conventional 200 OK response.
> 
> Not objecting to the header, but I think we would have failed in our effort
> if the message might possibly be proxied.  The intent was for this to be
> peer-to-peer.
>  
>> And perhaps an option tag of "optionsmodeping". That way we can add
>> further options-modes, and option tags for them in the future as (and
>> if) needed.
> 
> So, you want a head like this:
> 
>   Options-Mode: ping
> 
> Or this:
> 
>   Require: optionsmodeping
> 
>> You could basically cut-and-paste the answer-mode draft and be done
>> writing in an hour.
> 
> Oh, I'll be happy to plagiarize your work ... but not sure exactly what you
> want me to copy :-)
> 
> Paul
> 
> 
> 

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
References: <aqw0yhx7u936e9xuwleac6av.1272063656876@email.android.com> <4BD58D45.3000709@cisco.com> <004f01cae641$4e690a90$eb3b1fb0$@com> <4BD73FE1.7050105@cisco.com> <005b01cae646$9892b0d0$c9b81270$@com> <4BD76496.50905@cisco.com> <005301cae68b$8f7d8110$ae788330$@com> <1BDEDBAC-7959-4767-85EA-BCF3E9D6C746@softarmor.com> <00da01cae72e$4f5590c0$ee00b240$@com> <8748ED98-3A25-4E30-B9D7-FE0C08ECA33B@softarmor.com> <010901cae73a$766dfec0$6349fc40$@com> <4BD8F0E7.1070000@cisco.com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul,

I don't see it quite the same way.  I don't want to entirely change OPTIONS,
but I do want to define procedures that devices should follow if they
implement an OPTIONS "ping" procedure.  This is achieved with a new header
or targeting a specific address.  It also means that devices that do not
implement the resulting RFC will still function as they do today.

If we go the route of introducing a whole new method, it would be years
before it gets introduced into the market.  We still need to keep the
OPTIONS "ping" around, because that's what virtually everybody is doing.

The only thing we proposed in the initial draft was some consistency with
respect to the way responses were given, etc.  We're not calling for drastic
changes in the behavior of OPTIONS.  In fact, if a device wanted to return a
complete response with all of the features it supports, that'd be fine with
me.  When used as a "ping", it's primarily the response code the peer is
looking for.  The only negative side effect is a bit more bandwidth on the
wire, which is likely inconsequential as compared to everything else.

Paul

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Wednesday, April 28, 2010 10:37 PM
> To: Paul E. Jones
> Cc: 'Dean Willis'; 'Michael Miller'; sipcore@ietf.org; 'Hadriel Kaplan'
> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
> 
> Now that we have established the goal is devising some options/headers
> to entirely change the behavior of OPTIONS, I would rather see a new
> method: PING. Given the assumptions being made I can see few if any
> downsides to this, and definite upsides.
> 
> IMO the value of using OPTIONS for ping is in those cases where the
> device being pinged doesn't know that it is being pinged.
> 
> 	Thanks,
> 	Paul
> 
> Paul E. Jones wrote:
> > Dean,
> >
> >> So for example, a new header field "Options-mode" with a value of
> >> "ping" means "If you support this extension, do not proxy this
> >> request, just send an immediate 200 OK response  and do not bother
> >> populating the Supported header field of the 200 OK response."
> >> Somebody who didn't recognize the header field would just send a
> >> conventional 200 OK response.
> >
> > Not objecting to the header, but I think we would have failed in our
> effort
> > if the message might possibly be proxied.  The intent was for this to
> be
> > peer-to-peer.
> >
> >> And perhaps an option tag of "optionsmodeping". That way we can add
> >> further options-modes, and option tags for them in the future as
> (and
> >> if) needed.
> >
> > So, you want a head like this:
> >
> >   Options-Mode: ping
> >
> > Or this:
> >
> >   Require: optionsmodeping
> >
> >> You could basically cut-and-paste the answer-mode draft and be done
> >> writing in an hour.
> >
> > Oh, I'll be happy to plagiarize your work ... but not sure exactly
> what you
> > want me to copy :-)
> >
> > Paul
> >
> >
> >
> 



From HKaplan@acmepacket.com  Wed Apr 28 20:51:07 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Paul E. Jones" <paulej@packetizer.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Wed, 28 Apr 2010 23:50:50 -0400
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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The problem is OPTIONS really is meant for getting capabilities, and althou=
gh it may also be used as a ping (and often is), to use one for *only* a pi=
ng and not return capabilities is likely to cause a bunch of folks here to =
say "then it's not an OPTIONS".  So I think that has to be kept in to keep =
this method name and get anywhere.

If we did identify the target of the OPTIONS as a specific resource, like "=
ping@ip/host", then I would have argued we could specify that it does retur=
n capabilities but that this "ping" logical UA/resource simply has no capab=
ilities beyond supporting the OPTIONS request.

But really if it's proxies you're talking about they generally can represen=
t their capabilities with fairly hard-coded static strings, afaict, and obv=
iously the system receiving the response can just ignore the capabilities r=
eturned if it doesn't care.  So I'm not sure it makes much difference, does=
 it?

-hadriel

> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Wednesday, April 28, 2010 10:55 PM
>=20
> I don't see it quite the same way.  I don't want to entirely change
> OPTIONS,
> but I do want to define procedures that devices should follow if they
> implement an OPTIONS "ping" procedure.  This is achieved with a new heade=
r
> or targeting a specific address.  It also means that devices that do not
> implement the resulting RFC will still function as they do today.
>=20
> If we go the route of introducing a whole new method, it would be years
> before it gets introduced into the market.  We still need to keep the
> OPTIONS "ping" around, because that's what virtually everybody is doing.
>=20
> The only thing we proposed in the initial draft was some consistency with
> respect to the way responses were given, etc.  We're not calling for
> drastic
> changes in the behavior of OPTIONS.  In fact, if a device wanted to retur=
n
> a
> complete response with all of the features it supports, that'd be fine
> with
> me.  When used as a "ping", it's primarily the response code the peer is
> looking for.  The only negative side effect is a bit more bandwidth on th=
e
> wire, which is likely inconsequential as compared to everything else.
>=20
> Paul
>=20

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Hadriel Kaplan'" <HKaplan@acmepacket.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hadriel,

> The problem is OPTIONS really is meant for getting capabilities, and
> although it may also be used as a ping (and often is), to use one for
> *only* a ping and not return capabilities is likely to cause a bunch of
> folks here to say "then it's not an OPTIONS".  So I think that has to
> be kept in to keep this method name and get anywhere.

We never proposed a use where capabilities are not returned.  If folks
wanted to do that, they could, but we're not prescribing different behavior
w.r.t. OPTIONS payload content.  Personally, I would reply with the same
payload, "ping" or not.
 
> If we did identify the target of the OPTIONS as a specific resource,
> like "ping@ip/host", then I would have argued we could specify that it
> does return capabilities but that this "ping" logical UA/resource
> simply has no capabilities beyond supporting the OPTIONS request.

What I really, really wanted to encourage is sending an OPTIONS message to
the IP address of the device you wish you ping and with a SIP URI of
"sip:hostport".  This does not address a "user" and therefore makes it a
little clearer that it's a device-level request.

In any case, I would still return the same payload, just because I'd likely
have it statically composed in memory and it would be trivial to return.
 
> But really if it's proxies you're talking about they generally can
> represent their capabilities with fairly hard-coded static strings,
> afaict, and obviously the system receiving the response can just ignore
> the capabilities returned if it doesn't care.  So I'm not sure it makes
> much difference, does it?

I think that's true of most devices: they can be fairly static responses.

What we were trying to do with this draft was not be disruptive to the
existing SIP spec, but to try to get OPTIONS handled in a consistent manner
when used as a "ping" mechanism.  We didn't want to call for a whole
different mode of operations, per se.

Paul
 
> -hadriel
> 
> > -----Original Message-----
> > From: Paul E. Jones [mailto:paulej@packetizer.com]
> > Sent: Wednesday, April 28, 2010 10:55 PM
> >
> > I don't see it quite the same way.  I don't want to entirely change
> > OPTIONS,
> > but I do want to define procedures that devices should follow if they
> > implement an OPTIONS "ping" procedure.  This is achieved with a new
> header
> > or targeting a specific address.  It also means that devices that do
> not
> > implement the resulting RFC will still function as they do today.
> >
> > If we go the route of introducing a whole new method, it would be
> years
> > before it gets introduced into the market.  We still need to keep the
> > OPTIONS "ping" around, because that's what virtually everybody is
> doing.
> >
> > The only thing we proposed in the initial draft was some consistency
> with
> > respect to the way responses were given, etc.  We're not calling for
> > drastic
> > changes in the behavior of OPTIONS.  In fact, if a device wanted to
> return
> > a
> > complete response with all of the features it supports, that'd be
> fine
> > with
> > me.  When used as a "ping", it's primarily the response code the peer
> is
> > looking for.  The only negative side effect is a bit more bandwidth
> on the
> > wire, which is likely inconsequential as compared to everything else.
> >
> > Paul
> >
> 



From christer.holmberg@ericsson.com  Wed Apr 28 22:24:45 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Paul E. Jones" <paulej@packetizer.com>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Thu, 29 Apr 2010 07:19:34 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt 
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

>>The problem is OPTIONS really is meant for getting=20
>>capabilities, and although it may also be used as a ping (and often is), =
to=20
>>use one for *only* a ping and not return capabilities is likely to=20
>>cause a bunch of folks here to say "then it's not an OPTIONS".  So I=20
>>think that has to be kept in to keep this method name and get anywhere.
>=20
>We never proposed a use where capabilities are not returned. =20
>If folks wanted to do that, they could, but we're not=20
>prescribing different behavior w.r.t. OPTIONS payload=20
>content.  Personally, I would reply with the same payload,=20
>"ping" or not.
>
>>If we did identify the target of the OPTIONS as a specific resource,=20
>>like "ping@ip/host", then I would have argued we could specify that it=20
>>does return capabilities but that this "ping" logical UA/resource=20
>>simply has no capabilities beyond supporting the OPTIONS request.
>=20
>What I really, really wanted to encourage is sending an=20
>OPTIONS message to the IP address of the device you wish you=20
>ping and with a SIP URI of "sip:hostport".  This does not=20
>address a "user" and therefore makes it a little clearer that=20
>it's a device-level request.
>=20
>In any case, I would still return the same payload, just=20
>because I'd likely have it statically composed in memory and=20
>it would be trivial to return.

The message body payload part is easy to get rid of - you just include an e=
mpty Accept header field in the OPTIONS request.

>>But really if it's proxies you're talking about they generally can=20
>>represent their capabilities with fairly hard-coded static strings,=20
>>afaict, and obviously the system receiving the response can just=20
>>ignore the capabilities returned if it doesn't care.  So=20
>>I'm not sure it makes much difference, does it?
>=20
>I think that's true of most devices: they can be fairly static responses.

I think one issue is how often you think these OPTIONS requests need to be =
sent. If they are not sent that often, it's probably not a problem to retur=
n what you would return in normal cases.

Regards,

Christer




>=20
> What we were trying to do with this draft was not be=20
> disruptive to the existing SIP spec, but to try to get=20
> OPTIONS handled in a consistent manner when used as a "ping"=20
> mechanism.  We didn't want to call for a whole different mode=20
> of operations, per se.
>=20
> Paul
> =20
> > -hadriel
> >=20
> > > -----Original Message-----
> > > From: Paul E. Jones [mailto:paulej@packetizer.com]
> > > Sent: Wednesday, April 28, 2010 10:55 PM
> > >
> > > I don't see it quite the same way.  I don't want to=20
> entirely change=20
> > > OPTIONS, but I do want to define procedures that devices should=20
> > > follow if they implement an OPTIONS "ping" procedure.  This is=20
> > > achieved with a new
> > header
> > > or targeting a specific address.  It also means that=20
> devices that do
> > not
> > > implement the resulting RFC will still function as they do today.
> > >
> > > If we go the route of introducing a whole new method, it would be
> > years
> > > before it gets introduced into the market.  We still need to keep=20
> > > the OPTIONS "ping" around, because that's what virtually=20
> everybody=20
> > > is
> > doing.
> > >
> > > The only thing we proposed in the initial draft was some=20
> consistency
> > with
> > > respect to the way responses were given, etc.  We're not=20
> calling for=20
> > > drastic changes in the behavior of OPTIONS.  In fact, if a device=20
> > > wanted to
> > return
> > > a
> > > complete response with all of the features it supports, that'd be
> > fine
> > > with
> > > me.  When used as a "ping", it's primarily the response code the=20
> > > peer
> > is
> > > looking for.  The only negative side effect is a bit more=20
> bandwidth
> > on the
> > > wire, which is likely inconsequential as compared to=20
> everything else.
> > >
> > > Paul
> > >
> >=20
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From HKaplan@acmepacket.com  Wed Apr 28 22:36:30 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Paul E. Jones" <paulej@packetizer.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Thu, 29 Apr 2010 01:36:13 -0400
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Wednesday, April 28, 2010 11:59 PM
>=20
> We never proposed a use where capabilities are not returned.  If folks
> wanted to do that, they could, but we're not prescribing different
> behavior
> w.r.t. OPTIONS payload content.  Personally, I would reply with the same
> payload, "ping" or not.

Oh.  Well someone did. :)

Good, so we can move onto flipping coins to decide what syntax to use to ma=
ke it clear the OPTIONS is using/expecting this draft's rules. :)

-hadriel=20


From alexander.milinski@nsn.com  Thu Apr 29 00:33:56 2010
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From: "Milinski, Alexander (NSN - DE/Munich)" <alexander.milinski@nsn.com>
To: "ext Hadriel Kaplan" <HKaplan@acmepacket.com>, "Paul E. Jones" <paulej@packetizer.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
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Cc: Michael Miller <mlm.michael.miller@gmail.com>, sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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One moment please, before you start throwing the dice ...

Having followed the debate and looked at the draft, it still isn't clear
to me:
What is broken today that the draft indends to fix?
What are the underlying requirements for the solution?

Alex


> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of ext Hadriel Kaplan
> Sent: Thursday, April 29, 2010 7:36 AM
> To: Paul E. Jones; 'Paul Kyzivat'
> Cc: 'Michael Miller'; sipcore@ietf.org
> Subject: Re: [sipcore] FW: I-D=20
> Action:draft-jones-sip-options-ping-00.txt
>=20
>=20
>=20
> > -----Original Message-----
> > From: Paul E. Jones [mailto:paulej@packetizer.com]
> > Sent: Wednesday, April 28, 2010 11:59 PM
> >=20
> > We never proposed a use where capabilities are not=20
> returned.  If folks
> > wanted to do that, they could, but we're not prescribing different
> > behavior
> > w.r.t. OPTIONS payload content.  Personally, I would reply=20
> with the same
> > payload, "ping" or not.
>=20
> Oh.  Well someone did. :)
>=20
> Good, so we can move onto flipping coins to decide what=20
> syntax to use to make it clear the OPTIONS is using/expecting=20
> this draft's rules. :)
>=20
> -hadriel=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul J,

I see two alternatives here, but you seem to be straddling them.

1) you can leave the behavior of OPTIONS alone.
    Document a set of practices for how a UAC can use
    OPTIONS to accomplish a ping. In this case, the
    UAS never knows whether an OPTIONS it receives
    is intended as a capability query or a ping.

    + this is relatively quick to accomplish because
      it doesn't require standards action

    + this is relatively quick to deploy, because
      only the UACs that want to ping need be updated.

    - the benefit is limited because you are stuck
      with the way OPTIONS currently works, and with
      the variations in behavior that it allows.

    - pinging this way may be more costly than
      necessary because of the extra work of applying
      policy, including capabilities, and the extra
      size of the message to carry the capabilities

2) define new standardized behavior for pinging,
    with roles for both the UAC and UAS, that is
    different from what OPTIONS does today.

    + the behavior is not constrained by the existing
      definition of OPTIONS. It can be defined
      precisely for the purpose.

    + the implementation can be optimized.
      the server can "fast path" it, and policy
      application can be simplified because little
      is disclosed.

    - it will take longer to get specified because it
      must be standards track. (Note that this will be
      pretty much the same regardless of whether it is
      defined as new optional behavior of OPTIONS, or
      as a new method.)

    - it will take longer to deploy because both
      UAC and UAS must support it.

I guess there is a possible variant on (1):
also do a *clarification* of OPTIONS behavior in cases where there is 
some ambiguity about what response code should be returned in particular 
circumstances. I would be ok with that *if* the clarification applied to 
all uses of OPTIONS. But once you start to do that you start to pick up 
the negatives of (2).

It would also be possible to do a combination of (1) and (2), where (2) 
is preferred if both UAS supports it, with fallback to (1) for those 
that don't.

What I don't see is a way to avoid standards action and yet define 
special optional behavior by the UAS that the UAC can request and know 
was provided. This seems to be what you are seeking.

	Thanks,
	Paul K

Paul E. Jones wrote:
> Paul,
> 
> I don't see it quite the same way.  I don't want to entirely change OPTIONS,
> but I do want to define procedures that devices should follow if they
> implement an OPTIONS "ping" procedure.  This is achieved with a new header
> or targeting a specific address.  It also means that devices that do not
> implement the resulting RFC will still function as they do today.
> 
> If we go the route of introducing a whole new method, it would be years
> before it gets introduced into the market.  We still need to keep the
> OPTIONS "ping" around, because that's what virtually everybody is doing.
> 
> The only thing we proposed in the initial draft was some consistency with
> respect to the way responses were given, etc.  We're not calling for drastic
> changes in the behavior of OPTIONS.  In fact, if a device wanted to return a
> complete response with all of the features it supports, that'd be fine with
> me.  When used as a "ping", it's primarily the response code the peer is
> looking for.  The only negative side effect is a bit more bandwidth on the
> wire, which is likely inconsequential as compared to everything else.
> 
> Paul
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Wednesday, April 28, 2010 10:37 PM
>> To: Paul E. Jones
>> Cc: 'Dean Willis'; 'Michael Miller'; sipcore@ietf.org; 'Hadriel Kaplan'
>> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
>> 00.txt
>>
>> Now that we have established the goal is devising some options/headers
>> to entirely change the behavior of OPTIONS, I would rather see a new
>> method: PING. Given the assumptions being made I can see few if any
>> downsides to this, and definite upsides.
>>
>> IMO the value of using OPTIONS for ping is in those cases where the
>> device being pinged doesn't know that it is being pinged.
>>
>> 	Thanks,
>> 	Paul
>>
>> Paul E. Jones wrote:
>>> Dean,
>>>
>>>> So for example, a new header field "Options-mode" with a value of
>>>> "ping" means "If you support this extension, do not proxy this
>>>> request, just send an immediate 200 OK response  and do not bother
>>>> populating the Supported header field of the 200 OK response."
>>>> Somebody who didn't recognize the header field would just send a
>>>> conventional 200 OK response.
>>> Not objecting to the header, but I think we would have failed in our
>> effort
>>> if the message might possibly be proxied.  The intent was for this to
>> be
>>> peer-to-peer.
>>>
>>>> And perhaps an option tag of "optionsmodeping". That way we can add
>>>> further options-modes, and option tags for them in the future as
>> (and
>>>> if) needed.
>>> So, you want a head like this:
>>>
>>>   Options-Mode: ping
>>>
>>> Or this:
>>>
>>>   Require: optionsmodeping
>>>
>>>> You could basically cut-and-paste the answer-mode draft and be done
>>>> writing in an hour.
>>> Oh, I'll be happy to plagiarize your work ... but not sure exactly
>> what you
>>> want me to copy :-)
>>>
>>> Paul
>>>
>>>
>>>
> 
> 
> 

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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>, 'Michael Miller' <mlm.michael.miller@gmail.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Christer Holmberg wrote:

> The message body payload part is easy to get rid of - you just include an empty Accept header field in the OPTIONS request.

That *could* work, if the UAS understands and does it right.

I'll venture to guess that most existing implementations would either:
a) return the SDP anyway (assuming there were prepared to return it.)
b) barf and return some error code, hopefully 406

If (a) happened, you could file a complaint against the vendor of the 
UAS. But I think you could make a good case that (b) is appropriate in 
this case, as it would if you put an empty Accept in an INVITE.

	Thanks,
	Paul

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Date: Thu, 29 Apr 2010 10:51:59 -0400
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Thursday, April 29, 2010 10:12 AM
>=20
> Christer Holmberg wrote:
>=20
> > The message body payload part is easy to get rid of - you just include
> an empty Accept header field in the OPTIONS request.
>=20
> That *could* work, if the UAS understands and does it right.
>=20
> I'll venture to guess that most existing implementations would either:
> a) return the SDP anyway (assuming there were prepared to return it.)
> b) barf and return some error code, hopefully 406
>=20
> If (a) happened, you could file a complaint against the vendor of the
> UAS. But I think you could make a good case that (b) is appropriate in
> this case, as it would if you put an empty Accept in an INVITE.

Unfortunately I think 3261 is ambiguous for that:
   A message body MAY be sent, the type of which is determined by the
   Accept header field in the OPTIONS request (application/sdp is the
   default if the Accept header field is not present).

-hadriel


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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Hadriel Kaplan'" <HKaplan@acmepacket.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Christer,

> >I think that's true of most devices: they can be fairly static
> responses.
> 
> I think one issue is how often you think these OPTIONS requests need to
> be sent. If they are not sent that often, it's probably not a problem
> to return what you would return in normal cases.

I expect that to vary for a number of reasons, but I would suspect that the
more frequent they are sent, the less important the bandwidth would be.
There's no requirement to even look at the SDP, and I doubt a device sending
a "ping" packet would.

In any case, as you suggest we could get rid of it if it was important via
the Accept header.  But, that actually would mean more processing.  It's
either CPU or bandwidth :-)

Paul



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Milinski, Alexander \(NSN - DE/Munich\)'" <alexander.milinski@nsn.com>,  "'ext Hadriel Kaplan'" <HKaplan@acmepacket.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Alex,

There are two objectives:
1) Legitimize the use of OPTIONS as a "ping" mechanism, since virtually
every product I've seen does that, but it's not described in any RFC
2) Try to converge on behavior that is in line with what RFC 3261 says, but
narrow the recommended response codes.  When a device cannot accept to
requests, I've seen the following response codes returned: 408, 480, 503,
505, and 600. The only thing place where I've seen implementation agreement
is returning a 200 when things are OK.

In meeting these objectives, we really, really wanted to not make any
changes to the core spec, but I have no objection to introducing a new
header, for example, to hint as to the purpose of the OPTIONS: I just don't
personally think it's that important, but others apparently do.

Paul

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Milinski, Alexander (NSN - DE/Munich)
> Sent: Thursday, April 29, 2010 3:34 AM
> To: ext Hadriel Kaplan; Paul E. Jones; Paul Kyzivat
> Cc: Michael Miller; sipcore@ietf.org
> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
> 
> One moment please, before you start throwing the dice ...
> 
> Having followed the debate and looked at the draft, it still isn't
> clear
> to me:
> What is broken today that the draft indends to fix?
> What are the underlying requirements for the solution?
> 
> Alex
> 
> 
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of ext Hadriel Kaplan
> > Sent: Thursday, April 29, 2010 7:36 AM
> > To: Paul E. Jones; 'Paul Kyzivat'
> > Cc: 'Michael Miller'; sipcore@ietf.org
> > Subject: Re: [sipcore] FW: I-D
> > Action:draft-jones-sip-options-ping-00.txt
> >
> >
> >
> > > -----Original Message-----
> > > From: Paul E. Jones [mailto:paulej@packetizer.com]
> > > Sent: Wednesday, April 28, 2010 11:59 PM
> > >
> > > We never proposed a use where capabilities are not
> > returned.  If folks
> > > wanted to do that, they could, but we're not prescribing different
> > > behavior
> > > w.r.t. OPTIONS payload content.  Personally, I would reply
> > with the same
> > > payload, "ping" or not.
> >
> > Oh.  Well someone did. :)
> >
> > Good, so we can move onto flipping coins to decide what
> > syntax to use to make it clear the OPTIONS is using/expecting
> > this draft's rules. :)
> >
> > -hadriel
> >
> > _______________________________________________
> > 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
> 



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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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[as a participant]

On 4/29/10 10:01 AM, Paul E. Jones wrote:
> There's no requirement to even look at the SDP, and I doubt a device 
> sending a "ping" packet would.

If you're trying to accomplish something orthogonal to OPTIONS (i.e, 
liveness tests), and planning to ignore the capabilities indicated by 
the responding entity, why do you want to use OPTIONS?

/a

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Adam Roach <adam@nostrum.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Thu, 29 Apr 2010 11:42:49 -0400
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Thursday, April 29, 2010 11:12 AM
> To: Paul E. Jones
>=20
> On 4/29/10 10:01 AM, Paul E. Jones wrote:
> > There's no requirement to even look at the SDP, and I doubt a device
> > sending a "ping" packet would.
>=20
> If you're trying to accomplish something orthogonal to OPTIONS (i.e,
> liveness tests), and planning to ignore the capabilities indicated by
> the responding entity, why do you want to use OPTIONS?

But it's _not_ really orthogonal to OPTIONS.  We're using the term "ping" h=
ere, but that's a bit misleading - the request is really being used to dete=
rmine the *state* of the next-hop.  Classic "ping" is just "are you there?"=
 - like ICMP, or a STUN or CRLF keepalive.  But this draft is also clarifyi=
ng what to respond with for actual SIP processing state - admin up/down, ov=
erloaded, etc.

One could argue that the returned state is in fact a returned "capability" =
- the capability of the responder to handle other requests.

And 3261 pretty clearly not only allows OPTIONS to do that, but describes h=
ow it is done.  Section 11.2:
   The response to an OPTIONS is constructed using the standard rules
   for a SIP response as discussed in Section 8.2.6.  The response code
   chosen MUST be the same that would have been chosen had the request
   been an INVITE.  That is, a 200 (OK) would be returned if the UAS is
   ready to accept a call, a 486 (Busy Here) would be returned if the
   UAS is busy, etc.  This allows an OPTIONS request to be used to
   determine the basic state of a UAS, which can be an indication of
   whether the UAS will accept an INVITE request.

-hadriel

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
References: <aqw0yhx7u936e9xuwleac6av.1272063656876@email.android.com> <4BD58D45.3000709@cisco.com> <004f01cae641$4e690a90$eb3b1fb0$@com> <4BD73FE1.7050105@cisco.com> <005b01cae646$9892b0d0$c9b81270$@com> <4BD76496.50905@cisco.com> <005301cae68b$8f7d8110$ae788330$@com> <1BDEDBAC-7959-4767-85EA-BCF3E9D6C746@softarmor.com> <00da01cae72e$4f5590c0$ee00b240$@com> <8748ED98-3A25-4E30-B9D7-FE0C08ECA33B@softarmor.com> <010901cae73a$766dfec0$6349fc40$@com> <4BD8F0E7.1070000@cisco.com> <011601cae747$5013a740$f03af5c0$@com> <4BD990F3.8020909@cisco.com>
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Date: Thu, 29 Apr 2010 13:01:25 -0400
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul,

> I see two alternatives here, but you seem to be straddling them.

I've been accused of that before :-)
 
> 1) you can leave the behavior of OPTIONS alone.
>     Document a set of practices for how a UAC can use
>     OPTIONS to accomplish a ping. In this case, the
>     UAS never knows whether an OPTIONS it receives
>     is intended as a capability query or a ping.

Yes and no.  My thought was a device that receives an OPTIONS directed to
the device (i.e., Request URI is sip:hostport), then it could assume it's
likely a "ping".
 
>     + this is relatively quick to accomplish because
>       it doesn't require standards action

It is true that there is minimal work, but what we need to do is legitimize
OPTIONS for use as a "ping" method and agree on response codes.
 
>     + this is relatively quick to deploy, because
>       only the UACs that want to ping need be updated.

Definitely: this is definitely not something we're suggesting to send across
the Internet.  This is a peer-to-peer relationship.
 
>     - the benefit is limited because you are stuck
>       with the way OPTIONS currently works, and with
>       the variations in behavior that it allows.

Understood, but what is the problem with that?  Apparently, many felt it
wasn't an issue since this is widely, inconsistently deployed
 
>     - pinging this way may be more costly than
>       necessary because of the extra work of applying
>       policy, including capabilities, and the extra
>       size of the message to carry the capabilities

Yeah, but our intent was that this would be done over a local link and
probably only when the device is otherwise not generating traffic.  Perhaps
the thing I believe is a non-issue is the composition and processing of
those messages.  Surely folks are not going to construct that response based
on querying a bunch of configuration information upon every request.  I'd
assume the information is readily in hand and perhaps even sitting in a
buffer read to be copied as-is.
 
> 2) define new standardized behavior for pinging,
>     with roles for both the UAC and UAS, that is
>     different from what OPTIONS does today.

We're not looking for something that is different, per se, but just to agree
on behavior.  This behavior is in line with what 3261 says, but let's not do
things that are all over the place.

>     + the behavior is not constrained by the existing
>       definition of OPTIONS. It can be defined
>       precisely for the purpose.

We definitely want to constrain behavior: that's a major reason preparing
the draft.
 
>     + the implementation can be optimized.
>       the server can "fast path" it, and policy
>       application can be simplified because little
>       is disclosed.

It could be, but what I'm having trouble with is what I would do differently
if it was a "ping" or not.  For example, if I build a GW and I received an
OPTIONS addressed to sip:192.168.1.10, I would reply the same way every
time: I'd assume it's likely a "ping".  And, that response should be a valid
OPTIONS reply regardless of whether it's a "ping" or not.  That is, no
different code path followed.  I would do the same thing if I were building
an SBC: I'd assume the message is a "ping" and comply with this spec: a
subset of what RFC 3261 describes.

If the "ping" was directed to "sip:+19194762048@192.168.1.10", I might
return something different, since there may be capabilities associated with
a given port that is different than the GW as a whole.

>     - it will take longer to get specified because it
>       must be standards track. (Note that this will be
>       pretty much the same regardless of whether it is
>       defined as new optional behavior of OPTIONS, or
>       as a new method.)

Understood, which is why I didn't really want to introduce new
functionality.  I do not object to introducing a header, for example: that
seems like a minimal and perhaps useful addition.
 
>     - it will take longer to deploy because both
>       UAC and UAS must support it.

Exactly, which is why I think defining a whole new method like PING is out
of the question.  SIP is too old at this point to introduce such new
methods, in my opinion.  It would take years to approve and we'd have to
live with both OPTIONS and PING for 20 years.
 
> I guess there is a possible variant on (1):
> also do a *clarification* of OPTIONS behavior in cases where there is
> some ambiguity about what response code should be returned in
> particular
> circumstances. I would be ok with that *if* the clarification applied
> to
> all uses of OPTIONS. But once you start to do that you start to pick up
> the negatives of (2).

I can't fix every problem with OPTIONS and I don't see a reason to expand
this draft beyond that.  If we were introducing new functionality, I'd
agree... but I'd prefer we didn't introduce new functionality.

> It would also be possible to do a combination of (1) and (2), where (2)
> is preferred if both UAS supports it, with fallback to (1) for those
> that don't.

What I think we should do is more along the lines of (1), with possible
consideration of introducing a header or tag to serve as a hint as to the
purpose of the OPTIONS message.  (I know that compels you to suggest that I
might be bipolar, but I'm only trying recognize the fact that some find this
useful.  Still, I'd expect the same processing on the box.)
 
> What I don't see is a way to avoid standards action and yet define
> special optional behavior by the UAS that the UAC can request and know
> was provided. This seems to be what you are seeking.

So, what's special?  We're constraining behavior, not expanding it.

Paul



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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul,

> > The message body payload part is easy to get rid of - you just
> include an empty Accept header field in the OPTIONS request.
> 
> That *could* work, if the UAS understands and does it right.

If it didn't, wouldn't it be non-compliant with 3261?
 
> I'll venture to guess that most existing implementations would either:
> a) return the SDP anyway (assuming there were prepared to return it.)
> b) barf and return some error code, hopefully 406
> 
> If (a) happened, you could file a complaint against the vendor of the
> UAS. But I think you could make a good case that (b) is appropriate in
> this case, as it would if you put an empty Accept in an INVITE.

Why would you argue that a device should do (b)?

Paul



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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Adam,

> On 4/29/10 10:01 AM, Paul E. Jones wrote:
> > There's no requirement to even look at the SDP, and I doubt a device
> > sending a "ping" packet would.
> 
> If you're trying to accomplish something orthogonal to OPTIONS (i.e,
> liveness tests), and planning to ignore the capabilities indicated by
> the responding entity, why do you want to use OPTIONS?

Because that's what lots of vendors have elected to do.  We submitted this
document as a reaction to the fact that it's widely, but inconsistently
implemented.  We didn't submit the document because we wanted to define a
new behavior and encourage people to adopt it.

Paul



From christer.holmberg@ericsson.com  Thu Apr 29 10:38:17 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: SIPCORE <sipcore@ietf.org>
Date: Thu, 29 Apr 2010 19:37:59 +0200
Thread-Topic: SigComp and Service-Route
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Subject: [sipcore] SigComp and Service-Route
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Hi,

I appologize if the following issue have been discussed and solved before.

When reading RFC 5049 (Sigcomp), the UA figures out whether the outbound
proxy supports sigcomp by looking at outbound proxie's entry in the
Service-Route in the REGISTER response.

My questions are:

1. How can the UA ensure that the outbound proxy will be listed in the
Service-Route in the first place? RFC 3608 (Service-Route) does not
mandate the registrar to insert any intermediaries in the Service-Route,
e.g based on Path. It's all based on local policy.

2. Even if the registrar would build the Service-Route based on Path,
how does it know that the Path address/port values will be
usable/routable in the UA-towards-registrar direction (Path is supposed
to be used in the registrar-towards-UA direction)? This is similar to
the Record-Route problem, and intermediaries with different network
addresses/ports per direction would have to insert "double Path".

I guess the outbound proxy could insert itself in the Service-Route, but
that is not described anywhere in RFC 5049, and it is also a SHOULD NOT
in RFC 3608.

Regards,

Christer

From christer.holmberg@ericsson.com  Thu Apr 29 10:45:08 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Thu, 29 Apr 2010 19:43:42 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

You DO insert Accept, but with no values. AFAIK, that will override any def=
ault values.=20

Regards,

Christer

________________________________________
From: Hadriel Kaplan [HKaplan@acmepacket.com]
Sent: Thursday, April 29, 2010 5:51 PM
To: Paul Kyzivat; Christer Holmberg
Cc: Paul E. Jones; 'Michael Miller'; sipcore@ietf.org
Subject: RE: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Thursday, April 29, 2010 10:12 AM
>
> Christer Holmberg wrote:
>
> > The message body payload part is easy to get rid of - you just include
> an empty Accept header field in the OPTIONS request.
>
> That *could* work, if the UAS understands and does it right.
>
> I'll venture to guess that most existing implementations would either:
> a) return the SDP anyway (assuming there were prepared to return it.)
> b) barf and return some error code, hopefully 406
>
> If (a) happened, you could file a complaint against the vendor of the
> UAS. But I think you could make a good case that (b) is appropriate in
> this case, as it would if you put an empty Accept in an INVITE.

Unfortunately I think 3261 is ambiguous for that:
   A message body MAY be sent, the type of which is determined by the
   Accept header field in the OPTIONS request (application/sdp is the
   default if the Accept header field is not present).

-hadriel=

From christer.holmberg@ericsson.com  Thu Apr 29 12:28:15 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Paul E. Jones" <paulej@packetizer.com>, "'Milinski, Alexander (NSN - DE/Munich)'" <alexander.milinski@nsn.com>, 'ext Hadriel Kaplan' <HKaplan@acmepacket.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Thu, 29 Apr 2010 21:23:16 +0200
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References: <aqw0yhx7u936e9xuwleac6av.1272063656876@email.android.com><4BD58D45.3000709@cisco.com> <004f01cae641$4e690a90$eb3b1fb0$@com><4BD73FE1.7050105@cisco.com> <005b01cae646$9892b0d0$c9b81270$@com><4BD76496.50905@cisco.com> <005301cae68b$8f7d8110$ae788330$@com><1BDEDBAC-7959-4767-85EA-BCF3E9D6C746@softarmor.com><00da01cae72e$4f5590c0$ee00b240$@com><8748ED98-3A25-4E30-B9D7-FE0C08ECA33B@softarmor.com><010901cae73a$766dfec0$6349fc40$@com> <4BD8F0E7.1070000@cisco.com><011601cae747$5013a740$f03af5c0$@com><430FC6BDED356B4C8498F634416644A91A8A5F15EA@mail><011f01cae750$3b424840$b1c6d8c0$@com> <430FC6BDED356B4C8498F634416644A91A8A5F15F0@mail> <3F4C11BC54A36642BFB5875D599F47BD029E4A89@DEMUEXC013.nsn-intra.net>, <018201cae7ae$308be240$91a3a6c0$@com>
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

>There are two objectives:
>1) Legitimize the use of OPTIONS as a "ping" mechanism, since virtually
>every product I've seen does that, but it's not described in any RFC
>2) Try to converge on behavior that is in line with what RFC 3261 says, bu=
t
>narrow the recommended response codes.  When a device cannot accept to
>requests, I've seen the following response codes returned: 408, 480, 503,
>505, and 600. The only thing place where I've seen implementation agreemen=
t
>is returning a 200 when things are OK.
>
>In meeting these objectives, we really, really wanted to not make any
>changes to the core spec, but I have no objection to introducing a new
>header, for example, to hint as to the purpose of the OPTIONS: I just don'=
t
>personally think it's that important, but others apparently do.

In order for me to form an opinion on that, I would need to know how freque=
nt the "pings" are expected to be sent.

OPTIONS is not the only method that is too "heavy" if sent too often. Peopl=
e have been using frequent re-registrations in order to do NAT keep-alives,=
 and that's also bad. That is one reason we have defined the usage of STUN =
for keep-alives, because it's much more "lightweight".

So, I think it would be useful to know how often (the minimum value) these =
pings are expected to be sent.=20

Regards,

Christer



> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Milinski, Alexander (NSN - DE/Munich)
> Sent: Thursday, April 29, 2010 3:34 AM
> To: ext Hadriel Kaplan; Paul E. Jones; Paul Kyzivat
> Cc: Michael Miller; sipcore@ietf.org
> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-
> 00.txt
>
> One moment please, before you start throwing the dice ...
>
> Having followed the debate and looked at the draft, it still isn't
> clear
> to me:
> What is broken today that the draft indends to fix?
> What are the underlying requirements for the solution?
>
> Alex
>
>
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of ext Hadriel Kaplan
> > Sent: Thursday, April 29, 2010 7:36 AM
> > To: Paul E. Jones; 'Paul Kyzivat'
> > Cc: 'Michael Miller'; sipcore@ietf.org
> > Subject: Re: [sipcore] FW: I-D
> > Action:draft-jones-sip-options-ping-00.txt
> >
> >
> >
> > > -----Original Message-----
> > > From: Paul E. Jones [mailto:paulej@packetizer.com]
> > > Sent: Wednesday, April 28, 2010 11:59 PM
> > >
> > > We never proposed a use where capabilities are not
> > returned.  If folks
> > > wanted to do that, they could, but we're not prescribing different
> > > behavior
> > > w.r.t. OPTIONS payload content.  Personally, I would reply
> > with the same
> > > payload, "ping" or not.
> >
> > Oh.  Well someone did. :)
> >
> > Good, so we can move onto flipping coins to decide what
> > syntax to use to make it clear the OPTIONS is using/expecting
> > this draft's rules. :)
> >
> > -hadriel
> >
> > _______________________________________________
> > 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
>


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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul E. Jones wrote:
> Paul,
> 
>> I see two alternatives here, but you seem to be straddling them.
> 
> I've been accused of that before :-)

And you're still at it here.

>> 1) you can leave the behavior of OPTIONS alone.
>>     Document a set of practices for how a UAC can use
>>     OPTIONS to accomplish a ping. In this case, the
>>     UAS never knows whether an OPTIONS it receives
>>     is intended as a capability query or a ping.
> 
> Yes and no.  My thought was a device that receives an OPTIONS directed to
> the device (i.e., Request URI is sip:hostport), then it could assume it's
> likely a "ping".

Note: I presented these as alternatives, but you are trying to subsume 
some of the features of (2) into (1) without also accepting the 
consequences.

Your statement here is saying that the recipient can discern a 
distinction between OPTIONS that mean ping from those that don't, based 
on the form of the R-URI.

That of course presumes that the recipient agrees that there is such a 
distinction. Since the existing definition of OPTIONS has no such 
distinction, this is an *extension* to OPTIONS, and hence achieving it 
would require standards action.

>>     + this is relatively quick to accomplish because
>>       it doesn't require standards action
> 
> It is true that there is minimal work, but what we need to do is legitimize
> OPTIONS for use as a "ping" method and agree on response codes.

As soon as you say that, you have said you don't want option (1). What 
you are asking for is option (2). Or *maybe* you are talking about is 
what I describe below as: "variant on (1): also do a *clarification* of 
OPTIONS behavior in cases where there is some ambiguity about what 
response code...". If that is the case, then lets discuss that one in 
all its gory details and implications.

>>     + this is relatively quick to deploy, because
>>       only the UACs that want to ping need be updated.
> 
> Definitely: this is definitely not something we're suggesting to send across
> the Internet.  This is a peer-to-peer relationship.

I don't get your point. It has no relation to the bullet above it is 
responding to. In any case, you won't get any traction here for a 
proposal that requires matching configuration of the two "peers" in this 
ping relationship. That's entirely contrary to the point I was making - 
that the UAS need not be aware this is a ping, or event that there is 
such a thing as a ping. Rather, here "ping" is a usage of OPTIONS whose 
significance is only known to the UAC.

>>     - the benefit is limited because you are stuck
>>       with the way OPTIONS currently works, and with
>>       the variations in behavior that it allows.
> 
> Understood, but what is the problem with that?  Apparently, many felt it
> wasn't an issue since this is widely, inconsistently deployed

You are also stuck with the ambiguity in what response is returned in 
various situations, as well as the possibility that undesired 
capabilities might be returned, or the the device might decide to 
forward the request. Those are all part of accepting OPTIONS for what it 
currently is.

>>     - pinging this way may be more costly than
>>       necessary because of the extra work of applying
>>       policy, including capabilities, and the extra
>>       size of the message to carry the capabilities
> 
> Yeah, but our intent was that this would be done over a local link and
> probably only when the device is otherwise not generating traffic.  Perhaps
> the thing I believe is a non-issue is the composition and processing of
> those messages.  Surely folks are not going to construct that response based
> on querying a bunch of configuration information upon every request.  I'd
> assume the information is readily in hand and perhaps even sitting in a
> buffer read to be copied as-is.

If the thing you are pinging is a white box to you, then maybe you know 
all that. But in any sort of multi-vendor situation that is likely not 
to be the case.

For instance, if you send OPTIONS to a gateway, it *might* decide it 
should investigate its current resources before responding with its 
capabilities.

The boxes you are familiar may not work that way. But don't make that 
assumption about all boxes.

>> 2) define new standardized behavior for pinging,
>>     with roles for both the UAC and UAS, that is
>>     different from what OPTIONS does today.
> 
> We're not looking for something that is different, per se, but just to agree
> on behavior.  This behavior is in line with what 3261 says, but let's not do
> things that are all over the place.

As best I can tell, you still want some options to be treated as ping, 
while allowing others to be treated as capability query and handled 
differently. That is *new standardized behavior* in my book.

Alternately, maybe you are asking for OPTIONS, on all devices in all 
circumstances, to be more deterministic than it currently is, so that it 
is more suitable for ping usage while still being simultaneously 
suitable as a capability query. This is *still* a normative change, 
though a smaller one than the other.

PLEASE clarify which of these you want.

>>     + the behavior is not constrained by the existing
>>       definition of OPTIONS. It can be defined
>>       precisely for the purpose.
> 
> We definitely want to constrain behavior: that's a major reason preparing
> the draft.

You misunderstand me. Of course the behavior is constrained. Its just 
that there are two different behaviors:
- capability query behavior
- ping behavior

But these could be specified independently, so that a ping can have a 
more circumscribed behavior than the capability query behavior has.

>>     + the implementation can be optimized.
>>       the server can "fast path" it, and policy
>>       application can be simplified because little
>>       is disclosed.
> 
> It could be, but what I'm having trouble with is what I would do differently
> if it was a "ping" or not.  For example, if I build a GW and I received an
> OPTIONS addressed to sip:192.168.1.10, I would reply the same way every
> time: I'd assume it's likely a "ping".  And, that response should be a valid
> OPTIONS reply regardless of whether it's a "ping" or not.  That is, no
> different code path followed.  I would do the same thing if I were building
> an SBC: I'd assume the message is a "ping" and comply with this spec: a
> subset of what RFC 3261 describes.

> If the "ping" was directed to "sip:+19194762048@192.168.1.10", I might
> return something different, since there may be capabilities associated with
> a given port that is different than the GW as a whole.

Then you are asking that the OPTIONS sent to sip:192.168.1.10 might 
produce a different result than OPTIONS sent to 
sip:+19194762048@192.168.1.10.

That is of course *permitted* under the current definition of OPTIONS. 
But it isn't *required*.

And you want there to be specific semantics to the sip:192.168.1.10 
form. In the context of your GW example that makes a certain amount of 
sense. When the target is some other sort of device, say a phone, that 
may not make much sense.

My point is that if you want to ascribe specific semantics to options 
when the host is an ip address and the user part is missing, then that 
will be a standards action to accomplish, and then you will have to wait 
while it is implemented. And, I think you may have difficulty getting 
everyone to agree.

>>     - it will take longer to get specified because it
>>       must be standards track. (Note that this will be
>>       pretty much the same regardless of whether it is
>>       defined as new optional behavior of OPTIONS, or
>>       as a new method.)
> 
> Understood, which is why I didn't really want to introduce new
> functionality. 

But you *do* want to introduce new functionality:

You want to specify that all those implementations that do it a way 
different than what you propose should change to do it this way. That is 
new.

> I do not object to introducing a header, for example: that
> seems like a minimal and perhaps useful addition.
>  
>>     - it will take longer to deploy because both
>>       UAC and UAS must support it.
> 
> Exactly, which is why I think defining a whole new method like PING is out
> of the question.  SIP is too old at this point to introduce such new
> methods, in my opinion.  It would take years to approve and we'd have to
> live with both OPTIONS and PING for 20 years.

It will take no longer to introduce a PING method, and no more effort to 
implement, than to introduce a new header or a new option tag, or simply 
to document new behavior and get it approved as a *standard* and then 
get it widely implemented.

>> I guess there is a possible variant on (1):
>> also do a *clarification* of OPTIONS behavior in cases where there is
>> some ambiguity about what response code should be returned in
>> particular
>> circumstances. I would be ok with that *if* the clarification applied
>> to
>> all uses of OPTIONS. But once you start to do that you start to pick up
>> the negatives of (2).
> 
> I can't fix every problem with OPTIONS and I don't see a reason to expand
> this draft beyond that.  If we were introducing new functionality, I'd
> agree... but I'd prefer we didn't introduce new functionality.
> 
>> It would also be possible to do a combination of (1) and (2), where (2)
>> is preferred if both UAS supports it, with fallback to (1) for those
>> that don't.
> 
> What I think we should do is more along the lines of (1), with possible
> consideration of introducing a header or tag to serve as a hint as to the
> purpose of the OPTIONS message.  (I know that compels you to suggest that I
> might be bipolar, but I'm only trying recognize the fact that some find this
> useful.  Still, I'd expect the same processing on the box.)
>  
>> What I don't see is a way to avoid standards action and yet define
>> special optional behavior by the UAS that the UAC can request and know
>> was provided. This seems to be what you are seeking.
> 
> So, what's special?  We're constraining behavior, not expanding it.

Its *harder* to get a change deployed that constrains behavior than it 
is to deploy one that expands behavior!

Constraining behavior presents backward compatibility issues to 
everybody that was doing something that used to be valid and no longer 
is. Expanding behavior presents no backward compatibility issues.

	Thanks,
	Paul K

From christer.holmberg@ericsson.com  Thu Apr 29 12:58:44 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Paul E. Jones" <paulej@packetizer.com>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Thu, 29 Apr 2010 21:54:15 +0200
Thread-Topic: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

>>>That said, I can see where it might be useful to have a ping@example.com=
 type address for such things as IP phones or
>>>even SBCs.  In any case, the "ping" address should be an address that th=
e manufacturer discloses and the peer entity should be configured to
>>>use.
>>>
>>I'm getting confused - are you thinking the IP Address of the OPTIONS'
>>request-URI would be a *different* IP Address from the one other SIP
>>requests are sent to??  I didn't grok that from the draft at all, and
>>it would make things very complicated.
>
>No, I certainly didn't mean to suggest there would a different IP address.
>
>What I am referring to is the Request URI.  Previously, I was suggesting
>that the request URI would be addressed to the IP address if it's a ping,
>but would be addressed to a user if it wasn't.  But, you and Dean have
>convinced me that that's not always a good way to differentiate an OPTIONS
>"ping" from one that's an actual OPTIONS request.
>
>So, I removed the restriction from my local draft that specified what the
>Request URI looks like.  What we need to decide is exactly what we want to
>have as a differentiator?  A specific request URI, a tag, a header, etc.
>
>Having said that, I'm still favorable to sending an OPTIONS "ping" address=
ed
>to the IP address of the device (e.g., sip:192.168.1.10).  Whether that
>contains a "ping" header or tag or other is likely not going to make much
>difference to an SBC that does not receive other requests addressed to tha=
t
>address.

If the receiver is a proxy I don't think the IP address would work as a dif=
ferentiator, because according to RFC 3261 OPTIONS requests
sent to proxies are addressed by only using the host part. So, you would us=
e sip:192.168.1.10 also for a "normal" OPTIONS request.

Regards,

Christer=

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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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I'll be happy to be shown to be unduly pessimistic.
I agree that the empty accept ought to result in a response without a 
body. I just expect that isn't what will happen in practice.
Certainly implementations can then be called out for violation.

	Thanks,
	Paul

Paul E. Jones wrote:
> Paul,
> 
>>> The message body payload part is easy to get rid of - you just
>> include an empty Accept header field in the OPTIONS request.
>>
>> That *could* work, if the UAS understands and does it right.
> 
> If it didn't, wouldn't it be non-compliant with 3261?
>  
>> I'll venture to guess that most existing implementations would either:
>> a) return the SDP anyway (assuming there were prepared to return it.)
>> b) barf and return some error code, hopefully 406
>>
>> If (a) happened, you could file a complaint against the vendor of the
>> UAS. But I think you could make a good case that (b) is appropriate in
>> this case, as it would if you put an empty Accept in an INVITE.
> 
> Why would you argue that a device should do (b)?
> 
> Paul
> 
> 
> 

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To: Paul Kyzivat <pkyzivat@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Thu, 29 Apr 2010 22:13:00 +0200
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Hi,

>I'll be happy to be shown to be unduly pessimistic.
>I agree that the empty accept ought to result in a response without a
>body. I just expect that isn't what will happen in practice.

That may be true, but I don't think it should be a reason for not specifyin=
g a solution. After all, the protocol doesn't break even if the response do=
es contain a body.

But, so far we have been mostly talking about the content of the response. =
However, I think the content of the request is also important.

Assume I send an OPTIONS "ping" request, and don't include any capabilities=
 etc. Assuming there is no "ping" identifier, the receiver may actually thi=
nk that I don't support any capabilities, and may use that information e.g.=
 when making routing/forking decissions about requests.

Regards,

Christer



Paul E. Jones wrote:
> Paul,
>
>>> The message body payload part is easy to get rid of - you just
>> include an empty Accept header field in the OPTIONS request.
>>
>> That *could* work, if the UAS understands and does it right.
>
> If it didn't, wouldn't it be non-compliant with 3261?
>
>> I'll venture to guess that most existing implementations would either:
>> a) return the SDP anyway (assuming there were prepared to return it.)
>> b) barf and return some error code, hopefully 406
>>
>> If (a) happened, you could file a complaint against the vendor of the
>> UAS. But I think you could make a good case that (b) is appropriate in
>> this case, as it would if you put an empty Accept in an INVITE.
>
> Why would you argue that a device should do (b)?
>
> Paul
>
>
>=

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Milinski, Alexander \(NSN - DE/Munich\)'" <alexander.milinski@nsn.com>, "'ext Hadriel Kaplan'" <HKaplan@acmepacket.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org
Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Christer,

> OPTIONS is not the only method that is too "heavy" if sent too often.
> People have been using frequent re-registrations in order to do NAT
> keep-alives, and that's also bad. That is one reason we have defined
> the usage of STUN for keep-alives, because it's much more
> "lightweight".
> 
> So, I think it would be useful to know how often (the minimum value)
> these pings are expected to be sent.

Since this was intended for use between two devices that have some kind of
relationship, not merely transmitted anywhere on the Internet, we
recommended that the interval be left to the administrator.

If I were implementing it, I would not even send a ping if I knew there was
activity (i.e., I would only send one when I had not seen any other SIP
message for some specified period of time .. again, configurable).

Paul




From paulej@packetizer.com  Thu Apr 29 18:11:52 2010
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>
Date: Thu, 29 Apr 2010 21:11:14 -0400
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: [sipcore] SIP OPTIONS "ping"
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Guys,

I decided to start this new thread, because the previous one is proving to
be a bit challenging to respond to.

Let's back up and review where we are and what we're trying to accomplish.

Many devices today use OPTIONS as a "ping" method.  The reason is that there
was nothing better or more appropriate for an out-of-dialog means of
determining status of a peer SIP device.  These devices include PBXes, SBCs,
etc.

The problems are:
1) Response codes vary from one device to another (e.g., 600 vs. 503 vs.
486)
2) Nowhere is it documented that an OPTIONS can be used as an
application-level "ping"

Taking note of the current state of affairs and desiring to improve it
without changing SIP in any significant way, we prepared the OPTIONS "ping"
draft.

Is it a ping message or not?  The thread has focused heavily on the contents
of the message, but we're not proposing changes to the contents of a
message.  We only wish to focus on response codes: the contents of headers
and body were not subject to change.

We proposed sending OPTIONS "ping" messages to the IP address of the peer
device with a request URI of "sip:hostport" since that seemed most logical
to us.  Christer pointed out that that's the address used when addressing a
SIP proxy server.  I think that's OK, since we're not suggesting that the
proxy server actually alter its behavior.  That said, we would like to
propose use of 503 to indicate that the device is overloaded, rather than
600: that's what RFC 3261 says. But, note that we're not saying a 600 is
illegal if that were the appropriate code -- it just isn't in this case.

We also expect this to be used only between two devices that agree to do so.
The interval for messages I also wanted to keep locally configured.

It might be useful to indicate that the message was a "ping" -- I have no
objection.  But, I would not implement a different code path, personally.
That's why I felt it wasn't terribly important to include.

If we do nothing, the world will continue to use OPTIONS as they do today.
Introducing PING will take time to standardize and we still have to support
OPTIONS for a decade.  So, why bother?

My recommendation is that we not boil the ocean and try to create the most
architecturally pure solution.  Let's recognize the use of OPTIONS in this
way as legitimate, keep it peer-to-peer (maxforward=1), recommend response
codes, and expect a locally provisioned usage policy (e.g., message
frequency).

I've captured the groups' comments in this draft, though I'd like to
reinstate the paragraph relating to the SIP Request URI (sip:hostport):
http://hive.packetizer.com/users/paulej/internet-drafts/pre-draft-jones-sip-
options-ping-01.pdf

In any case, I'll go along with whatever the group wishes to do.
 
Paul



From HKaplan@acmepacket.com  Fri Apr 30 01:39:42 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Paul E. Jones" <paulej@packetizer.com>, 'Paul Kyzivat' <pkyzivat@cisco.com>, 'Christer Holmberg' <christer.holmberg@ericsson.com>
Date: Fri, 30 Apr 2010 04:39:11 -0400
Thread-Topic: Indicator for SIP OPTIONS "ping"
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: [sipcore] Indicator for SIP OPTIONS "ping"
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Hi Paul,
Good idea to reset.

Right, so one of the sub-threads previous was on the need or not for an ind=
icator.

In the snippet of your email below, you're saying the OPTIONS is no differe=
nt and the code path would be the same.  That's the part that confuses me. =
 If the code path is the same on the responder, such that it doesn't do any=
thing different than what it does today, then by definition it won't fix th=
e interop issues we've all seen in the wild.  The whole point of documentin=
g consistent behavior for response codes is to fix the interop issues, and =
that means the box getting the OPTIONS request has to know the sender expec=
ts the behavior of this document.  For the responder to know that, there ne=
eds to be something in the OPTIONS message to tell it that.  And for the bo=
x that sent the OPTIONS, it may also need to know the responder does the dr=
aft so that it can treat the response codes appropriately, by something in =
the response telling it so.

I don't think that means it needs to be a new Method name, because we're no=
t actually changing the OPTIONS from being an OPTIONS - using OPTIONS for t=
his purpose is in rfc3261, it just wasn't succinct enough to avoid interop =
issues.

And the advantage of continuing to use an OPTIONS is that if the receiver d=
oes NOT support this new draft, it will still "work" as before.  In that se=
nse using a new target URI username is probably not the best idea.  We coul=
d use a new options-tag in Supported, but since that would also still need =
something else in the message to say "I'm using this options-tag mechanism =
for this request", there's not much benefit to that.  So the easiest/simple=
st/most-likely-to-succeed path probably is Paul Kyzivat's suggestion of jus=
t defining a new header for both the request and response.=20

-hadriel

> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Thursday, April 29, 2010 9:11 PM
> Subject: SIP OPTIONS "ping"
>=20
> Is it a ping message or not?  The thread has focused heavily on the
> contents
> of the message, but we're not proposing changes to the contents of a
> message.  We only wish to focus on response codes: the contents of header=
s
> and body were not subject to change.
>=20
> We proposed sending OPTIONS "ping" messages to the IP address of the peer
> device with a request URI of "sip:hostport" since that seemed most logica=
l
> to us.  Christer pointed out that that's the address used when addressing
> a
> SIP proxy server.  I think that's OK, since we're not suggesting that the
> proxy server actually alter its behavior.  That said, we would like to
> propose use of 503 to indicate that the device is overloaded, rather than
> 600: that's what RFC 3261 says. But, note that we're not saying a 600 is
> illegal if that were the appropriate code -- it just isn't in this case.
>=20
> We also expect this to be used only between two devices that agree to do
> so.
> The interval for messages I also wanted to keep locally configured.
>=20
> It might be useful to indicate that the message was a "ping" -- I have no
> objection.  But, I would not implement a different code path, personally.
> That's why I felt it wasn't terribly important to include.

From pkyzivat@cisco.com  Fri Apr 30 07:44:06 2010
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Indicator for SIP OPTIONS "ping"
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Hadriel Kaplan wrote:
> Hi Paul,
> Good idea to reset.
> 
> Right, so one of the sub-threads previous was on the need or not for an indicator.
> 
> In the snippet of your email below, you're saying the OPTIONS is no different and the code path would be the same.  That's the part that confuses me.  If the code path is the same on the responder, such that it doesn't do anything different than what it does today, then by definition it won't fix the interop issues we've all seen in the wild.  The whole point of documenting consistent behavior for response codes is to fix the interop issues, and that means the box getting the OPTIONS request has to know the sender expects the behavior of this document.  For the responder to know that, there needs to be something in the OPTIONS message to tell it that.  And for the box that sent the OPTIONS, it may also need to know the responder does the draft so that it can treat the response codes appropriately, by something in the response telling it so.

Right!

> I don't think that means it needs to be a new Method name, because we're not actually changing the OPTIONS from being an OPTIONS - using OPTIONS for this purpose is in rfc3261, it just wasn't succinct enough to avoid interop issues.

If there is no intent to have *different* behavior for the ping usage, 
then there is also no reason to indicate in the OPTIONS request that it 
is being used as a ping. Nor is there any need to have a side agreement 
between the UAC and UAS that the UAC will be pinging the UAS.

> And the advantage of continuing to use an OPTIONS is that if the receiver does NOT support this new draft, it will still "work" as before.  In that sense using a new target URI username is probably not the best idea.  We could use a new options-tag in Supported, but since that would also still need something else in the message to say "I'm using this options-tag mechanism for this request", there's not much benefit to that.  So the easiest/simplest/most-likely-to-succeed path probably is Paul Kyzivat's suggestion of just defining a new header for both the request and response. 

There really isn't any need to put *anything* new into the request.
All that is really needed is a header that says:

"This response complies with the *new improved* OPTIONS definition."

Unfortunately, just to accomplish that small step will still require a 
standards track draft to be approved. And if we are going to go to that 
much trouble, would it be worthwhile to do a separate ping mechanism?

I'm not advocating that we *should* do that - just asking.

If the clarified OPTIONS is what is desired, then maybe it isn't just 
about OPTIONS - is is really about more specific use of error codes for 
*all* requests?

	Thanks,
	Paul

> -hadriel
> 
>> -----Original Message-----
>> From: Paul E. Jones [mailto:paulej@packetizer.com]
>> Sent: Thursday, April 29, 2010 9:11 PM
>> Subject: SIP OPTIONS "ping"
>>
>> Is it a ping message or not?  The thread has focused heavily on the
>> contents
>> of the message, but we're not proposing changes to the contents of a
>> message.  We only wish to focus on response codes: the contents of headers
>> and body were not subject to change.
>>
>> We proposed sending OPTIONS "ping" messages to the IP address of the peer
>> device with a request URI of "sip:hostport" since that seemed most logical
>> to us.  Christer pointed out that that's the address used when addressing
>> a
>> SIP proxy server.  I think that's OK, since we're not suggesting that the
>> proxy server actually alter its behavior.  That said, we would like to
>> propose use of 503 to indicate that the device is overloaded, rather than
>> 600: that's what RFC 3261 says. But, note that we're not saying a 600 is
>> illegal if that were the appropriate code -- it just isn't in this case.
>>
>> We also expect this to be used only between two devices that agree to do
>> so.
>> The interval for messages I also wanted to keep locally configured.
>>
>> It might be useful to indicate that the message was a "ping" -- I have no
>> objection.  But, I would not implement a different code path, personally.
>> That's why I felt it wasn't terribly important to include.
> 

From rbarnes@bbn.com  Fri Apr 30 09:39:33 2010
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Subject: [sipcore] ESW7: Detailed agenda and reminder to register
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Dear colleagues,

This is an update to let you know that the detailed agenda has been  
posted for the upcoming Emergency Services Workshop:
<http://www.emergency-services-coordination.info/esw7.html>

Also, just a reminder that if you are planning to attend, please fill  
out the registration form on the website:
<https://intranet.umiacs.umd.edu/conferences/esc2010/reg.htm>

Thanks for your interest,
The ESW Coordination Team

From fluffy@cisco.com  Fri Apr 30 12:22:31 2010
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS "ping"
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Thanks for the reset.

I think the reasons this thread has been so unproductive is we are =
unclear on what the problem is we are trying to solve. "We need a way to =
do PING" is not a problem. It is a solution to some problem. If the =
problem is figure out which of multiple next hops to use based on load =
balancing and if the hop is dead or alive, well the PING approach here =
will fail miserably (see the overload discussion and simulations) and we =
have WG going to the IESG to try and solve that problem.=20

Let me make it very clear I am against the working group working on this =
draft until someone can describe what the problem is. The current draft =
hand waves at a few problems - does not say what it is trying to do. =
Then proposes some syntax with no backing semantics that would help =
solve any of the problems it mentions.=20

Cullen

On Apr 29, 2010, at 7:11 PM, Paul E. Jones wrote:

> Guys,
>=20
> I decided to start this new thread, because the previous one is =
proving to
> be a bit challenging to respond to.
>=20
> Let's back up and review where we are and what we're trying to =
accomplish.
>=20
> Many devices today use OPTIONS as a "ping" method.  The reason is that =
there
> was nothing better or more appropriate for an out-of-dialog means of
> determining status of a peer SIP device.  These devices include PBXes, =
SBCs,
> etc.
>=20
> The problems are:
> 1) Response codes vary from one device to another (e.g., 600 vs. 503 =
vs.
> 486)
> 2) Nowhere is it documented that an OPTIONS can be used as an
> application-level "ping"
>=20
> Taking note of the current state of affairs and desiring to improve it
> without changing SIP in any significant way, we prepared the OPTIONS =
"ping"
> draft.
>=20
> Is it a ping message or not?  The thread has focused heavily on the =
contents
> of the message, but we're not proposing changes to the contents of a
> message.  We only wish to focus on response codes: the contents of =
headers
> and body were not subject to change.
>=20
> We proposed sending OPTIONS "ping" messages to the IP address of the =
peer
> device with a request URI of "sip:hostport" since that seemed most =
logical
> to us.  Christer pointed out that that's the address used when =
addressing a
> SIP proxy server.  I think that's OK, since we're not suggesting that =
the
> proxy server actually alter its behavior.  That said, we would like to
> propose use of 503 to indicate that the device is overloaded, rather =
than
> 600: that's what RFC 3261 says. But, note that we're not saying a 600 =
is
> illegal if that were the appropriate code -- it just isn't in this =
case.
>=20
> We also expect this to be used only between two devices that agree to =
do so.
> The interval for messages I also wanted to keep locally configured.
>=20
> It might be useful to indicate that the message was a "ping" -- I have =
no
> objection.  But, I would not implement a different code path, =
personally.
> That's why I felt it wasn't terribly important to include.
>=20
> If we do nothing, the world will continue to use OPTIONS as they do =
today.
> Introducing PING will take time to standardize and we still have to =
support
> OPTIONS for a decade.  So, why bother?
>=20
> My recommendation is that we not boil the ocean and try to create the =
most
> architecturally pure solution.  Let's recognize the use of OPTIONS in =
this
> way as legitimate, keep it peer-to-peer (maxforward=3D1), recommend =
response
> codes, and expect a locally provisioned usage policy (e.g., message
> frequency).
>=20
> I've captured the groups' comments in this draft, though I'd like to
> reinstate the paragraph relating to the SIP Request URI =
(sip:hostport):
> =
http://hive.packetizer.com/users/paulej/internet-drafts/pre-draft-jones-si=
p-
> options-ping-01.pdf
>=20
> In any case, I'll go along with whatever the group wishes to do.
>=20
> Paul
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




From adam@estacado.net  Fri Apr 30 14:59:23 2010
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Subject: [sipcore] Call for Consensus: INFO Specification and Congestion
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[as chair]

During IESG review of the INFO specification, several members expressed 
concerns about the congestion properties of INFO. The draft authors and 
IESG worked together to craft text that satisfies these concerns.

Since the resulting text contains normative language, we are bringing 
the proposal to the SIPCORE working group for a 2-week consensus call. 
Please comment on the following proposed text before Friday, May 14th. 
If you agree with the proposal, please send a short response indicating 
that fact.

"10.9.  Rate and size of INFO Requests

     "INFO messages differ from many other sorts of SIP messages in that
      they carry application information, and the size and rate of the
      INFO message is directly determined by the application. This can
      cause application information traffic to interfere with other traffic
      on that infrastructure, or to self-interfere when data rates become
      too high.

     "A designer of an INFO package, and the application that uses it,
      need to consider the impact that the size and the rate of the INFO
      messages have on the network and on other traffic, since it normally
      cannot be ensured that INFO messages will be carried over a
      congestion-controlled transport protocol end-to-end. Even if an
      INFO message is sent over a such a transport protocol, a downstream
      SIP entity might forward the message over a transport protocol that
      does not provide congestion control.

     "Therefore, an application SHOULD generally have at most one 
outstanding INFO message.
      The guidelines in RFC3261 Section 18 MUST be followed for sending 
and receiving INFO
      requests and responses, such as the guideline to not exceed messages
      sizes of 1300 bytes when the path MTU is unknown. RFC5405 provides
      additional guidelines for applications using the User Datagram
      Protocol (UDP) that may be useful background reading."

Thanks.

/a

From stpeter@stpeter.im  Fri Apr 30 21:45:50 2010
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Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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On 4/30/10 3:59 PM, Adam Roach wrote:
> [as chair]
>=20
> During IESG review of the INFO specification, several members expressed=

> concerns about the congestion properties of INFO. The draft authors and=

> IESG worked together to craft text that satisfies these concerns.
>=20
> Since the resulting text contains normative language, we are bringing
> the proposal to the SIPCORE working group for a 2-week consensus call.
> Please comment on the following proposed text before Friday, May 14th.
> If you agree with the proposal, please send a short response indicating=

> that fact.
>=20
> "10.9.  Rate and size of INFO Requests
>=20
>     "INFO messages differ from many other sorts of SIP messages in that=

>      they carry application information, and the size and rate of the
>      INFO message is directly determined by the application. This can
>      cause application information traffic to interfere with other traf=
fic
>      on that infrastructure, or to self-interfere when data rates becom=
e
>      too high.
>=20
>     "A designer of an INFO package, and the application that uses it,
>      need to consider the impact that the size and the rate of the INFO=

>      messages have on the network and on other traffic, since it normal=
ly
>      cannot be ensured that INFO messages will be carried over a
>      congestion-controlled transport protocol end-to-end. Even if an
>      INFO message is sent over a such a transport protocol, a downstrea=
m
>      SIP entity might forward the message over a transport protocol tha=
t
>      does not provide congestion control.
>=20
>     "Therefore, an application SHOULD generally have at most one
> outstanding INFO message.
>      The guidelines in RFC3261 Section 18 MUST be followed for sending
> and receiving INFO
>      requests and responses, such as the guideline to not exceed messag=
es
>      sizes of 1300 bytes when the path MTU is unknown. RFC5405 provides=

>      additional guidelines for applications using the User Datagram
>      Protocol (UDP) that may be useful background reading."

Seems good to me.

Peter

--=20
Peter Saint-Andre
https://stpeter.im/




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