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Subject: Re: [sipcore] Outbound clarification
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Hi Dale,

I thought that too, but based on the definition of the ;ob parameter, it
has no sense in a Contact header of a REGISTER request, since such request
does never belong to a dialog.

'''
The "ob" parameter is a SIP URI parameter that has a different meaning
depending on context. In a Path header field value, it is used by the first
edge proxy to indicate that a flow token was added to the URI. In a Contact
or Route header field value, it indicates that the UA would like other
requests in the same dialog to be routed over the same flow.
'''




2013/6/28 Dale R. Worley <worley@ariadne.com>

> > From: JosC) Luis MillC!n <jmillan@aliax.net>
> >
> > May it mean that the GRUU was obtained from a REGISTER request which
> > Contact header contained the ;ob parameter?
>
> Yes, I believe that is what it means.  Every GRUU refers to an
> underlying Contact address, which is where requests to that GRUU will
> be routed to.  That Contact address is what was used in the REGISTER
> request to obtain the GRUU.  The question seems to be, "Does this GRUU
> refer to a Contact address that includes an "ob" parameter?"
>
> Dale
>



--=20
Jos=C3=A9 Luis Mill=C3=A1n

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<div dir=3D"ltr"><div style>Hi Dale,</div><div style><br></div><div style>I=
 thought that too, but based on the definition of the ;ob parameter, it has=
 no sense in a Contact header of a REGISTER request, since such request doe=
s never belong to a dialog.</div>
<div><br></div><div>&#39;&#39;&#39;</div>The &quot;ob&quot; parameter is a =
SIP URI parameter that has a different meaning depending on context. In a P=
ath header field value, it is used by the first edge proxy to indicate that=
 a flow token was added to the URI. In a Contact or Route header field valu=
e, it indicates that the UA would like other requests in the same dialog to=
 be routed over the same flow.<div>
&#39;&#39;&#39;</div><div><br></div><div><br></div></div><div class=3D"gmai=
l_extra"><br><br><div class=3D"gmail_quote">2013/6/28 Dale R. Worley <span =
dir=3D"ltr">&lt;<a href=3D"mailto:worley@ariadne.com" target=3D"_blank">wor=
ley@ariadne.com</a>&gt;</span><br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">&gt; From: JosC) Luis MillC!n &lt;<a href=3D=
"mailto:jmillan@aliax.net">jmillan@aliax.net</a>&gt;<br>
<div class=3D"im">&gt;<br>
&gt; May it mean that the GRUU was obtained from a REGISTER request which<b=
r>
&gt; Contact header contained the ;ob parameter?<br>
<br>
</div>Yes, I believe that is what it means. =C2=A0Every GRUU refers to an<b=
r>
underlying Contact address, which is where requests to that GRUU will<br>
be routed to. =C2=A0That Contact address is what was used in the REGISTER<b=
r>
request to obtain the GRUU. =C2=A0The question seems to be, &quot;Does this=
 GRUU<br>
refer to a Contact address that includes an &quot;ob&quot; parameter?&quot;=
<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Dale<br>
</font></span></blockquote></div><br><br clear=3D"all"><div><br></div>-- <b=
r>Jos=C3=A9 Luis Mill=C3=A1n
</div>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 3 Jul 2013 02:26:53 +0200
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>, Jose Luis Millan <jmillan@aliax.net>
Subject: Re: [sipcore] Outbound clarification
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2013/6/28 Dale R. Worley <worley@ariadne.com>:
> Yes, I believe that is what it means.  Every GRUU refers to an
> underlying Contact address, which is where requests to that GRUU will
> be routed to.  That Contact address is what was used in the REGISTER
> request to obtain the GRUU.  The question seems to be, "Does this GRUU
> refer to a Contact address that includes an "ob" parameter?"

The problem is:

alice ---- Proxy+Registrar ---- bob

- alice has a GRUU registration.
- alice calls bob with ;gr in Contact.
- Let's suppose P+R adds Outbound flow token in R-R.
- Call is established.
- Later bob sends BYE.
- Such a BYE arrives to P+R with a Route header containg an outbound
flow token and with a RURI containing ;gr.

How should P+R route the BYE? by using Outbound (looking for the
connection associated to the flow token in the Route header)? or by
performing GRUU (replacing the RURI with the real Contact? In the
former, if the TCP connection of alice was re-established during the
dialog, the BYE could not be sent.



--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 5 Jul 2013 11:27:02 +0200
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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2013/6/27 Paul Kyzivat <pkyzivat@alum.mit.edu>:
> The simple answer would be that websocket proxy handling a dialog
> establishing request should *always* R-R. But there has been an assertion
> that there may be some sip-websocket UACs that are also capable of being
> sip-websocket servers. In that case, it would be desirable for the proxy =
to
> *not* R-R, so that it won't be in the signaling path for the entire dialo=
g.
> But how would the proxy know that it is safe to not R-R?
>
> One thing that comes to mind for me is that the UAC could decide whether =
or
> not to tag its contact URI with ";gr". If it were so tagged, then that sa=
ys
> it is globally routable, so it would be safe for the proxy to bypass the
> R-R. If the contact isn't a gruu, then the proxy would decide that it mus=
t
> R-R.

Hi, let me show a similar example:

  alice --- (SCTP) --- Proxy --- (UDP) --- bob

- alice calls to bob with Contact SCTP URI.

- bob does not implement SCTP.

- if Proxy does not R-R then bob has no way to contact alice for
sending her in-dialog requests.

Now my question: how does the proxy know whether it must do R-R or not
in this scenario?

IMHO there is no reply for this question, and anyhow, this is not an
issue related to SIP-over-WS. So IMHO there is no need to address this
issue into the draft (instead it should be covered in some other work
like RFC 5626bis or RFC 5627bis). Am I wrong?

Whether the proxy does R-R or not if decided by local policy. It's
safe for a proxy to always add R-R in connections from clients (via
UDP, TCP, WS transports). And as Paul pointed out, the proxy could
decide not to add R-R in case the UA's Contact has ;gr so R-R is not
needed.



Please let me insist on it a bit more:

>  But there has been an assertion
> that there may be some sip-websocket UACs that are also capable of being
> sip-websocket servers. In that case, it would be desirable for the proxy =
to
> *not* R-R, so that it won't be in the signaling path for the entire dialo=
g.
> But how would the proxy know that it is safe to not R-R?

Any TCP SIP UA is capable of listening for incoming connections but
that does not mean that the proxy should not R-R. If the UA is behind
NAT then RFC 5626 must be used (I avoid here proprietary similar
mechanisms), and RFC 5626 *alreay* mandates R-R for such a mechanism
to work. If GRUU is also in use then the proxy can decide not to R-R
(not needed since incoming requests will arrive via the registration
path which will of course include Outbound stuff).


So honestly, I don't understand why draft-ietf-sipcore-sip-websocket
should address issues that are not different than when using other
transports, and that are already covered by other specifications.
Correct me if I'm wrong.


Best regards.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 5 Jul 2013 11:36:51 +0200
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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Please, let me insist even a bit more on it:



2013/7/5 I=C3=B1aki Baz Castillo <ibc@aliax.net>:
> Any TCP SIP UA is capable of listening for incoming connections but
> that does not mean that the proxy should not R-R. If the UA is behind
> NAT then RFC 5626 must be used (I avoid here proprietary similar
> mechanisms), and RFC 5626 *alreay* mandates R-R for such a mechanism
> to work. If GRUU is also in use then the proxy can decide not to R-R
> (not needed since incoming requests will arrive via the registration
> path which will of course include Outbound stuff).



a) If the SIP-WS UA runs in a browser then the website developer can
make usage of RFC 5626 so the UA *requests* Outbound usage to its
proxy.

b) If the SIP-WS UA runs in a desktop and is both a SIP WS Client and
SIP WS Server (and let's assume it has public IP) then it can avoid
requesting Outbound to the proxy. If the proxy does R-R it is because
the proxy wants to be in the patch of the dialog (local policy) and
not because it is required for the dialog to work.


Of course, in case b), if the peer does not speak WS and the proxy
does not R-R here we have a problem, but this is *exactly* the same
problem as if the caller was a UA using SCTP in its Contact URI. So
again, this is about local policy in the proxy whether to decide it
does R-R or not.

I don't find "Record-Route" word in RFC 4168 so I expect there should
be no "MUST" keyword about Record-Route in
draft-ietc-sipcore-sip-websocket. Let me know if I'm wrong.



Summarizing:

- This is not an issue introduced by this new WS transport.
- The issue is not new.
- There are already standard mechanism to make it to work (the
*client* can request them to the proxy !).
- I see no reason at all to mandate nothing in the sip-ws draft about this.
- Guidelines section in the draft already covers it in an informative way.



Hope I am right.

Best regards.



--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From ibc@aliax.net  Fri Jul  5 05:55:36 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 5 Jul 2013 14:55:09 +0200
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Subject: [sipcore] Need RFC 5627-bis (GRUU improvements and clarifications, and RFC 3261 update)
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Hi, after leading a lot with Outbound and GRUU I think there are well
known problems and specification issues that IMHO should be addressed
in some kind of RFC 5627-bis.


The problem
------------------

alice --- OP --- Registrar --- bob

- OP is an Outbound EDGE Proxy.
- Registrar is GRUU capable.

Flow showing the problem:

- alice registers requesting GRUU.
- OP adds Path.
- alice obtains a GRUU URI.

case A)

1) alice calls bob by using her GRUU in Contact. alice also requests
Outbound usage by adding ;ob in Contact of INVITE.

2) OP adds Record-Route.

3) OP routes to registrar which adds Record-Route and routes to Bob.

4) Bob sends INFO to alice.

5) Registrar receives INFO with GRUU RURI (;gr), removes Route
pointing to it, and still has Route with Outbound flow token
identifier pointing to OP.

6) Registrar decides to route based on RURI GRUU so performs lookup
and obtains the binding for such a GRUU URI, and of course adds Route
mirroring the registration Path.

7) INFO arrives to OP with Route belonging to the dialog (with
Outbound flow identifier) and *also* Route produced by the Path.
Typically this means "duplicated  route set". So it could fail routing
the request (loop), and OF COURSE, Route set is broken (RFC 3261)
since Route set has been modified within a dialog.


case B)

1) alice calls bob by using her GRUU in Contact. alice does NOT
request Outbound usage (she does NOT add ;ob in Contact of INVITE)

2) OP does NOT add R-R.

3) OP routes to registrar which adds Record-Route and routes to Bob.

4) Bob sends INFO to alice.

5) Registrar receives INFO with GRUU RURI (;gr), removes Route pointing to =
it.

6) Registrar routes based on GRUU so performs lookup and obtains the
binding for such a GRUU URI, and of course adds Route mirroring the
registration Path.

7) INFO arrives to OP with Route produced by the registration Path. OP
will be able to route the request, of course, but Route set has also
been modified (breaks RFC 3261).

8) Now alice wants to send BYE to bob. It just adds to the request the
Route mirroring R-R received in the 200 for the initial INVITE, which
just contains the Route pointing to the registrar (since OP did not
R-R).

9) OP receives an in-dialog request *without* a Route pointing to it.
In several proxy configurations this request would be rejected (why
does the in-dialog request arrive to OP if it did not R-R?). One could
argue "alice should send BYE to Registrar by following the Route set
rules", but this is not how things work in Outbound, even less if the
transport is WebSocket and the UA can just talk with OP).



In both cases Route set is modified during a dialog and breaks RFC
3261. My opinion: This is GOOD since it makes possible for the dialog
to continue alive even if the TCP connection between alice and OP was
closed and re-opened (new Outbound flow token identifier, refreshed in
a new registration so new Path).

The "nice" solution to avoid problem in case a) dot 7 is:

If a Registrar receives an in-dialog request with a GRUU RURI pointing
to it, it MUST remove all the Route header present in the request (and
ignore them), and then do lookup, replace RURI with the real binding
and add Route mirroring registration Path values. After lot of
thinking this seems to be the best and only? solution for GRUU to
properly work for in-dialog requests.


So I propose a new work RFC 5627-bis to clarify all of this, and also
to "update RFC 3261" stating that dialog Route set can be modified if
GRUU is used as Contact by any of the peers.


Thoughts?

Best regards.





--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From pkyzivat@alum.mit.edu  Fri Jul  5 08:53:46 2013
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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I don't claim to have the answer here, but the discussion is helpful.

The analogy to other transports doesn't hold wrt TCP/UDP because until 
now both TCP and UDP support was *required*. There is a better analogy 
to IPv4 vs. IPv6.

But in general all of these things end up being treated as examples or 
best practices rather than normative requirements. You make a good case 
for leaving it at that.

	Thanks,
	Paul


On 7/5/13 5:36 AM, Iñaki Baz Castillo wrote:
> Please, let me insist even a bit more on it:
>
>
>
> 2013/7/5 Iñaki Baz Castillo <ibc@aliax.net>:
>> Any TCP SIP UA is capable of listening for incoming connections but
>> that does not mean that the proxy should not R-R. If the UA is behind
>> NAT then RFC 5626 must be used (I avoid here proprietary similar
>> mechanisms), and RFC 5626 *alreay* mandates R-R for such a mechanism
>> to work. If GRUU is also in use then the proxy can decide not to R-R
>> (not needed since incoming requests will arrive via the registration
>> path which will of course include Outbound stuff).
>
>
>
> a) If the SIP-WS UA runs in a browser then the website developer can
> make usage of RFC 5626 so the UA *requests* Outbound usage to its
> proxy.
>
> b) If the SIP-WS UA runs in a desktop and is both a SIP WS Client and
> SIP WS Server (and let's assume it has public IP) then it can avoid
> requesting Outbound to the proxy. If the proxy does R-R it is because
> the proxy wants to be in the patch of the dialog (local policy) and
> not because it is required for the dialog to work.
>
>
> Of course, in case b), if the peer does not speak WS and the proxy
> does not R-R here we have a problem, but this is *exactly* the same
> problem as if the caller was a UA using SCTP in its Contact URI. So
> again, this is about local policy in the proxy whether to decide it
> does R-R or not.
>
> I don't find "Record-Route" word in RFC 4168 so I expect there should
> be no "MUST" keyword about Record-Route in
> draft-ietc-sipcore-sip-websocket. Let me know if I'm wrong.
>
>
>
> Summarizing:
>
> - This is not an issue introduced by this new WS transport.
> - The issue is not new.
> - There are already standard mechanism to make it to work (the
> *client* can request them to the proxy !).
> - I see no reason at all to mandate nothing in the sip-ws draft about this.
> - Guidelines section in the draft already covers it in an informative way.
>
>
>
> Hope I am right.
>
> Best regards.
>
>
>
> --
> Iñaki Baz Castillo
> <ibc@aliax.net>
>


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Subject: [sipcore] I-D Action: draft-ietf-sipcore-rfc4244bis-callflows-04.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Session Initiation Protocol Core Working =
Group of the IETF.

	Title           : Session Initiation Protocol (SIP) History-Info Header Ca=
ll Flow Examples
	Author(s)       : Mary Barnes
                          Francois Audet
                          Shida Schubert
                          Hans Erik van Elburg
                          Christer Holmberg
	Filename        : draft-ietf-sipcore-rfc4244bis-callflows-04.txt
	Pages           : 46
	Date            : 2013-07-08

Abstract:
   This document describes use cases and documents call flows which
   require the History-Info header field to capture the Request-URIs as
   a Session Initiation Protocol (SIP) Request is retargeted.  The use
   cases are described along with the corresponding call flow diagrams
   and messaging details.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc4244bis-callflows

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-rfc4244bis-callflows-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-rfc4244bis-callflows-=
04


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] Need RFC 5627-bis (GRUU improvements and clarifications, and RFC 3261 update)
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--001a11c2012cb63b7a04e110681c
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I do agree with I=C3=B1aki that this scenario, which takes more relevance w=
ith
the time, requires a clarification.

RFC 5627 Section 6.1 states that in order to avoid the exposed spiral case,
the Path URI Must be discarded when the request contains Route header/s,
but in fact, the Path determines the right route to the UA.

'''

Special considerations apply to the processing of any Path headers
stored in the registration (see RFC 3327
<http://tools.ietf.org/html/rfc3327> [3
<http://tools.ietf.org/html/rfc5627#ref-3>]).  If the received
request has Route header field values beyond the one pointing to the

authoritative proxy itself (this will happen when the request is a
mid-dialog request), the Path URI MUST be discarded.  This is
permitted by RFC 3327 <http://tools.ietf.org/html/rfc3327> [3
<http://tools.ietf.org/html/rfc5627#ref-3>] as a matter of local
policy; usage of GRUUs
will require this policy in order to avoid call spirals and likely
call failures.

'''

While RFC 5626 Section 7 states near the contrary. The proxy MUST send the
request over the flow determined by the Path header:

'''

The proxy uses the next-hop target of the message and the value of
any stored Path header field vector in the registration binding to
decide how to forward and populate the Route header in the request.
If the proxy is co-located with the registrar and stored information
about the flow to the UA that created the binding, then the proxy
MUST send the request over the same 'logical flow' saved with the
binding, since that flow is known to deliver data to the specific
target UA instance's network flow that was saved with the binding.

'''



2013/7/5 I=C3=B1aki Baz Castillo <ibc@aliax.net>

> Hi, after leading a lot with Outbound and GRUU I think there are well
> known problems and specification issues that IMHO should be addressed
> in some kind of RFC 5627-bis.
>
>
> The problem
> ------------------
>
> alice --- OP --- Registrar --- bob
>
> - OP is an Outbound EDGE Proxy.
> - Registrar is GRUU capable.
>
> Flow showing the problem:
>
> - alice registers requesting GRUU.
> - OP adds Path.
> - alice obtains a GRUU URI.
>
> case A)
>
> 1) alice calls bob by using her GRUU in Contact. alice also requests
> Outbound usage by adding ;ob in Contact of INVITE.
>
> 2) OP adds Record-Route.
>
> 3) OP routes to registrar which adds Record-Route and routes to Bob.
>
> 4) Bob sends INFO to alice.
>
> 5) Registrar receives INFO with GRUU RURI (;gr), removes Route
> pointing to it, and still has Route with Outbound flow token
> identifier pointing to OP.
>
> 6) Registrar decides to route based on RURI GRUU so performs lookup
> and obtains the binding for such a GRUU URI, and of course adds Route
> mirroring the registration Path.
>
> 7) INFO arrives to OP with Route belonging to the dialog (with
> Outbound flow identifier) and *also* Route produced by the Path.
> Typically this means "duplicated  route set". So it could fail routing
> the request (loop), and OF COURSE, Route set is broken (RFC 3261)
> since Route set has been modified within a dialog.
>
>
> case B)
>
> 1) alice calls bob by using her GRUU in Contact. alice does NOT
> request Outbound usage (she does NOT add ;ob in Contact of INVITE)
>
> 2) OP does NOT add R-R.
>
> 3) OP routes to registrar which adds Record-Route and routes to Bob.
>
> 4) Bob sends INFO to alice.
>
> 5) Registrar receives INFO with GRUU RURI (;gr), removes Route pointing t=
o
> it.
>
> 6) Registrar routes based on GRUU so performs lookup and obtains the
> binding for such a GRUU URI, and of course adds Route mirroring the
> registration Path.
>
> 7) INFO arrives to OP with Route produced by the registration Path. OP
> will be able to route the request, of course, but Route set has also
> been modified (breaks RFC 3261).
>
> 8) Now alice wants to send BYE to bob. It just adds to the request the
> Route mirroring R-R received in the 200 for the initial INVITE, which
> just contains the Route pointing to the registrar (since OP did not
> R-R).
>
> 9) OP receives an in-dialog request *without* a Route pointing to it.
> In several proxy configurations this request would be rejected (why
> does the in-dialog request arrive to OP if it did not R-R?). One could
> argue "alice should send BYE to Registrar by following the Route set
> rules", but this is not how things work in Outbound, even less if the
> transport is WebSocket and the UA can just talk with OP).
>
>
>
> In both cases Route set is modified during a dialog and breaks RFC
> 3261. My opinion: This is GOOD since it makes possible for the dialog
> to continue alive even if the TCP connection between alice and OP was
> closed and re-opened (new Outbound flow token identifier, refreshed in
> a new registration so new Path).
>
> The "nice" solution to avoid problem in case a) dot 7 is:
>
> If a Registrar receives an in-dialog request with a GRUU RURI pointing
> to it, it MUST remove all the Route header present in the request (and
> ignore them), and then do lookup, replace RURI with the real binding
> and add Route mirroring registration Path values. After lot of
> thinking this seems to be the best and only? solution for GRUU to
> properly work for in-dialog requests.
>
>
> So I propose a new work RFC 5627-bis to clarify all of this, and also
> to "update RFC 3261" stating that dialog Route set can be modified if
> GRUU is used as Contact by any of the peers.
>
>
> Thoughts?
>
> Best regards.
>
>
>
>
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>



--=20
Jos=C3=A9 Luis Mill=C3=A1n

--001a11c2012cb63b7a04e110681c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I do agree with I=C3=B1aki that this scenario, which takes=
 more relevance with the time, requires a clarification.<div><br></div><div=
>RFC 5627 Section 6.1 states that in order to avoid the exposed spiral case=
, the Path URI Must be discarded when the request contains Route header/s, =
but in fact, the Path determines the right route to the UA.</div>
<div><br></div><div>&#39;&#39;&#39;</div><div><pre class=3D"" style=3D"font=
-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">Special consid=
erations apply to the processing of any Path headers
stored in the registration (see <a href=3D"http://tools.ietf.org/html/rfc33=
27">RFC 3327</a> [<a href=3D"http://tools.ietf.org/html/rfc5627#ref-3" titl=
e=3D"&quot;Session Initiation Protocol (SIP) Extension Header Field for Reg=
istering Non-Adjacent Contacts&quot;">3</a>]).  If the received
request has Route header field values beyond the one pointing to the
</pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0=
px;color:rgb(0,0,0)">authoritative proxy itself (this will happen when the =
request is a
mid-dialog request), the Path URI MUST be discarded.  This is
permitted by <a href=3D"http://tools.ietf.org/html/rfc3327">RFC 3327</a> [<=
a href=3D"http://tools.ietf.org/html/rfc5627#ref-3" title=3D"&quot;Session =
Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjace=
nt Contacts&quot;">3</a>] as a matter of local policy; usage of GRUUs
will require this policy in order to avoid call spirals and likely
call failures.</pre></div><div>&#39;&#39;&#39;</div><div><br></div><div>Whi=
le RFC 5626 Section 7 states near the contrary. The proxy MUST send the req=
uest over the flow determined by the Path header:</div><div><br></div>
<div>&#39;&#39;&#39;</div><div><pre class=3D"" style=3D"font-size:1em;margi=
n-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><pre class=3D"" style=3D"font=
-size:1em;margin-top:0px;margin-bottom:0px">The proxy uses the next-hop tar=
get of the message and the value of
any stored Path header field vector in the registration binding to
decide how to forward and populate the Route header in the request.
If the proxy is co-located with the registrar and stored information
about the flow to the UA that created the binding, then the proxy
MUST send the request over the same &#39;logical flow&#39; saved with the
binding, since that flow is known to deliver data to the specific
target UA instance&#39;s network flow that was saved with the binding.</pre=
></pre></div><div>&#39;&#39;&#39;</div><div><br></div></div><div class=3D"g=
mail_extra"><br><br><div class=3D"gmail_quote">2013/7/5 I=C3=B1aki Baz Cast=
illo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blan=
k">ibc@aliax.net</a>&gt;</span><br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi, after leading a lot with Outbound and GR=
UU I think there are well<br>
known problems and specification issues that IMHO should be addressed<br>
in some kind of RFC 5627-bis.<br>
<br>
<br>
The problem<br>
------------------<br>
<br>
alice --- OP --- Registrar --- bob<br>
<br>
- OP is an Outbound EDGE Proxy.<br>
- Registrar is GRUU capable.<br>
<br>
Flow showing the problem:<br>
<br>
- alice registers requesting GRUU.<br>
- OP adds Path.<br>
- alice obtains a GRUU URI.<br>
<br>
case A)<br>
<br>
1) alice calls bob by using her GRUU in Contact. alice also requests<br>
Outbound usage by adding ;ob in Contact of INVITE.<br>
<br>
2) OP adds Record-Route.<br>
<br>
3) OP routes to registrar which adds Record-Route and routes to Bob.<br>
<br>
4) Bob sends INFO to alice.<br>
<br>
5) Registrar receives INFO with GRUU RURI (;gr), removes Route<br>
pointing to it, and still has Route with Outbound flow token<br>
identifier pointing to OP.<br>
<br>
6) Registrar decides to route based on RURI GRUU so performs lookup<br>
and obtains the binding for such a GRUU URI, and of course adds Route<br>
mirroring the registration Path.<br>
<br>
7) INFO arrives to OP with Route belonging to the dialog (with<br>
Outbound flow identifier) and *also* Route produced by the Path.<br>
Typically this means &quot;duplicated =C2=A0route set&quot;. So it could fa=
il routing<br>
the request (loop), and OF COURSE, Route set is broken (RFC 3261)<br>
since Route set has been modified within a dialog.<br>
<br>
<br>
case B)<br>
<br>
1) alice calls bob by using her GRUU in Contact. alice does NOT<br>
request Outbound usage (she does NOT add ;ob in Contact of INVITE)<br>
<br>
2) OP does NOT add R-R.<br>
<br>
3) OP routes to registrar which adds Record-Route and routes to Bob.<br>
<br>
4) Bob sends INFO to alice.<br>
<br>
5) Registrar receives INFO with GRUU RURI (;gr), removes Route pointing to =
it.<br>
<br>
6) Registrar routes based on GRUU so performs lookup and obtains the<br>
binding for such a GRUU URI, and of course adds Route mirroring the<br>
registration Path.<br>
<br>
7) INFO arrives to OP with Route produced by the registration Path. OP<br>
will be able to route the request, of course, but Route set has also<br>
been modified (breaks RFC 3261).<br>
<br>
8) Now alice wants to send BYE to bob. It just adds to the request the<br>
Route mirroring R-R received in the 200 for the initial INVITE, which<br>
just contains the Route pointing to the registrar (since OP did not<br>
R-R).<br>
<br>
9) OP receives an in-dialog request *without* a Route pointing to it.<br>
In several proxy configurations this request would be rejected (why<br>
does the in-dialog request arrive to OP if it did not R-R?). One could<br>
argue &quot;alice should send BYE to Registrar by following the Route set<b=
r>
rules&quot;, but this is not how things work in Outbound, even less if the<=
br>
transport is WebSocket and the UA can just talk with OP).<br>
<br>
<br>
<br>
In both cases Route set is modified during a dialog and breaks RFC<br>
3261. My opinion: This is GOOD since it makes possible for the dialog<br>
to continue alive even if the TCP connection between alice and OP was<br>
closed and re-opened (new Outbound flow token identifier, refreshed in<br>
a new registration so new Path).<br>
<br>
The &quot;nice&quot; solution to avoid problem in case a) dot 7 is:<br>
<br>
If a Registrar receives an in-dialog request with a GRUU RURI pointing<br>
to it, it MUST remove all the Route header present in the request (and<br>
ignore them), and then do lookup, replace RURI with the real binding<br>
and add Route mirroring registration Path values. After lot of<br>
thinking this seems to be the best and only? solution for GRUU to<br>
properly work for in-dialog requests.<br>
<br>
<br>
So I propose a new work RFC 5627-bis to clarify all of this, and also<br>
to &quot;update RFC 3261&quot; stating that dialog Route set can be modifie=
d if<br>
GRUU is used as Contact by any of the peers.<br>
<br>
<br>
Thoughts?<br>
<br>
Best regards.<br>
<br>
<br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
_______________________________________________<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>
</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>Jos=C3=A9 Lu=
is Mill=C3=A1n
</div>

--001a11c2012cb63b7a04e110681c--

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SIPCORE -

On the INSIPID list, James asked me for a callflow showing 100s and 
CANCELs that he could mark up with Session-IDs.

I went looking for a callflow that had forking that showed 100s and 
CANCELs. I was hoping to find a simple one
(either one or two proxies and two UAs). One that showed 183s 
establishing dialogs would be even better.

I was really surprised to find very few examples of forking in ladder 
diagrams at _all_. There are none in 3665 or 3666.
Adam pointed to a diagram in the 199 RFC, but it doesn't show 100s or 
CANCELs.

Can anybody find something already published that does?

If not, I'll make one for INSIPIDs purposes. (But I boggle).

RjS

From worley@shell01.TheWorld.com  Wed Jul 10 13:04:09 2013
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Subject: Re: [sipcore] Already published forking callflow that shows 100s and	CANCELs
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> From: Robert Sparks <rjsparks@nostrum.com>
> 
> On the INSIPID list, James asked me for a callflow showing 100s and 
> CANCELs that he could mark up with Session-IDs.
> 
> I went looking for a callflow that had forking that showed 100s and 
> CANCELs. I was hoping to find a simple one
> (either one or two proxies and two UAs). One that showed 183s 
> establishing dialogs would be even better.

These look like plausible sources to me:

rfc 3087 section 4.2.2
rfc 3665 section 3.2, 3.8
rfc 5359 section 2.9

Dale

From rjsparks@nostrum.com  Wed Jul 10 13:37:28 2013
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Subject: Re: [sipcore] Already published forking callflow that shows 100s and CANCELs
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On 7/10/13 3:02 PM, Dale R. Worley wrote:
>> From: Robert Sparks <rjsparks@nostrum.com>
>>
>> On the INSIPID list, James asked me for a callflow showing 100s and
>> CANCELs that he could mark up with Session-IDs.
>>
>> I went looking for a callflow that had forking that showed 100s and
>> CANCELs. I was hoping to find a simple one
>> (either one or two proxies and two UAs). One that showed 183s
>> establishing dialogs would be even better.
> These look like plausible sources to me:
>
> rfc 3087 section 4.2.2
> rfc 3665 section 3.2, 3.8
> rfc 5359 section 2.9
>
> Dale
Thanks Dale. That last one is a good spot.
(I had looked already at the first two, but they either don't show the 
100s/CANCELS, or don't have forking)

RjS

From mary.ietf.barnes@gmail.com  Wed Jul 10 15:49:56 2013
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Subject: [sipcore] Is there a meeting in Berlin?
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I didn't see it on the main agenda, but I also couldn't find a note to
the mailing indicating such.  Is it correct that SIPCORE is not
meeting in Berlin?

There is one document that has had some discussion on the DISPATCH
mailing list that could most likely be progressed in SIPCORE:
http://www.ietf.org/id/draft-johansson-dane-sip-00.txt


Thanks,
Mary.

From pkyzivat@alum.mit.edu  Wed Jul 10 18:54:11 2013
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Subject: Re: [sipcore] Is there a meeting in Berlin?
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No, no meeting.

On 7/10/13 6:49 PM, Mary Barnes wrote:
> I didn't see it on the main agenda, but I also couldn't find a note to
> the mailing indicating such.  Is it correct that SIPCORE is not
> meeting in Berlin?
>
> There is one document that has had some discussion on the DISPATCH
> mailing list that could most likely be progressed in SIPCORE:
> http://www.ietf.org/id/draft-johansson-dane-sip-00.txt
>
>
> Thanks,
> Mary.
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


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Subject: Re: [sipcore] Already published forking callflow that shows 100s and CANCELs
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> From: Robert Sparks <rjsparks@nostrum.com>
> 
> Thanks Dale. That last one is a good spot.
> (I had looked already at the first two, but they either don't show the 
> 100s/CANCELS, or don't have forking)

The technique I used this for query is to search all the RFCs for ones
that contain these regexps:

	|.*|.*|.*|	(to find railroad diagrams containing forking)
	100 trying	(case insensitive)
	CANCEL

There are only 13 RFCs that pass the test, and searching for the first
regexp with an editor find the railroad diagrams directly.

In *nix, this can be done with one command line:

grep --files-with-match -E '\<SIP\>' ~/RFCs/rfc[3-9]???.txt |
    xargs grep --files-with-match -- '|.*|.*|.*|' | 
    xargs grep --files-with-match -i -- '100 trying' | 
    xargs grep --files-with-match -- 'CANCEL'

Similar searches can answer a remarkable number of questions about the
corpus of RFCs, especially questions about "How is *** normally
formatted in RFCs?"

Dale

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Date: Thu, 11 Jul 2013 15:45:24 -0400 (EDT)
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From: worley@ariadne.com (Dale R. Worley)
Sender: worley@ariadne.com (Dale R. Worley)
To: sipcore@ietf.org
Subject: [sipcore] Copy edits for draft-ietf-sipcore-rfc4244bis-callflows-04
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Thanks to Shida for revising draft-ietf-sipcore-rfc4244bis-callflows!

This is a bunch of copy-edits to
draft-ietf-sipcore-rfc4244bis-callflows-04.  Section pointers are
marked with "*" at the left margin.  Discussion items are marked with
"." at the left margin.  All copy-edits are displayed in "diff"
format.

All of my copy-edits to draft-ietf-sipcore-rfc4244bis-callflows-03
have been resolved, excepting any that may be repeated below

* Section 3.1.  Sequentially Forking (History-Info in Response)

. In message F11, the text "index=1.2.1" appears twice.

    F11 486 Busy Here home -> example.com
 
    SIP/2.0  486 Busy Here
    Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bKx5st
    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK4321
    From: Alice <sip:alice@example.com>;tag=sr3dds
    To: Bob <sip:bob@example.com>;tag=55rdds
    Call-Id: 12345600@example.com
    History-Info: <sip:bob@example.com>;index=1
    History-Info: <sip:bob@192.0.2.4?Reason=SIP%3Bcause%3D302>;\
                  index=1.1;rc=1
    History-Info: <sip:office@example.com?Reason=SIP%3Bcause%3D408>;\
                                                            index=1.2;mp=1
    History-Info: <sip:office@192.0.2.5?Reason=SIP%3Bcause%3D408>;\
-                 index=1.2.1;index=1.2.1;rc=1.2
+                 index=1.2.1;rc=1.2
    History-Info: <sip:home@example.com>;index=1.3;mp=1
    History-Info: <sip:home@192.0.2.6>;index=1.3.1;rc=1.3
    CSeq: 1 INVITE
    Content-Length: 0


* Section 3.6.  PBX Voicemail Example

. In message F7, it appears that the Reason values 480 in index 1.3
and 1.3.1 are both incorrectly changed to 408, because the value 480
is used in the preceding message (which is the first appearance of
1.3 and 1.3.3).

    F6 INVITE Example.com -> VM
 
    INVITE sip:vm@192.0.2.6;target=sip:bob%40example.com;cause=480\
                                             SIP/2.0
    Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK4523
    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
    Max-Forward: 69
    From: Alice <sip:alice@example.com>;tag=kkaz-
    To: Bob <sip:bob@example.com>
    Supported: histinfo
    Call-Id: 12345600@example.com
    CSeq: 1 INVITE
    History-Info: <sip:bob@example.com>;index=1
    History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
                        index=1.1;rc=1
    History-Info: <sip:carol@example.com;cause=480?Reason=SIP%3Bcause%3D\
                                                            408>;index=1.2;mp=1
    History-Info: <sip:carol@192.0.2.4;cause=480?Reason=SIP%3Bcause%3D\
                                                            408>;index=1.2.1;rc=1.2
    History-Info: <sip:vm@example.com;\
                        target=sip:bob%40example.com;cause=480>;\
                        index=1.3;mp=1
    History-Info: <sip:vm@192.0.2.6;\
                        target=sip:bob%40example.com;cause=480>;\
                        index=1.3.1;rc=1.3
    Contact: Alice <sip:alice@192.0.2.3>
    Content-Type: application/sdp
    Content-Length: <appropriate value>
 
 
    F7 200 OK VM -> Example.com
 
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK4523;\
                                     received=192.0.2.101
    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
    From: Alice <sip:alice@example.com>;tag=kkaz-
    To: Bob <sip:bob@example.com>;tag=3dweggs
    Supported: histinfo
    Call-Id: 12345600@example.com
    CSeq: 1 INVITE
    History-Info: <sip:bob@example.com>;index=1
    History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
                       index=1.1;rc=1
    History-Info: <sip:carol@example.com;cause=480?Reason=SIP%3Bcause%3D\
                                                                             408>;index=1.2;mp=1
    History-Info: <sip:carol@192.0.2.4;cause=480?Reason=SIP%3Bcause%3D\
                       408>;index=1.2.1;rc=1.2
    History-Info: <sip:vm@example.com;\
-                      target=sip:bob%40example.com;cause=408>;\
+                      target=sip:bob%40example.com;cause=480>;\
                       index=1.3;mp=1
    History-Info: <sip:vm@192.0.2.6;\
-                      target=sip:bob%40example.com;cause=408>;\
+                      target=sip:bob%40example.com;cause=480>;\
                       index=1.3.1;rc=1.3
    Contact: <sip:vm@192.0.2.6>
    Content-Type: application/sdp
    Content-Length: <appropriate value>



* Section 3.7.  Consumer Voicemail Example

. In message F4, index 1.1, there is a '>' that is doubled.  In F5,
index 1.1, this '>>' is turned into '>">'.  In F6, index 1.1, it is
again '>>'.  In message F7, index 1.1, it is shown correctly as '>'.

. In message F6, index 1.2.2 has the portion "index=1.2.2;mp=1.2" on a
separate line that does not start with whitespace.  As such, it is not
a continuation of the previous header line.  What was probably
intended was that that line was the tail of the preceding line, which
now contains just a semicolon.

. In message F6, index 1.2 has "Reason=...408", but the fork
descendant from 1.2 has not terminated yet.  This Reason parameter
should be on index 1.2.1 instead.  F7 needs the same fix on index 1.2,
although index 1.2.1 is correct.

. In message F6, the URI in index 1.2.2 probably needs a "cause=408"
parameter.  This isn't absolutely certain, as the target of the URI is
a service whose exact interface is unspecified, but it seems like
adding "cause" would be the best practice re RFC 4458.  The same
change would then be needed in message F7.  The "cause=408" parameter
would then probably be carried over into the URI of index 1.2.2.1 (in
both F6 and F7).  In any case, the "cause=408" *header* parameter in
message F6, index 1.2.2.1 is incorrect.

. In message F6, index 1.2.2.1, the "rc" parameter should be 1.2.2
rather than 1.3.

    F4 INVITE Example.com -> Carol
 
    INVITE sip:carol@192.0.2.4 SIP/2.0
    Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK24s5
    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
    Max-Forward: 69
    From: Alice <sip:alice@example.com>;tag=kkaz-
    To: Bob <sip:bob@example.com>
    Supported: histinfo
    Call-Id: 12345600@example.com
    CSeq: 1 INVITE
    History-Info: <sip:bob@example.com>;index=1
    History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
-                                                           %3Btext%3D%22Moved%20Temporarily%22>>\
+                                                           %3Btext%3D%22Moved%20Temporarily%22>\
                  ;index=1.1;rc=1
    History-Info: <sip:carol@example.com>;index=1.2;mp=1
    History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
    Contact: Alice <sip:alice@192.0.2.3>
    Content-Type: application/sdp
    Content-Length: <appropriate value>
 
 
    F5 180 Ringing Carol -> Example.com
 
    SIP/2.0 180 Ringing
    Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK24s5;\
                    received=192.0.2.101
    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
    From: Alice <sip:alice@example.com>;tag=kkaz-
    To: Bob <sip:bob@example.com>;tag=setss3x
    Supported: histinfo
    Call-Id: 12345600@example.com
    CSeq: 1 INVITE
    History-Info: <sip:bob@example.com>;index=1
    History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
-                                                           %3Btext%3D%22Moved%20Temporarily%22>">\
+                                                           %3Btext%3D%22Moved%20Temporarily%22>\
                  ;index=1.1;rc=1
    History-Info: <sip:carol@example.com>;index=1.2;mp=1
    History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
    Contact: <sip:carol@192.0.2.4>
    Content-Type: application/sdp
    Content-Length: <appropriate value>
 
 
    F6 INVITE Example.com -> VM
 
    INVITE sip:vm@192.0.2.6;target=sip:carol%40example.com SIP/2.0
    Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bKbbg4
    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
    Max-Forward: 69
    From: Alice <sip:alice@example.com>;tag=kkaz-
    To: Bob <sip:bob@example.com>
    Supported: histinfo
    Call-Id: 12345600@example.com
    CSeq: 1 INVITE
    History-Info: <sip:bob@example.com>;index=1
    History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
-                                                           %3Btext%3D%22Moved%20Temporarily%22>>\
+                                                           %3Btext%3D%22Moved%20Temporarily%22>\
                  ;index=1.1;rc=1
-   History-Info: <sip:carol@example.com?Reason=SIP%3Bcause%3D408>;\
+   History-Info: <sip:carol@example.com>;\
                       index=1.2;mp=1
-   History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
+   History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;index=1.2.1;rc=1.2
-   History-Info: <sip:vm@example.com;target=sip:carol%40example.com>\
-                                                                           ;
-   index=1.2.2;mp=1.2
+   History-Info: <sip:vm@example.com;target=sip:carol%40example.com;cause=408>\
+                                                                           ;index=1.2.2;mp=1.2
-   History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com>\
-                                                                           ;cause=408;index=1.2.2.1;rc=1.3
+   History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com;cause=408>\
+                                                                           ;index=1.2.2.1;rc=1.2.2
    Contact: Alice <sip:alice@192.0.2.3>
    Content-Type: application/sdp
    Content-Length: <appropriate value>
 

    F7 200 OK VM -> Example.com
 
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bKbbg4
    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
    From: Alice <sip:alice@example.com>;tag=kkaz-
    To: Bob <sip:bob@example.com>;tag=3dweggs
    Supported: histinfo
    Call-Id: 12345600@example.com
    CSeq: 1 INVITE
    History-Info: <sip:bob@example.com>;index=1
    History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
                                                            %3Btext%3D%22Moved%20Temporarily%22>\
                  ;index=1.1;rc=1
-   History-Info: <sip:carol@example.com?Reason=SIP%3Bcause%3D408>;\
+   History-Info: <sip:carol@example.com>;\
                       index=1.2;mp=1
    History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;\
                                                                             index=1.2.1;rc=1.2
-   History-Info: <sip:vm@example.com;target=sip:carol%40example.com>;\
+   History-Info: <sip:vm@example.com;target=sip:carol%40example.com;cause=408>;\
                        index=1.2.2;mp=1.2
-   History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com>;\
+   History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com;cause=408>;\
                        index=1.2.2.1;rc=1.2.2
    Contact: <sip:carol@192.0.2.5>
    Content-Type: application/sdp
    Content-Length: <appropriate value>

Dale

From shida@ntt-at.com  Thu Jul 11 13:32:21 2013
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Subject: Re: [sipcore] Copy edits for draft-ietf-sipcore-rfc4244bis-callflows-04
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Hi Dale;

 Thank you again for a thorough review and I apologize for=20
redundant error on my side.=20

 I will try to reflect these fixes and update it before the submission=20=

deadline, coming Monday.

 Thanks again
  Shida

On Jul 11, 2013, at 12:45 PM, Dale R. Worley wrote:

> Thanks to Shida for revising draft-ietf-sipcore-rfc4244bis-callflows!
>=20
> This is a bunch of copy-edits to
> draft-ietf-sipcore-rfc4244bis-callflows-04.  Section pointers are
> marked with "*" at the left margin.  Discussion items are marked with
> "." at the left margin.  All copy-edits are displayed in "diff"
> format.
>=20
> All of my copy-edits to draft-ietf-sipcore-rfc4244bis-callflows-03
> have been resolved, excepting any that may be repeated below
>=20
> * Section 3.1.  Sequentially Forking (History-Info in Response)
>=20
> . In message F11, the text "index=3D1.2.1" appears twice.
>=20
>    F11 486 Busy Here home -> example.com
>=20
>    SIP/2.0  486 Busy Here
>    Via: SIP/2.0/TCP proxy.example.com:5060;branch=3Dz9hG4bKx5st
>    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=3Dz9hG4bK4321
>    From: Alice <sip:alice@example.com>;tag=3Dsr3dds
>    To: Bob <sip:bob@example.com>;tag=3D55rdds
>    Call-Id: 12345600@example.com
>    History-Info: <sip:bob@example.com>;index=3D1
>    History-Info: <sip:bob@192.0.2.4?Reason=3DSIP%3Bcause%3D302>;\
>                  index=3D1.1;rc=3D1
>    History-Info: <sip:office@example.com?Reason=3DSIP%3Bcause%3D408>;\
>                                                            =
index=3D1.2;mp=3D1
>    History-Info: <sip:office@192.0.2.5?Reason=3DSIP%3Bcause%3D408>;\
> -                 index=3D1.2.1;index=3D1.2.1;rc=3D1.2
> +                 index=3D1.2.1;rc=3D1.2
>    History-Info: <sip:home@example.com>;index=3D1.3;mp=3D1
>    History-Info: <sip:home@192.0.2.6>;index=3D1.3.1;rc=3D1.3
>    CSeq: 1 INVITE
>    Content-Length: 0
>=20
>=20
> * Section 3.6.  PBX Voicemail Example
>=20
> . In message F7, it appears that the Reason values 480 in index 1.3
> and 1.3.1 are both incorrectly changed to 408, because the value 480
> is used in the preceding message (which is the first appearance of
> 1.3 and 1.3.3).
>=20
>    F6 INVITE Example.com -> VM
>=20
>    INVITE sip:vm@192.0.2.6;target=3Dsip:bob%40example.com;cause=3D480\
>                                             SIP/2.0
>    Via: SIP/2.0/TCP proxy.example.com:5060;branch=3Dz9hG4bK4523
>    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=3Dz9hG4bK42t2
>    Max-Forward: 69
>    From: Alice <sip:alice@example.com>;tag=3Dkkaz-
>    To: Bob <sip:bob@example.com>
>    Supported: histinfo
>    Call-Id: 12345600@example.com
>    CSeq: 1 INVITE
>    History-Info: <sip:bob@example.com>;index=3D1
>    History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302>;\
>                        index=3D1.1;rc=3D1
>    History-Info: =
<sip:carol@example.com;cause=3D480?Reason=3DSIP%3Bcause%3D\
>                                                            =
408>;index=3D1.2;mp=3D1
>    History-Info: <sip:carol@192.0.2.4;cause=3D480?Reason=3DSIP%3Bcause%3=
D\
>                                                            =
408>;index=3D1.2.1;rc=3D1.2
>    History-Info: <sip:vm@example.com;\
>                        target=3Dsip:bob%40example.com;cause=3D480>;\
>                        index=3D1.3;mp=3D1
>    History-Info: <sip:vm@192.0.2.6;\
>                        target=3Dsip:bob%40example.com;cause=3D480>;\
>                        index=3D1.3.1;rc=3D1.3
>    Contact: Alice <sip:alice@192.0.2.3>
>    Content-Type: application/sdp
>    Content-Length: <appropriate value>
>=20
>=20
>    F7 200 OK VM -> Example.com
>=20
>    SIP/2.0 200 OK
>    Via: SIP/2.0/TCP proxy.example.com:5060;branch=3Dz9hG4bK4523;\
>                                     received=3D192.0.2.101
>    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=3Dz9hG4bK42t2
>    From: Alice <sip:alice@example.com>;tag=3Dkkaz-
>    To: Bob <sip:bob@example.com>;tag=3D3dweggs
>    Supported: histinfo
>    Call-Id: 12345600@example.com
>    CSeq: 1 INVITE
>    History-Info: <sip:bob@example.com>;index=3D1
>    History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302>;\
>                       index=3D1.1;rc=3D1
>    History-Info: =
<sip:carol@example.com;cause=3D480?Reason=3DSIP%3Bcause%3D\
>                                                                        =
     408>;index=3D1.2;mp=3D1
>    History-Info: <sip:carol@192.0.2.4;cause=3D480?Reason=3DSIP%3Bcause%3=
D\
>                       408>;index=3D1.2.1;rc=3D1.2
>    History-Info: <sip:vm@example.com;\
> -                      target=3Dsip:bob%40example.com;cause=3D408>;\
> +                      target=3Dsip:bob%40example.com;cause=3D480>;\
>                       index=3D1.3;mp=3D1
>    History-Info: <sip:vm@192.0.2.6;\
> -                      target=3Dsip:bob%40example.com;cause=3D408>;\
> +                      target=3Dsip:bob%40example.com;cause=3D480>;\
>                       index=3D1.3.1;rc=3D1.3
>    Contact: <sip:vm@192.0.2.6>
>    Content-Type: application/sdp
>    Content-Length: <appropriate value>
>=20
>=20
>=20
> * Section 3.7.  Consumer Voicemail Example
>=20
> . In message F4, index 1.1, there is a '>' that is doubled.  In F5,
> index 1.1, this '>>' is turned into '>">'.  In F6, index 1.1, it is
> again '>>'.  In message F7, index 1.1, it is shown correctly as '>'.
>=20
> . In message F6, index 1.2.2 has the portion "index=3D1.2.2;mp=3D1.2" =
on a
> separate line that does not start with whitespace.  As such, it is not
> a continuation of the previous header line.  What was probably
> intended was that that line was the tail of the preceding line, which
> now contains just a semicolon.
>=20
> . In message F6, index 1.2 has "Reason=3D...408", but the fork
> descendant from 1.2 has not terminated yet.  This Reason parameter
> should be on index 1.2.1 instead.  F7 needs the same fix on index 1.2,
> although index 1.2.1 is correct.
>=20
> . In message F6, the URI in index 1.2.2 probably needs a "cause=3D408"
> parameter.  This isn't absolutely certain, as the target of the URI is
> a service whose exact interface is unspecified, but it seems like
> adding "cause" would be the best practice re RFC 4458.  The same
> change would then be needed in message F7.  The "cause=3D408" =
parameter
> would then probably be carried over into the URI of index 1.2.2.1 (in
> both F6 and F7).  In any case, the "cause=3D408" *header* parameter in
> message F6, index 1.2.2.1 is incorrect.
>=20
> . In message F6, index 1.2.2.1, the "rc" parameter should be 1.2.2
> rather than 1.3.
>=20
>    F4 INVITE Example.com -> Carol
>=20
>    INVITE sip:carol@192.0.2.4 SIP/2.0
>    Via: SIP/2.0/TCP proxy.example.com:5060;branch=3Dz9hG4bK24s5
>    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=3Dz9hG4bK42t2
>    Max-Forward: 69
>    From: Alice <sip:alice@example.com>;tag=3Dkkaz-
>    To: Bob <sip:bob@example.com>
>    Supported: histinfo
>    Call-Id: 12345600@example.com
>    CSeq: 1 INVITE
>    History-Info: <sip:bob@example.com>;index=3D1
>    History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
> -                                                           =
%3Btext%3D%22Moved%20Temporarily%22>>\
> +                                                           =
%3Btext%3D%22Moved%20Temporarily%22>\
>                  ;index=3D1.1;rc=3D1
>    History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>    History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc=3D1.2
>    Contact: Alice <sip:alice@192.0.2.3>
>    Content-Type: application/sdp
>    Content-Length: <appropriate value>
>=20
>=20
>    F5 180 Ringing Carol -> Example.com
>=20
>    SIP/2.0 180 Ringing
>    Via: SIP/2.0/TCP proxy.example.com:5060;branch=3Dz9hG4bK24s5;\
>                    received=3D192.0.2.101
>    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=3Dz9hG4bK42t2
>    From: Alice <sip:alice@example.com>;tag=3Dkkaz-
>    To: Bob <sip:bob@example.com>;tag=3Dsetss3x
>    Supported: histinfo
>    Call-Id: 12345600@example.com
>    CSeq: 1 INVITE
>    History-Info: <sip:bob@example.com>;index=3D1
>    History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
> -                                                           =
%3Btext%3D%22Moved%20Temporarily%22>">\
> +                                                           =
%3Btext%3D%22Moved%20Temporarily%22>\
>                  ;index=3D1.1;rc=3D1
>    History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>    History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc=3D1.2
>    Contact: <sip:carol@192.0.2.4>
>    Content-Type: application/sdp
>    Content-Length: <appropriate value>
>=20
>=20
>    F6 INVITE Example.com -> VM
>=20
>    INVITE sip:vm@192.0.2.6;target=3Dsip:carol%40example.com SIP/2.0
>    Via: SIP/2.0/TCP proxy.example.com:5060;branch=3Dz9hG4bKbbg4
>    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=3Dz9hG4bK42t2
>    Max-Forward: 69
>    From: Alice <sip:alice@example.com>;tag=3Dkkaz-
>    To: Bob <sip:bob@example.com>
>    Supported: histinfo
>    Call-Id: 12345600@example.com
>    CSeq: 1 INVITE
>    History-Info: <sip:bob@example.com>;index=3D1
>    History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
> -                                                           =
%3Btext%3D%22Moved%20Temporarily%22>>\
> +                                                           =
%3Btext%3D%22Moved%20Temporarily%22>\
>                  ;index=3D1.1;rc=3D1
> -   History-Info: <sip:carol@example.com?Reason=3DSIP%3Bcause%3D408>;\
> +   History-Info: <sip:carol@example.com>;\
>                       index=3D1.2;mp=3D1
> -   History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc=3D1.2
> +   History-Info: =
<sip:carol@192.0.2.4?Reason=3DSIP%3Bcause%3D408>;index=3D1.2.1;rc=3D1.2
> -   History-Info: <sip:vm@example.com;target=3Dsip:carol%40example.com>\=

> -                                                                      =
     ;
> -   index=3D1.2.2;mp=3D1.2
> +   History-Info: =
<sip:vm@example.com;target=3Dsip:carol%40example.com;cause=3D408>\
> +                                                                      =
     ;index=3D1.2.2;mp=3D1.2
> -   History-Info: <sip:vm@192.0.2.5;target=3Dsip:carol%40example.com>\
> -                                                                      =
     ;cause=3D408;index=3D1.2.2.1;rc=3D1.3
> +   History-Info: =
<sip:vm@192.0.2.5;target=3Dsip:carol%40example.com;cause=3D408>\
> +                                                                      =
     ;index=3D1.2.2.1;rc=3D1.2.2
>    Contact: Alice <sip:alice@192.0.2.3>
>    Content-Type: application/sdp
>    Content-Length: <appropriate value>
>=20
>=20
>    F7 200 OK VM -> Example.com
>=20
>    SIP/2.0 200 OK
>    Via: SIP/2.0/TCP proxy.example.com:5060;branch=3Dz9hG4bKbbg4
>    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=3Dz9hG4bK42t2
>    From: Alice <sip:alice@example.com>;tag=3Dkkaz-
>    To: Bob <sip:bob@example.com>;tag=3D3dweggs
>    Supported: histinfo
>    Call-Id: 12345600@example.com
>    CSeq: 1 INVITE
>    History-Info: <sip:bob@example.com>;index=3D1
>    History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
>                                                            =
%3Btext%3D%22Moved%20Temporarily%22>\
>                  ;index=3D1.1;rc=3D1
> -   History-Info: <sip:carol@example.com?Reason=3DSIP%3Bcause%3D408>;\
> +   History-Info: <sip:carol@example.com>;\
>                       index=3D1.2;mp=3D1
>    History-Info: <sip:carol@192.0.2.4?Reason=3DSIP%3Bcause%3D408>;\
>                                                                        =
     index=3D1.2.1;rc=3D1.2
> -   History-Info: =
<sip:vm@example.com;target=3Dsip:carol%40example.com>;\
> +   History-Info: =
<sip:vm@example.com;target=3Dsip:carol%40example.com;cause=3D408>;\
>                        index=3D1.2.2;mp=3D1.2
> -   History-Info: <sip:vm@192.0.2.5;target=3Dsip:carol%40example.com>;\
> +   History-Info: =
<sip:vm@192.0.2.5;target=3Dsip:carol%40example.com;cause=3D408>;\
>                        index=3D1.2.2.1;rc=3D1.2.2
>    Contact: <sip:carol@192.0.2.5>
>    Content-Type: application/sdp
>    Content-Length: <appropriate value>
>=20
> Dale
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From internet-drafts@ietf.org  Thu Jul 11 22:49:57 2013
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Date: Thu, 11 Jul 2013 22:49:56 -0700
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-rfc4244bis-callflows-05.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Session Initiation Protocol Core Working =
Group of the IETF.

	Title           : Session Initiation Protocol (SIP) History-Info Header Ca=
ll Flow Examples
	Author(s)       : Mary Barnes
                          Francois Audet
                          Shida Schubert
                          Hans Erik van Elburg
                          Christer Holmberg
	Filename        : draft-ietf-sipcore-rfc4244bis-callflows-05.txt
	Pages           : 46
	Date            : 2013-07-11

Abstract:
   This document describes use cases and documents call flows which
   require the History-Info header field to capture the Request-URIs as
   a Session Initiation Protocol (SIP) Request is retargeted.  The use
   cases are described along with the corresponding call flow diagrams
   and messaging details.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc4244bis-callflows

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-rfc4244bis-callflows-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-rfc4244bis-callflows-=
05


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From internet-drafts@ietf.org  Mon Jul 15 04:36:37 2013
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-rfc4244bis-callflows-06.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Session Initiation Protocol Core Working =
Group of the IETF.

	Title           : Session Initiation Protocol (SIP) History-Info Header Ca=
ll Flow Examples
	Author(s)       : Mary Barnes
                          Francois Audet
                          Shida Schubert
                          Hans Erik van Elburg
                          Christer Holmberg
	Filename        : draft-ietf-sipcore-rfc4244bis-callflows-06.txt
	Pages           : 46
	Date            : 2013-07-14

Abstract:
   This document describes use cases and documents call flows which
   require the History-Info header field to capture the Request-URIs as
   a Session Initiation Protocol (SIP) Request is retargeted.  The use
   cases are described along with the corresponding call flow diagrams
   and messaging details.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc4244bis-callflows

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-rfc4244bis-callflows-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-rfc4244bis-callflows-=
06


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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Cc: "'SIPCORE \(Session Initiation Protocol Core\) WG'" <sipcore@ietf.org>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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Hi Inaki,

I have mentioned earlier =
(http://www.ietf.org/mail-archive/web/sipcore/current/msg05614.html) =
that the issue raised by me is related to SIP routing and you are mixing =
it with NAT & Firewall.

In case you wish to cover the issue in the holistic way, Please suggest =
the complete text.

I have disagreement for the statement that SIP over SCTP does not =
document those issue, so SIP over WebSocket will not document the same.=20

Thanks
Partha

PS: I have work experience in SCTP transport with RR added by proxy to =
overcome this issue.

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of I=C3=B1aki Baz Castillo
> Sent: Friday, July 05, 2013 3:07 PM
> To: Paul Kyzivat
> Cc: SIPCORE (Session Initiation Protocol Core) WG
> Subject: Re: [sipcore] Open issues in =
draft-ietf-sipcore-sip-websocket-
> 09
>=20
> Please, let me insist even a bit more on it:
>=20
>=20
>=20
> 2013/7/5 I=C3=B1aki Baz Castillo <ibc@aliax.net>:
> > Any TCP SIP UA is capable of listening for incoming connections but
> > that does not mean that the proxy should not R-R. If the UA is =
behind
> > NAT then RFC 5626 must be used (I avoid here proprietary similar
> > mechanisms), and RFC 5626 *alreay* mandates R-R for such a mechanism
> > to work. If GRUU is also in use then the proxy can decide not to R-R
> > (not needed since incoming requests will arrive via the registration
> > path which will of course include Outbound stuff).
>=20
>=20
>=20
> a) If the SIP-WS UA runs in a browser then the website developer can
> make usage of RFC 5626 so the UA *requests* Outbound usage to its
> proxy.
>=20
> b) If the SIP-WS UA runs in a desktop and is both a SIP WS Client and
> SIP WS Server (and let's assume it has public IP) then it can avoid
> requesting Outbound to the proxy. If the proxy does R-R it is because
> the proxy wants to be in the patch of the dialog (local policy) and
> not because it is required for the dialog to work.
>=20
>=20
> Of course, in case b), if the peer does not speak WS and the proxy
> does not R-R here we have a problem, but this is *exactly* the same
> problem as if the caller was a UA using SCTP in its Contact URI. So
> again, this is about local policy in the proxy whether to decide it
> does R-R or not.
>=20
> I don't find "Record-Route" word in RFC 4168 so I expect there should
> be no "MUST" keyword about Record-Route in
> draft-ietc-sipcore-sip-websocket. Let me know if I'm wrong.
>=20
>=20
>=20
> Summarizing:
>=20
> - This is not an issue introduced by this new WS transport.
> - The issue is not new.
> - There are already standard mechanism to make it to work (the
> *client* can request them to the proxy !).
> - I see no reason at all to mandate nothing in the sip-ws draft about
> this.
> - Guidelines section in the draft already covers it in an informative
> way.
>=20
>=20
>=20
> Hope I am right.
>=20
> Best regards.
>=20
>=20
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From brett@broadsoft.com  Wed Jul 31 07:25:03 2013
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From: Brett Tate <brett@broadsoft.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] SIP and Internationalisation
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Subject: Re: [sipcore] SIP and Internationalisation
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> There are probably some similar issues for password=20
> normalization, although I'm not sure that's as much=20
> of an issue, since passwords, by their nature, are=20
> generated in a more repeatable way than usernames are.=20
> I would expect that people who have spent far more time=20
> than I have thinking about the problem have some ideas here.

Although some of their work is experimental, the httpauth working group is =
updating RFC 2617 for internationalization and security.

http://tools.ietf.org/wg/httpauth/



From stpeter@stpeter.im  Wed Jul 31 08:03:27 2013
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP and Internationalisation
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On 7/31/13 4:24 PM, Brett Tate wrote:
>> There are probably some similar issues for password 
>> normalization, although I'm not sure that's as much 
>> of an issue, since passwords, by their nature, are 
>> generated in a more repeatable way than usernames are. 
>> I would expect that people who have spent far more time 
>> than I have thinking about the problem have some ideas here.
> 
> Although some of their work is experimental, the httpauth working group is updating RFC 2617 for internationalization and security.
> 
> http://tools.ietf.org/wg/httpauth/

See the username profile we've been defining in the PRECIS WG (which I
think the httpauth folks will end up using):

http://datatracker.ietf.org/doc/draft-ietf-precis-saslprepbis/

Peter

-- 
Peter Saint-Andre
https://stpeter.im/


