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Subject: [sipcore] RFC 7134 on The Management Policy of the Resource Priority Header (RPH) Registry Changed to "IETF Review"
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 7134

        Title:      The Management Policy of the 
                    Resource Priority Header (RPH) Registry Changed 
                    to "IETF Review" 
        Author:     B. Rosen
        Status:     Standards Track
        Stream:     IETF
        Date:       March 2014
        Mailbox:    br@brianrosen.net
        Pages:      2
        Characters: 3565
        Updates:    RFC4412

        I-D Tag:    draft-rosen-rph-reg-policy-01.txt

        URL:        http://www.rfc-editor.org/rfc/rfc7134.txt

RFC 4412 defines the "Resource-Priority Namespaces" and
"Resource-Priority Priority-values" registries.  The management policy of
these registries is "Standards Action".  This document normatively updates
RFC 4412 to change the management policy of these registries to "IETF Review".

This document is a product of the Session Initiation Protocol Core Working Group of the IETF.

This is now a Proposed Standard.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
status of this protocol.  Distribution of this memo is unlimited.

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The RFC Editor Team
Association Management Solutions, LLC



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Subject: [sipcore] RFC 7131 on Session Initiation Protocol (SIP) History-Info Header Call Flow Examples
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 7131

        Title:      Session Initiation Protocol (SIP) History-Info 
                    Header Call Flow Examples 
        Author:     M. Barnes, F. Audet,
                    S. Schubert, H. van Elburg,
                    C. Holmberg
        Status:     Informational
        Stream:     IETF
        Date:       March 2014
        Mailbox:    mary.ietf.barnes@gmail.com, 
                    francois.audet@skype.net, 
                    shida@ntt-at.com,
                    ietf.hanserik@gmail.com, 
                    christer.holmberg@ericsson.com
        Pages:      52
        Characters: 93039
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-sipcore-rfc4244bis-callflows-08.txt

        URL:        http://www.rfc-editor.org/rfc/rfc7131.txt

This document describes use cases and documents call flows that
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.

This document is a product of the Session Initiation Protocol Core Working Group of the IETF.


INFORMATIONAL: This memo provides information for the Internet community.
It does not specify an Internet standard of any kind. Distribution of
this memo is unlimited.

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The RFC Editor Team
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From nobody Thu Mar  6 06:18:52 2014
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From: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [sipcore] IPv6 in the sip core wg
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References: <C774C9EA-4E79-4846-A834-BF86D2DD8018@edvina.net> <52A7486E.2090005@alum.mit.edu> <FFB57ECD-8CDB-44E9-9A3F-5418AAC01C5B@iii.ca> <26C3B24F-FCBE-4D10-ADD5-E28B6E95A8FB@edvina.net> <BCD747C2-B0E9-492E-97E2-58B078AF5F74@iii.ca> <52A8CFC3.3080309@alum.mit.edu> <CAHBDyN6qK6_Cone+wkrcV_LZCca3b_dbf6rkzwnATZg4R6kn5Q@mail.gmail.com> <52A9F990.1030201@alum.mit.edu> <40B29D11-A4EE-4F7B-97C9-612313CFFB7E@cisco.com> <949EF20990823C4C85C18D59AA11AD8B0F858B@FR712WXCHMBA11.zeu.alcatel-lucent.com> <A054AE81-F690-42DE-8B77-1F7E4F0EA7B1@cisco.com> <949EF20990823C4C85C18D59AA11AD8B0F87DC@FR712WXCHMBA11.zeu.alcatel-lucent.com> <A86516A3-8DAA-43B0-8345-7B26182B99E3@cisco.com> <52AB35A6.8030701@alum.mit.edu> <597D373F-6ACF-4D81-83F0-8CB5A08202E1@cisco.com> <52AF5C0C.9000206@alum.mit.edu> <949EF20990823C4C85C18D59AA11AD8B0FA8CF@FR712WXCHMBA11.zeu.alcatel-lucent.com> <52B1D794.1080602@alum.mit.edu> <52DEF1ED.8040905@nostrum.com> <2AF35782-4049-4AD6-B1E6-A97171DB94EE@cisco.com> <52DEF76B.9050706@nostrum.com> <BC2BF195-1140-48C4-AF4E-C4CC6951E2FB@cisco.com>
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Cc: Olle E Johansson <oej@edvina.net>, SIPCORE WG <sipcore@ietf.org>
Subject: Re: [sipcore] IPv6 in the sip core wg
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--_000_53C929533CC04738A41F61EEF3853514ciscocom_
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Content-Transfer-Encoding: quoted-printable

As per our discussion during the meeting...

I propose we narrow the scope of the proposed milestone to better suit the =
existing work.  The following two word edit is proposed:

OLD:

June 2014  Request publication of procedures for dual-stack client and serv=
er handling of SIP URIs


June 2014  Request publication of DNS look-up procedures for dual-stack cli=
ent and server handling of SIP URIs


Are folks ok with this new text?  After further study of SIP dual-stack gap=
s, as Dan York proposed we do, we can add appropriately scoped milestones a=
s they arise.

Cheers,

Gonzalo


On Jan 21, 2014, at 10:42 PM, Cullen Jennings (fluffy) <fluffy@cisco.com<ma=
ilto:fluffy@cisco.com>> wrote:


works for me

On Jan 21, 2014, at 3:40 PM, Adam Roach <adam@nostrum.com<mailto:adam@nostr=
um.com>> wrote:

On 1/21/14 16:24, Cullen Jennings (fluffy) wrote:
No, that implies that 3263 and 6157 are wrong and products that implement n=
eed to change - If you want that milestone, first you need to be able be cl=
ear about what is wrong with 3263 or 6157.  How about

June 2014  Request publication of procedures for dual-stack client and serv=
er handling of SIP URIs

That runs the risk of being confused with what 6157 already does. How about=
:

 June 2014 Request publication of Happy Eyeballs procedures for handling of=
 SIP URIs

Which seems a fair summary of what Olle and Gonzalo's current document inte=
nds to do.

/a



--_000_53C929533CC04738A41F61EEF3853514ciscocom_
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Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space;">
As per our discussion during the meeting...
<div><br>
</div>
<div>I propose we narrow the scope of the proposed milestone to better suit=
 the existing work. &nbsp;The following two word edit is proposed:</div>
<div><br>
</div>
<div>OLD:</div>
<div><br>
</div>
<div>
<div>June 2014 &nbsp;Request publication of procedures for dual-stack clien=
t and server handling of SIP URIs</div>
<br>
</div>
<div><br>
</div>
<div>
<div>June 2014 &nbsp;Request publication of <b>DNS look-up</b> procedures f=
or dual-stack client and server&nbsp;handling of SIP URIs</div>
<div><br>
</div>
<div><br>
</div>
<div>Are folks ok with this new text? &nbsp;After further study of SIP dual=
-stack gaps, as Dan York proposed we do, we can add appropriately scoped mi=
lestones as they arise.</div>
<div><br>
</div>
<div>Cheers,</div>
<div><br>
</div>
<div>Gonzalo</div>
<div><br>
</div>
<br>
On Jan 21, 2014, at 10:42 PM, Cullen Jennings (fluffy) &lt;<a href=3D"mailt=
o:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wrote:<br>
<br>
<blockquote type=3D"cite"><br>
works for me<br>
<br>
On Jan 21, 2014, at 3:40 PM, Adam Roach &lt;<a href=3D"mailto:adam@nostrum.=
com">adam@nostrum.com</a>&gt; wrote:<br>
<br>
<blockquote type=3D"cite">On 1/21/14 16:24, Cullen Jennings (fluffy) wrote:=
<br>
<blockquote type=3D"cite">No, that implies that 3263 and 6157 are wrong and=
 products that implement need to change - If you want that milestone, first=
 you need to be able&nbsp;be clear about what is wrong with 3263 or 6157. &=
nbsp;How about<br>
<br>
<blockquote type=3D"cite">June 2014 &nbsp;Request publication of procedures=
 for dual-stack client and server handling of SIP URIs<br>
</blockquote>
</blockquote>
<br>
That runs the risk of being confused with what 6157 already does. How about=
:<br>
<br>
&nbsp;June 2014 Request publication of Happy Eyeballs procedures for handli=
ng of SIP URIs<br>
<br>
Which seems a fair summary of what Olle and Gonzalo's current document inte=
nds to do.<br>
<br>
/a<br>
</blockquote>
<br>
</blockquote>
<br>
</div>
</body>
</html>

--_000_53C929533CC04738A41F61EEF3853514ciscocom_--


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
Thread-Topic: [sipcore] IPv6 in the sip core wg
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Subject: Re: [sipcore] IPv6 in the sip core wg
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Works for me

On Mar 6, 2014, at 2:18 PM, Gonzalo Salgueiro (gsalguei) <gsalguei@cisco.co=
m> wrote:

> As per our discussion during the meeting...
>=20
> I propose we narrow the scope of the proposed milestone to better suit th=
e existing work.  The following two word edit is proposed:
>=20
> OLD:
>=20
> June 2014  Request publication of procedures for dual-stack client and se=
rver handling of SIP URIs
>=20
>=20
> June 2014  Request publication of DNS look-up procedures for dual-stack c=
lient and server handling of SIP URIs
>=20
>=20
> Are folks ok with this new text?  After further study of SIP dual-stack g=
aps, as Dan York proposed we do, we can add appropriately scoped milestones=
 as they arise.
>=20
> Cheers,
>=20
> Gonzalo
>=20
>=20
> On Jan 21, 2014, at 10:42 PM, Cullen Jennings (fluffy) <fluffy@cisco.com>=
 wrote:
>=20
>>=20
>> works for me
>>=20
>> On Jan 21, 2014, at 3:40 PM, Adam Roach <adam@nostrum.com> wrote:
>>=20
>>> On 1/21/14 16:24, Cullen Jennings (fluffy) wrote:
>>>> No, that implies that 3263 and 6157 are wrong and products that implem=
ent need to change - If you want that milestone, first you need to be able =
be clear about what is wrong with 3263 or 6157.  How about
>>>>=20
>>>>> June 2014  Request publication of procedures for dual-stack client an=
d server handling of SIP URIs
>>>=20
>>> That runs the risk of being confused with what 6157 already does. How a=
bout:
>>>=20
>>>  June 2014 Request publication of Happy Eyeballs procedures for handlin=
g of SIP URIs
>>>=20
>>> Which seems a fair summary of what Olle and Gonzalo's current document =
intends to do.
>>>=20
>>> /a
>>=20
>=20


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To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Cc: SIPCORE WG <sipcore@ietf.org>, Olle E Johansson <oej@edvina.net>, "Gonzalo Salgueiro \(gsalguei\)" <gsalguei@cisco.com>
Subject: Re: [sipcore] IPv6 in the sip core wg
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Ok for me.

/O
On 06 Mar 2014, at 14:25, Cullen Jennings (fluffy) <fluffy@cisco.com> =
wrote:

>=20
> Works for me
>=20
> On Mar 6, 2014, at 2:18 PM, Gonzalo Salgueiro (gsalguei) =
<gsalguei@cisco.com> wrote:
>=20
>> As per our discussion during the meeting...
>>=20
>> I propose we narrow the scope of the proposed milestone to better =
suit the existing work.  The following two word edit is proposed:
>>=20
>> OLD:
>>=20
>> June 2014  Request publication of procedures for dual-stack client =
and server handling of SIP URIs
>>=20
>>=20
>> June 2014  Request publication of DNS look-up procedures for =
dual-stack client and server handling of SIP URIs
>>=20
>>=20
>> Are folks ok with this new text?  After further study of SIP =
dual-stack gaps, as Dan York proposed we do, we can add appropriately =
scoped milestones as they arise.
>>=20
>> Cheers,
>>=20
>> Gonzalo
>>=20
>>=20
>> On Jan 21, 2014, at 10:42 PM, Cullen Jennings (fluffy) =
<fluffy@cisco.com> wrote:
>>=20
>>>=20
>>> works for me
>>>=20
>>> On Jan 21, 2014, at 3:40 PM, Adam Roach <adam@nostrum.com> wrote:
>>>=20
>>>> On 1/21/14 16:24, Cullen Jennings (fluffy) wrote:
>>>>> No, that implies that 3263 and 6157 are wrong and products that =
implement need to change - If you want that milestone, first you need to =
be able be clear about what is wrong with 3263 or 6157.  How about
>>>>>=20
>>>>>> June 2014  Request publication of procedures for dual-stack =
client and server handling of SIP URIs
>>>>=20
>>>> That runs the risk of being confused with what 6157 already does. =
How about:
>>>>=20
>>>> June 2014 Request publication of Happy Eyeballs procedures for =
handling of SIP URIs
>>>>=20
>>>> Which seems a fair summary of what Olle and Gonzalo's current =
document intends to do.
>>>>=20
>>>> /a
>>>=20
>>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Tue Mar 11 09:19:01 2014
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From: <marianne.mohali@orange.com>
To: "Mary Barnes (mary.ietf.barnes@gmail.com)" <mary.ietf.barnes@gmail.com>, "Shida Schubert <shida@agnada.com> (shida@agnada.com)" <shida@agnada.com>, "Christer Holmberg (christer.holmberg@ericsson.com)" <christer.holmberg@ericsson.com>
Thread-Topic: [sipcore] Inconcistency between RFC 7044 and RFC7131
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Date: Tue, 11 Mar 2014 16:18:49 +0000
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Subject: [sipcore]  Inconcistency between RFC 7044 and RFC7131
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Hello,

Checking the 2 published RFCs I identified an inconsistency regarding the P=
BX Voicemail use case recommendation between RFC 7044 and RFC 7131:

RFC 7044 states:

12.1.  PBX Voicemail

The original target is determined by finding the first hi-entry

   tagged with "rc" and using the hi-entry referenced by the index of

   the "rc" header field parameter as the target for determining the

   appropriate mailbox.

RFC 7131 states:
=A73.6 PBX Voicemail Example:
The original target is determined by finding the first
   hi-entry tagged with "rc" or "mp" and using the hi-entry referenced
   by the index of "rc" or "mp" header field parameter as the target for
   determining the appropriate mailbox.

Hum hum... I think RFC 7131 is correct. Is that right?

Best Regards,
Marianne


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<p class=3D"MsoNormal">Hello,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Checking the 2 published RFCs I identified an incons=
istency regarding the PBX Voicemail use case recommendation between RFC 704=
4 and RFC 7131:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">RFC 7044 states:<o:p></o:p></p>
<pre><span class=3D"mh"><b>12.1.&nbsp; PBX Voicemail</b></span><b><o:p></o:=
p></b></pre>
<pre>The original target is determined by finding the first hi-entry<o:p></=
o:p></pre>
<pre>&nbsp;&nbsp; tagged with &quot;rc&quot; and using the hi-entry referen=
ced by the index of<o:p></o:p></pre>
<pre>&nbsp;&nbsp; the &quot;rc&quot; header field parameter as the target f=
or determining the<o:p></o:p></pre>
<pre>&nbsp;&nbsp; appropriate mailbox.<o:p></o:p></pre>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">RFC 7131 states:<o:p></o:p></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Courier New&quot;">=A73.6 PBX Voicemail Example:
<o:p></o:p></span></b></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">The original target is determined by finding the first<o:p=
></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; hi-entry tagged with &quot;rc&quot; or &quot;=
mp&quot; and using the hi-entry referenced<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; by the index of &quot;rc&quot; or &quot;mp&qu=
ot; header field parameter as the target for<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; determining the appropriate mailbox.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hum hum&#8230; I think RFC 7131 is correct. Is that =
right?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Best Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Marianne<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<PRE>______________________________________________________________________=
___________________________________________________

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pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu=
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From: Andrew Allen <aallen@blackberry.com>
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I have submitted a draft that discusses the problem posed by B2BUA intermed=
iaries for out of dialog requests that are related to another dialog and id=
entifies some possible solutions.

It can be found at:

http://www.ietf.org/id/draft-allen-dispatch-routing-out-of-dialog-request-0=
0.txt


I would like to receive comments on this and a view as to where such work s=
hould be dispatched to.

Andrew

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<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I have submitted a draft that discusses the problem =
posed by B2BUA intermediaries for out of dialog requests that are related t=
o another dialog and identifies some possible solutions.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">It can be found at:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><a href=3D"http://www.ietf.org/id/draft-allen-dispat=
ch-routing-out-of-dialog-request-00.txt">http://www.ietf.org/id/draft-allen=
-dispatch-routing-out-of-dialog-request-00.txt</a><o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I would like to receive comments on this and a view =
as to where such work should be dispatched to.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Andrew<o:p></o:p></p>
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From nobody Wed Mar 12 11:38:21 2014
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Date: Wed, 12 Mar 2014 13:38:09 -0500
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, "SIPCORE \(sipcore@ietf.org\)" <sipcore@ietf.org>, "straw@ietf.org" <straw@ietf.org>
Subject: Re: [sipcore] draft-allen-dispatch-routing-out-of-dialog-request-00 posted
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Please do NOT crosspost.  If you want this document to be considered by the
DISPATCH WG process, then please keep the discussion there.  Right now,
this email is cached with my SIPCORE WG emails since the IETF mailing lists
do NOT send duplicate messages.  So it's very likely that I will miss any
discussion that might be relevant to making a decision in the DISPATCH WG.
 The right thing to do is to have the discussion on the DISPATCH WG mailing
list and send a note to any related WGs that the discussion is underway on
the DISPATCH WG mailing list.

Thanks,
Mary.


On Wed, Mar 12, 2014 at 1:20 PM, Andrew Allen <aallen@blackberry.com> wrote:

>
>
> I have submitted a draft that discusses the problem posed by B2BUA
> intermediaries for out of dialog requests that are related to another
> dialog and identifies some possible solutions.
>
>
>
> It can be found at:
>
>
>
>
> http://www.ietf.org/id/draft-allen-dispatch-routing-out-of-dialog-request-00.txt
>
>
>
>
>
> I would like to receive comments on this and a view as to where such work
> should be dispatched to.
>
>
>
> Andrew
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>
>

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<div dir=3D"ltr">Please do NOT crosspost. =A0If you want this document to b=
e considered by the DISPATCH WG process, then please keep the discussion th=
ere. =A0Right now, this email is cached with my SIPCORE WG emails since the=
 IETF mailing lists do NOT send duplicate messages. =A0So it&#39;s very lik=
ely that I will miss any discussion that might be relevant to making a deci=
sion in the DISPATCH WG. =A0The right thing to do is to have the discussion=
 on the DISPATCH WG mailing list and send a note to any related WGs that th=
e discussion is underway on the DISPATCH WG mailing list.<div>
<br></div><div>Thanks,</div><div>Mary.=A0</div></div><div class=3D"gmail_ex=
tra"><br><br><div class=3D"gmail_quote">On Wed, Mar 12, 2014 at 1:20 PM, An=
drew Allen <span dir=3D"ltr">&lt;<a href=3D"mailto:aallen@blackberry.com" t=
arget=3D"_blank">aallen@blackberry.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">I have submitted a draft that discusses the problem =
posed by B2BUA intermediaries for out of dialog requests that are related t=
o another dialog and identifies some possible solutions.<u></u><u></u></p>

<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">It can be found at:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><a href=3D"http://www.ietf.org/id/draft-allen-dispat=
ch-routing-out-of-dialog-request-00.txt" target=3D"_blank">http://www.ietf.=
org/id/draft-allen-dispatch-routing-out-of-dialog-request-00.txt</a><u></u>=
<u></u></p>

<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">I would like to receive comments on this and a view =
as to where such work should be dispatched to.<span class=3D"HOEnZb"><font =
color=3D"#888888"><u></u><u></u></font></span></p><span class=3D"HOEnZb"><f=
ont color=3D"#888888">
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">Andrew<u></u><u></u></p>
</font></span></div>
</div>

<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>
<br></blockquote></div><br></div>

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Subject: [sipcore] Passing of Francois Audet
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A number of you all will remember Francois Audet, who was a colleague of
mine at Nortel and a regular IETF attendee (until he went to Skype ;)  He
was a co-author on the recently published 4244bis drafts.  He unfortunately
lost his battle with cancer on Friday.  I have received the following
information from a family friend via Facebook:

Thanks for all your support messages, both on & offline.
I've talked to Genevi=E8ve over the week-end, and she reminded me that ther=
e
is just no real treatment of Brain Tumors, and that form of cancer is
growing at some alarming rate.
She agreed that the best support message would be to sponsor the research,
for which donations can be made as a memorial gift for Francois Audet.
I've selected a few organizations:
- Stanford Brain Tumor Center (http://medicalgiving.stanford.edu/, this is
where Fran=E7ois was treated. Make sure to indicate the Brain Tumor Center =
as
the recipient).
- UCSF Brain Tumor Center (
https://makeagift.ucsf.edu/site/SPageServer?pagename=3DA1_API_GordonMurrayG=
ivingForm&ACode=3DB2934
)
- National Brain Tumor Society (http://www.braintumor.org/)

Note that he also left behind a son, Olivier.

That's all the information I have at this point.

Regards,
Mary.

--bcaec53f35f501d2c604f4cf76ff
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">A number of you all will remember Francois Audet, who was =
a colleague of mine at Nortel and a regular IETF attendee (until he went to=
 Skype ;) =A0He was a co-author on the recently published 4244bis drafts. =
=A0He unfortunately lost his battle with cancer on Friday. =A0I have receiv=
ed the following information from a family friend via Facebook:<div>
<br></div><div><div>Thanks for all your support messages, both on &amp; off=
line.</div><div>I&#39;ve talked to Genevi=E8ve over the week-end, and she r=
eminded me that there is just no real treatment of Brain Tumors, and that f=
orm of cancer is growing at some alarming rate.</div>
<div>She agreed that the best support message would be to sponsor the resea=
rch, for which donations can be made as a memorial gift for Francois Audet.=
</div><div>I&#39;ve selected a few organizations:</div><div>- Stanford Brai=
n Tumor Center (<a href=3D"http://medicalgiving.stanford.edu/">http://medic=
algiving.stanford.edu/</a>, this is where Fran=E7ois was treated. Make sure=
 to indicate the Brain Tumor Center as the recipient).</div>
<div>- UCSF Brain Tumor Center (<a href=3D"https://makeagift.ucsf.edu/site/=
SPageServer?pagename=3DA1_API_GordonMurrayGivingForm&amp;ACode=3DB2934">htt=
ps://makeagift.ucsf.edu/site/SPageServer?pagename=3DA1_API_GordonMurrayGivi=
ngForm&amp;ACode=3DB2934</a>)</div>
<div>- National Brain Tumor Society (<a href=3D"http://www.braintumor.org/"=
>http://www.braintumor.org/</a>)</div></div><div><br></div><div>Note that h=
e also left behind a son, Olivier.=A0</div><div><br></div><div>That&#39;s a=
ll the information I have at this point.=A0</div>
<div><br></div><div>Regards,</div><div>Mary.</div></div>

--bcaec53f35f501d2c604f4cf76ff--


From nobody Mon Mar 17 10:40:13 2014
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From: "Roy, Radhika R CIV USARMY (US)" <radhika.r.roy.civ@mail.mil>
To: Mary Barnes <mary.ietf.barnes@gmail.com>, "rai@ietf.org" <rai@ietf.org>
Thread-Topic: [sipcore] Passing of Francois Audet (UNCLASSIFIED)
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Subject: Re: [sipcore] Passing of Francois Audet (UNCLASSIFIED)
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Classification: UNCLASSIFIED
Caveats: NONE

Hi, Marry and all:

My heartfelt prayer to God for resting his soul in Heaven! My heartfelt
sympathy to his son Oliver and family as we all is a part of this =
eternal
cycle!

I would know him since the mid-1990s while he used to participate in ATM
Forum and a liaison to the ITU-T H.323 WG. His crystal clear voice and
technical explanations has kept amazing me and all others since his last
breath!

Best regards,
Radhika


-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Mary Barnes
Sent: Monday, March 17, 2014 11:56 AM
To: rai@ietf.org
Cc: DISPATCH; SIPCORE
Subject: [sipcore] Passing of Francois Audet

A number of you all will remember Francois Audet, who was a colleague of
mine at Nortel and a regular IETF attendee (until he went to Skype ;)  =
He
was a co-author on the recently published 4244bis drafts.  He =
unfortunately
lost his battle with cancer on Friday.  I have received the following
information from a family friend via Facebook:

Thanks for all your support messages, both on & offline.
I've talked to Genevi=E8ve over the week-end, and she reminded me that =
there
is just no real treatment of Brain Tumors, and that form of cancer is
growing at some alarming rate.
She agreed that the best support message would be to sponsor the =
research,
for which donations can be made as a memorial gift for Francois Audet.
I've selected a few organizations:
- Stanford Brain Tumor Center (http://medicalgiving.stanford.edu/, this =
is
where Fran=E7ois was treated. Make sure to indicate the Brain Tumor =
Center as
the recipient).
- UCSF Brain Tumor Center
(https://makeagift.ucsf.edu/site/SPageServer?pagename=3DA1_API_GordonMurr=
ayGiv
ingForm&ACode=3DB2934)
- National Brain Tumor Society (http://www.braintumor.org/)

Note that he also left behind a son, Olivier.=20

That's all the information I have at this point.=20

Regards,
Mary.



Classification: UNCLASSIFIED
Caveats: NONE


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From nobody Tue Mar 18 12:22:20 2014
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Date: Tue, 18 Mar 2014 14:54:51 -0400
From: Tom Taylor <tom.taylor.stds@gmail.com>
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Subject: Re: [sipcore] [dispatch] Passing of Francois Audet
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I'm really sorry to hear this. I worked with Francois on H.323 in 
particular, he representing the enterprise side and I the carrier side 
of the business. I was aware of his previous work with the ATM Forum, 
and of course, we both turned to SIP eventually.

Beyond being a pleasure to work with, Francois was probably one of the 
most competent standards representatives I've ever known. I'm sure he 
brought the same competence and focus to his PLM job at Skype, however 
long that lasted given his illness. My condolences to his present 
friends and family.

Tom Taylor

On 17/03/2014 11:56 AM, Mary Barnes wrote:
> A number of you all will remember Francois Audet, who was a colleague of
> mine at Nortel and a regular IETF attendee (until he went to Skype ;)  He
> was a co-author on the recently published 4244bis drafts.  He unfortunately
> lost his battle with cancer on Friday.  I have received the following
> information from a family friend via Facebook:
...


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(Please forgive the crosspost, and reply only to the rai list):
Please see
<http://mailarchive.ietf.org/arch/msg/rai/llvQvf7FWMRM-3ZV0R1aDE5wpRg>


From nobody Thu Mar 20 12:17:40 2014
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Subject: [sipcore] Problems with GRUUs (was: Comments on draft-allen-dispatch-routing-out-of-dialog-request-00)
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> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
> 
> I read this draft as well as the referenced draft 
> draft-kaplan-dispatch-gruu-problematic-00 (which somehow slipped by 
> without my noticing.)

There is a thread that we all participated in starting at
https://www.ietf.org/mail-archive/web/dispatch/current/msg02682.html.
Unfortunately, the discussion wasn't very conclusive and there was no
-01 version to clarify ambiguities or maintain interest.

Let me comment on the discussion of
draft-kaplan-dispatch-gruu-problematic-00 first:

There was an inconclusive discussion of the meaning of "globally
routable".  After thinking about it for a bit more, I think that we
need to qualify the term "globally routable", and doing so will make a
number of issues clearer further on.

The central question is, what is this "global" of which we speak?
After all, there are parts of the Internet from/to which SIP messages
are blocked, and so in principle there are *no* truly
globally-routable URIs.

What we have to think about is area within which a URI is routable.
To make this useful, we look an area, which we can call a "garden",
that is closed with regard to SIP operations:

    A. SIP dialogs cannot have one endpoint inside the garden and one
    endpoint outside the garden.

    B. Identifiers (including Call-Ids and Contact URIs) cannot leak from
    inside the garden to outside the garden.

(A garden can be a set of hosts or a set of host network interfaces,
or possibly even finer distinctions.)

(A draft of message used the term "universe", but SIP probably isn't
ready for multiple universes connected by B2BUAs.)

Note that a true B2BUA may have one interface inside a garden and
one interface outside the garden, because all dialogs that it
handles are terminated at one interface, and a coupled but distinct
dialog is initiated at the other interface.

When we label a URI as a GRUU, the needed property is that it is
globally routed *within the garden within which the GRUU exists*.

(My understanding is:)  In particular, the concatenation of all 3GPP
provider networks is a garden, because dialogs may neither exit nor
enter the the 3GPP walled garden, and the GRUUs that 3GPP entities
construct are globally routable within that garden.  Thus, these are
true GRUUs (within the 3GPP garden).

Now, back to draft-kaplan-dispatch-gruu-problematic-00 itself:

    3. Problems with Access Proxies 

This section of the draft discusses several problems, the first of
which is where (1) a UA A accesses an authoritative proxy AP via an
edge proxy EP, (2) UA A uses its GRUU as its Contact URI, and (3)
the far-end UA B makes a request within the dialog.

The dialog-forming request from A to B will typically have these
headers:

    A		EP		AP		B
    ----------->
     Contact: sip:a@example.com;gr=xxx
		--------------->
		  Contact: sip:a@example.com;gr=xxx
		  Record-Route: <sip:ep.example.com;lr>
				--------------->
				  Contact: sip:a@example.com;gr=xxx
				  Record-Route: <sip:ap.example.com;lr>
				  Record-Route: <sip:ep.example.com;lr>

Because the GRUU must be translated by AP, a request from B to A
within the dialog takes an unintended route:

    A		EP		AP		B
				  <--------------
				  METHOD sip:a@example.com;gr=xxx
				  Route: <sip:ap.example.com;lr>
				  Route: <sip:ep.example.com;lr>
		  <---------------
		  METHOD sip:a@example.com;gr=xxx
		  Route: <sip:ep.example.com;lr>
		  --------------->
		  METHOD sip:a@example.com;gr=xxx
		  <---------------
		  METHOD sip:10.1.1.1
		  Route: <sip:ep@example.com;lr>
     <------------
     METHOD sip:10.1.1.1

As the draft says:

   The obvious solution is for the originating UA (Alice) to insert a 
   Record-Route of her actual/real registered Contact-URI in the 
   dialog-forming request.  Arguably this should have been the model 
   from the beginning for [RFC5627].

The effect is that for in-dialog use, the contact URI contained in the
first Record-Route is used as the ultimate target of the request, but
for out-of-dialog use, the GRUU is used:

    A		EP		AP		B
    ----------->
     Contact: sip:a@example.com;gr=xxx
     Record-Route: <sip:10.1.1.1;lr>
		--------------->
		  Contact: sip:a@example.com;gr=xxx
		  Record-Route: <sip:ep.example.com;lr>
		  Record-Route: <sip:10.1.1.1;lr>
				--------------->
				  Contact: sip:a@example.com;gr=xxx
				  Record-Route: <sip:ap.example.com;lr>
				  Record-Route: <sip:ep.example.com;lr>
				  Record-Route: <sip:10.1.1.1;lr>

A subsequenct in-dialog request is routed properly:
				  <--------------
				  METHOD sip:a@example.com;gr=xxx
				  Route: <sip:ap.example.com;lr>
				  Route: <sip:ep.example.com;lr>
				  Route: <sip:10.1.1.1;lr>
		  <---------------
		  METHOD sip:a@example.com;gr=xxx
		  Route: <sip:ep.example.com;lr>
		  Route: <sip:10.1.1.1;lr>
     <------------
     METHOD sip:a@example.com;gr=xxx
     Route: <sip:10.1.1.1;lr>

(UA A, upon receiving the request, recognizes the remaining Route URI
as containing its own URI, and also recognizes the request-URI as
being its own URI.)

But an out-of-dialog request to A's Contact uses the GRUU:
				  <--------------
				  METHOD sip:a@example.com;gr=xxx
		  <---------------
		  METHOD sip:10.1.1.1
		  Route: <sip:ep.example.com;lr>
     <------------
      METHOD sip:10.1.1.1

In regard to practical deployments where UAs use GRUUs but not
implement this fix, the draft continues:

   An uglier but sometimes easier to 
   deploy solution is for the Access Proxy to insert a Record-Route in 
   the dialog-forming request on the originating UA's (Alice's) behalf, 
   using the Via information as a pseudo-Contact.  This latter method 
   has issues if the originating UA is not directly communicating with 
   the Access Proxy (i.e., if there is another Proxy in-between).  
   Access Proxies which are B2BUAs with registration caching (e.g., 
   SBC's) can insert a Record-Route of the matched registered contact, 
   instead, to avoid such issues. 

The second problem raised in this section is where the edge proxy is
not a proxy but rather a B2BUA, which in a quasi-proxy manner replaces
the Contact in A's dialog-forming request with its own Contact
sip:alias-a@ep.example.com, but maintains a mapping from
sip:alias-a@ep.example.com to the original Contact URI.

Assuming that this is the only change in the scenario, it results in
the same sort of poor routing:

    A		EP		AP		B
    ----------->
     Contact: sip:a@example.com;gr=xxx
		--------------->
		  Contact: sip:alias-a@ep.example.com
				--------------->
				  Contact: sip:alias-a@ep.example.com
				  Record-Route: <sip:ap.example.com;lr>
				  <--------------
				  METHOD sip:alias-a@ep.example.com
		  <---------------
		  METHOD sip:alias-a@ep.example.com
		  --------------->
		  METHOD sip:a@example.com;gr=xxx
		  <---------------
		  METHOD sip:10.1.1.1
		  Route: <sip:ep@example.com;lr>
     <------------
     METHOD sip:10.1.1.1

However, if UA A adds a protective Record-Route to its dialog-forming
request (or if EP adds one for it), and that route is maintained as
part of the mapping of sip:alias-a@ep.example.com, the desired
behavior is obtained:

    A		EP		AP		B
    ----------->
     Contact: sip:a@example.com;gr=xxx
     Record-Route: <sip:10.1.1.1;lr>
		--------------->
		  Contact: sip:alias-a@ep.example.com
				--------------->
				  Contact: sip:alias-a@ep.example.com
				  Record-Route: <sip:ap.example.com;lr>
				  <--------------
				  METHOD sip:alias-a@ep.example.com
				  Route: <sip:ap.example.com;lr>
		  <---------------
		  METHOD sip:alias-a@ep.example.com
     <------------
     METHOD sip:a@example.com;gr=xxx
     Route: <sip:10.1.1.1;lr>

However, there is a third problem posed which strikes closer to the
essential questions about GRUUs:

   This type of scenario actually occurred in initial deployments.  In 
   the initial deployment case, the Access Proxy was not inserting a 
   Record-Route of itself, but was instead a B2BUA inserting a new 
   Contact-URI of itself, which it mapped back to the original (GRUU) 
   Contact-URI.  The authoritative Registrar-Proxy never sees the GRUU 
   Contact nor a Record-Route.  Therefore, in the initial deployment, 
   the B2BUA received its own Contact URI in the Request-URI of the 
   BYE, mapped it to "sip:alice@example.com;gr=foo", and failed to 
   resolve that because it did not have any special/new handling for 
   GRUUs. 

Using the call flow immediately above here for reference, this
scenario is when EP translated sip:alias-a@ep.example.com into
sip:a@example.com;gr=xxx, EP was unable to resolve the new URI at all.
How could that come about?  After all, EP has an interface that can
talk to the proxy for domain example.com, and presumably sees suitable
DNS records.  By default, the result would be one of the annoying
"looping" flows.

What seems to be implied is that EP, *on its external interface*,
cannot see any resolution for domain example.com, that is, there is no
ingress SIP router for the domain example.com from that part of the
internet.  And this is not an unusual possibility in deployment.  But
comparing this scenario to my long-winded discussion of "gardens"
above, what is going on is that {EP<-->AP<-->B} is one garden, and
{A<-->EP} is a second garden -- and the GRUUs for example.com aren't
routable in the second, "exterior" garden.

Thus, the problem is not that the GRUU isn't globally routable per se,
but rather that UA A was allowed to learn of a (purported) GRUU *that
wasn't routable in the garden in which it sits*.

So what we see is that there is an additional requirement on B2BUAs
that sit on garden boundaries:  They have to make sure that a GRUU
cannot leak into a garden within which it is not routable.

In the above scenario, EP must be responsible for removing the GRUUs
from registration responses to A, because the network on which A sits
cannot route those GRUUs.  Once A cannot see its GRUU, it doesn't use
it as its Contact URI.

Of course, this makes the administrator of EP responsible for knowing
which networks that it is attached to can route URIs for the domain in
question, but presumably that's part of his job.  And for that matter,
with access to a network, one can experimentally determine whether SIP
requests addressed to a particular domain work or not, and where they
are routed to (for the first hop).

A further case is discussed:

   Even if the Access Proxy had been Record-Routing, however, it would 
   have failed in certain cases.  A request from Alice to Bob need not 
   go to any Proxy with access to the Registration database.  For 
   example, Access Proxies can typically route calls for Emergency 
   numbers directly to gateways/peers, or Bob could simply be a 
   conference server known to the Access Proxy to which it can route 
   directly.  In such cases, no authoritative Registrar-Proxy would be 
   in the path to perform the mid-dialog Request-URI rewriting.  
   Neither [RFC3261] nor [RFC5627] require all requests from a UA to 
   route through that UA's authoritative Registrar-Proxy, and yet that 
   is the only way a GRUU model could work as currently defined. 

This situation works correctly for within-dialog requests because they
do not require resolving the GRUU:

    A		EP		AP		B
    ----------->
     Contact: sip:a@example.com;gr=xxx
     Record-Route: <sip:10.1.1.1;lr>
		--------------->
		  Contact: sip:a@example.com;gr=xxx
		  Record-Route: <sip:ep.example.com;lr>
		  Record-Route: <sip:10.1.1.1;lr>
				--------------->
				  Contact: sip:a@example.com;gr=xxx
				  Record-Route: <sip:ep.example.com;lr>
				  Record-Route: <sip:10.1.1.1;lr>

		  <-----------------------------
		  METHOD sip:a@example.com;gr=xxx
		  Route: <sip:ep.example.com;lr>
		  Route: <sip:10.1.1.1;lr>
     <------------
     METHOD sip:a@example.com;gr=xxx
     Route: <sip:10.1.1.1;lr>

and if the destination is in a location where example.com is properly
resolvable, out-of-dialog requests to A's Contact are routed to AP
(due to their URI) and are resolved by AP:

				  <--------------
				  METHOD sip:a@example.com;gr=xxx
		  <---------------
		  METHOD sip:10.1.1.1
		  Route: <sip:ep.example.com;lr>
     <------------
      METHOD sip:10.1.1.1

But if B is in a location where example.com is not resolvable, the
out-of-dialog request would fail.  Assuming EP is a proxy, in that
situation A and B are in the same garden (because a dialog can travel
from one to the other), and since the garden containing A and B cannot
resolve the purported GRUU, UA A should not have been allowed to see
the GRUU.  If EP is a full B2BUA, then {EP<-->B} is a garden within
which the GRUU is not routable, and the B2BUA is responsible for not
presenting it as the Contact of the EP-to-B dialog.

Dale


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Subject: [sipcore] Problems with GRUUs (was: Comments on draft-allen-dispatch-routing-out-of-dialog-request-00)
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To continue with the analysis of
draft-kaplan-dispatch-gruu-problematic-00:

    4. Problems with B2BUAs 

To summarize this case:

    Because B2BUAs replace the Contact URI of forwarded requests, the 
    final SIP UAS will not receive the GRUU of the original UAC, and may 
    not receive a GRUU at all.  The Contact URI it receives will be the 
    URI of the last B2BUA crossed before the request reached the UAS, 
    which may not be a GRUU nor have GRUU properties.
    the Contact URI may contain an IP Address or hostname which is not 
    reachable from everywhere on the Internet, and/or it may only be 
    usable for the duration of the dialog its request created. 

    Attempts have been made to resolve this by having B2BUAs not change 
    the Contact URI if it is a GRUU (i.e., has a 'gr' param).  [...]
    However passing GRUUs through a B2BUA can only be done in very 
    specific conditions, and often with unforeseen consequences.  Since 
    any future out-of-dialog requests may use the GRUU as their target, 
    the B2BUA cannot expect to receive such requests if the GRUU is not 
    its own URI.  [...]

    In particular, if the B2BUA changed the Call-ID or tags of the 
    original dialog, as many do, then an out-of-dialog REFER or INVITE 
    with Replaces header won't work if it reaches the UA without passing 
    through the same B2BUA, because the dialog identifiers of the 
    Replaces won't be fixed by the B2BUA to match what the UA expects.  [...]

    Fundamentally, it is not logical for most B2BUAs to "pass on" a 
    GRUU.  [...]

    One option is for the B2UA to generate a self-made GRUU of its own, 
    which it can map to the Contact URI it received.  Unfortunately, by 
    definition such a GRUU's de-referenced B2BUA must be publicly 
    reachable on the public Internet, by *any device that receives it*, 
    on both IPv4 and IPv6.  Such is almost never the case, cannot be 
    known in advance, and may cause more issues than it solves.  

With reference to the previous discussion of "gardens", if the B2BUA
wants to provide the outgoing request with a GRUU Contact, and the
B2BUA wants to be in the routing of any additional dialog established
by sending a request to the outgoing GRUU, then the B2BUA must provide
one of its own GRUUs as the Contact.  Critically, that GRUU must be
globally routable *within to the garden into which the outgoing
request is being sent*, and presumably the B2BUA can arrange for that
to be true.  (If there is no DNS name that routes to the B2BUA within
that garden, then of course the B2BUA cannot create such a GRUU.  But
of course in that situation, there is no way for the B2BUA to provide
a GRUU Contact.)

(Note that by the analysis in the previous message, there should also
be a Record-Route header providing an IP address URI for the B2BUA,
unless the B2BUA is a target for its GRUU's domain name.)

In regard to IPv4 vs. IPv6, see the discussion of sectin 5.3.

The draft continues with this observation:

    5. Problems with Global Reachability 

    In theory, creating a GRUU which is not globally routable is simply 
    a user error or system defect, however in practice it's difficult if 
    not impossible for the system or user to know a priori whether a 
    GRUU is globally routable or not.

The essential point of the solution is that while it is impossible for
an entity that looks at an apparent GRUU to tell if the URI is
routable, entities that allow URIs to move between gardens are
responsible for ensuring that an apparent GRUU is globally routable
within the garden to which it is being sent.

    5.1. Access-Restricted Domains 

    In practice, even if they use SIP on the public Internet, most SIP 
    domains are access-restricted, meaning only authorized subscriber 
    UAs and known peers are allowed to send requests into the domain.  
    Any SIP request from unauthorized agents are rejected or discarded.  

Presumably, this is a situation in which the external interfaces of
the domain and its "authorized subscriber UAs and known peers" form a
garden, because clearly it is not intended for dialogs to reach from
those to any other destination on the Internet.  Thus, as long as the
domain's name has proper DNS records visible globally, and we are
willing to have out-of-dialog requests to GRUUs to pass into the
domain to its registrar, proper GRUUs can be constructed for the
domain and the exterior garden.

There is a messier case where the set of "authorized subscriber UAs
and known peers" do not form a garden, because they are allowed to
make calls to other UAs which are not in the authorized set.  But in
that case, some call transfer scenarios fail regardless of the URIs
that are being used, because the two UAs to be connected are not
permitted to contact each other.

    5.2. Routing-Restricted Domains 

    The problem with using GRUUs is not that routing-restricted domains 
    can't provide GRUUs to its UAs for use as Contact-URIs in outbound 
    requests - technically they could if they were not also access-
    restricted, although most if not all routing-restricted domains are 
    also access-restricted.  The real problem is that SIP requests *sent 
    to* the routing-restricted domain can't use GRUUs for Contact-URIs, 
    because the routing-restricted domain itself cannot de-reference a 
    GRUU unless the GRUU's domain happens to be one it can route to 
    based on its local policies.  Since there is virtually no way for 
    the originating or receiving domain to know where a GRUU used in a 
    Contact-URI may end up going to, or what the local policies of a 
    domain that attempts to de-reference the GRUU are, they can't be 
    used anytime safely. 

This is a scenario that needs further explanation.  As far as I
understand it, it's true that there is "no way for the originating or
receiving domain to know where a GRUU used in a Contact-URI may end up
going to", but the same is true of any other Contact URI.  There is an
unspoken assumption that there is an alternative way of constructing
Contact URIs that does not have these problems, but that alternative
way is not described.

    5.3. IPv6 Reachability 

    The stipulation for GRUUs in [RFC5627] is that they be globally 
    reachable on the "public Internet".  Unfortunately, there are *two* 
    public Internets: one using IPv4 and one using IPv6.  One of the 
    non-obvious implications of "global reachability" is that the domain 
    of the GRUU must be reachable on both, because the GRUU may be 
    passed on to IPv6-only UAs or domains, or vice-versa. 

In regard to IPv4 vs. IPv6, the administrators must decide whether the
relevant garden includes IPv6-only devices, and for that matter
IPv4-only devices.  If it is desired to be able to use GRUUs in a
garden that contains both IPv6-only devices and IPv4-only devices, SIP
listening entities must be established for each protocol, and suitable
DNS records must point to them.

    6. Alternatives to GRUU 

    Alternatively, some B2BUAs can generate "Global Contacts", [...]

But since "Global Contacts" are not described, I cannot comment on
them.

Dale


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Subject: Re: [sipcore] Problems with GRUUs
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I'm still reading and digesting your emails, but thought I'd try to =
comment on any smaller bits I can now...

On Mar 20, 2014, at 3:17 PM, Dale R. Worley <worley@ariadne.com> wrote:

> (My understanding is:)  In particular, the concatenation of all 3GPP
> provider networks is a garden, because dialogs may neither exit nor
> enter the the 3GPP walled garden, and the GRUUs that 3GPP entities
> construct are globally routable within that garden.  Thus, these are
> true GRUUs (within the 3GPP garden).

Not as far as I know. In certain situations that might be true - for =
example providers inter-connected through a common exchange network such =
as GSMA IPX might be able to use their own GRUUs amongst each other =
maybe - but in the classic scenario no: 3GPP provider-A may not have a =
direct connection to, nor route to, nor accept requests directly from, =
3GPP Provider-Z. Especially if Provider-A and Provider-Z are in =
different countries.

Even in the IPX case, a request from Provider-A's egress SBC, to =
Provider-B through Provider-B's ingress SBC, to some endpoint inside =
Provider-B... the endpoint cannot send an out-of-dialog request, using a =
received Provider-A GRUU Contact, directly to Provider-A. It must go to =
Provider-B's SBC first. (and this is a simplified scenario - in practice =
the initial dialog will have crossed through various B2BUAs inside =
Provider-B, and out-of-dialog ones need to go through *them* too)

-hadriel


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On Mar 20, 2014, at 3:17 PM, Dale R. Worley <worley@ariadne.com> wrote:

> The central question is, what is this "global" of which we speak?
> After all, there are parts of the Internet from/to which SIP messages
> are blocked, and so in principle there are *no* truly
> globally-routable URIs.
>=20
> What we have to think about is area within which a URI is routable.
> To make this useful, we look an area, which we can call a "garden",
> that is closed with regard to SIP operations:
>=20
>    A. SIP dialogs cannot have one endpoint inside the garden and one
>    endpoint outside the garden.
>=20
>    B. Identifiers (including Call-Ids and Contact URIs) cannot leak =
from
>    inside the garden to outside the garden.
>=20
> (A garden can be a set of hosts or a set of host network interfaces,
> or possibly even finer distinctions.)
>=20
> (A draft of message used the term "universe", but SIP probably isn't
> ready for multiple universes connected by B2BUAs.)
>=20
> Note that a true B2BUA may have one interface inside a garden and
> one interface outside the garden, because all dialogs that it
> handles are terminated at one interface, and a coupled but distinct
> dialog is initiated at the other interface.
>=20
> When we label a URI as a GRUU, the needed property is that it is
> globally routed *within the garden within which the GRUU exists*.

Then GRUUs serve no practical purpose, because the above definition is =
what just any plane-Jane RFC 3261 Contact URI already accomplishes.  The =
whole *point* of defining the GRUU mechanism was to have a Contact URI =
which could be used to reach a specific endpoint from *anywhere*.  =
Right?

If we're going to constrain the scope of what "anywhere" means, to boil =
down to: "so long as we can talk to each other", then we've done nothing =
but added a 'gr' param to a URI, afaict.

Or am I misunderstanding you?

-hadriel


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Subject: [sipcore] Comments on draft-allen-dispatch-routing-out-of-dialog-request-00
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To get back to the main item of business,
draft-allen-dispatch-routing-out-of-dialog-request-00:

In general, it would be useful to see examples of the failures that we
are trying to prevent, and examples of how the failures are prevented
by the new mechanism(s).  One reason is that failures often depend
fairly sensitively on the details of the intended call flow, so it's
difficult to know that one is addressing the same scenario that the
draft is.

Also, generally, it would be useful to perform some sort of
interoperability analysis on any new mechanisms to verify that they
don't impose a burden on entities that are outside of the
administrative domains that deploy a new mechanism.

   However, if REFER and SUBSCRIBE requests are sent outside the related
   existing dialog then the requests will not be routed by the
   manipulating B2BUA and thus will either [...] fail to arrive at the
   remote UA due to URI manipulations or fail at the remote UA because
   parameters in the request (e.g Target-Dialog, Replaces, Refer-To URI,
   etc) do not match the values at the remote UAS.

Of course, if either (1) the B2BUA must be included in any additional
dialog, or (2) the B2BUA changes the dialog identifiers, then the
B2BUA has to place one of its URIs as the Contact URI.  But if it does
so, it will then see any out-of-dialog requests, it can then adjust
the request appropriately before passing it along.

For instance, set up a call from A to B:

    A           EP              B
    ----------->
     INVITE sip:b@example.com
     Call-Id: P
     Contact: sip:a@example.com
                --------------->
                  INVITE sip:b@example.com
                  Call-Id: Q
                  Contact: sip:a-external@ep.example.com
                <---------------
                  200 OK
                  Call-Id: Q
                  Contact: sip:b@example.com
    <-----------
     200 OK
     Call-Id: P
     Contact: sip:b-internal@ep.example.com

Now A sends an out-of-dialog REFER to B to cause it to transfer to C.
The REFER gets routed to EP because A uses the alias for B that EP
provided.  Additionally, EP translates the Target-Dialog header.  And
if EP desires to be in the new dialog that will be created, it can
translate the Refer-To header as well:

    ----------->
     REFER sip:b-internal@ep.example.com
     Call-Id: R
     Contact: sip:a@example.com
     Target-Dialog: P;local-tag=xxx;remote-tag=xxx
     Refer-To: sip:c@example.com
                --------------->
                  REFER sip:b@ep.example.com
                  Call-Id: S
                  Contact: sip:a-external@ep.example.com
                  Target-Dialog: Q;local-tag=xxx;remote-tag=xxx
                  Refer-To: sip:c-external@ep.example.com

                <---------------
                  INVITE sip:c-external@ep.example.com
                  Call-Id: T
                  Contact: sip:b@example.com
                -------------------------------------->
                  INVITE sip:c@example.com
                  Call-Id: U
                  Contact: sip:b-external@ep.example.com

Returning to the draft:

   One way B2BUAs have have addressed this problem is by acting as two
   UAs back-to-back with the Contact URI being overwritten to be the URI
   of the B2BUA.  However this means that GRUU of the UA is overwritten
   by the B2BUA and the meaning of the Contact header field parameters
   becomes obscure.  Do the Contact header field parameters reflect the
   capabilities of the Contact address (i.e the B2BUA) or do they
   reflect the capabilities of the remote UA?  If they relect the
   capabilities of the B2BUA then the identfication of the capabilities
   of the remote UAS has been lost.  If they reflect the capabilities of
   the remote UA then they falsely identify that the B2BUA contact
   address has the capabilities of the remote UA.

(BTW, aren't Contact header field parameters described by RFC 3840
rather than RFC 6809 (which is your reference)?)

Assuming that the B2BUA is largely transparent to SIP requests and
SDP, then the capabilities of the B2BUA contact address *are the same*
as the capabilities of the remote UA.  If the remote UA advertises
"video" then the B2BUA contact also has video capabilities, etc.

Of course, if the B2BUA is somehow restricting or editing what passes
through, then it must properly edit the Contact parameters to remove
any capabilities that it is blocking.

Dale


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Subject: Re: [sipcore] Problems with GRUUs (was: Comments on draft-allen-dispatch-routing-out-of-dialog-request-00)
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> From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
> 
> > When we label a URI as a GRUU, the needed property is that it is
> > globally routed *within the garden within which the GRUU exists*.
> 
> Then GRUUs serve no practical purpose, because the above definition
> is what just any plane-Jane RFC 3261 Contact URI already
> accomplishes.  The whole *point* of defining the GRUU mechanism was
> to have a Contact URI which could be used to reach a specific
> endpoint from *anywhere*.  Right?
> 
> If we're going to constrain the scope of what "anywhere" means, to
> boil down to: "so long as we can talk to each other", then we've
> done nothing but added a 'gr' param to a URI, afaict.
> 
> Or am I misunderstanding you?

You're misunderstanding me, but the distinction is tricky, and it's
possible that I haven't forumulated it with sufficient accuracy.

The critical point of a GRUU is that *if an entity can see that URI*
then *that entity can use it for contacting that UA*.  Now, in a flat,
end-to-end network, that's nearly the same as the old definition.  (In
parts of the network that aren't accessible to SIP at all, there is no
entity that can see the URI, and so the fact that they can't contact
the UA is irrelevant.)

What makes this interesting is the situation where the network is
partitioned by SBCs, that is, where SIP traffic can get from one
garden to another only by passing through an SBC.  Even if the GRUUs
for the selected domain work fine in one garden (ex. the interior of
the organization), they might not be usable at all in the other garden
(ex. the public internet).  But the fact that the GRUU works fine in
one garden is an interesting fact, and they work just like one would
hope within that garden.

What raises gardens from just being subsets of the network is that
URIs can be carried from one entity to another in various ways, so a
garden has to be "sealed off" from the rest of the network in a way
that prevents URIs (and dialog identifiers) from leaking out of the
garden.

Functionally, the point is that SBCs have to be aware, for each
network segment that it is connected to, whether GRUU URIs for its
domain are valid there, and censor (or otherwise revise) GRUUs that
would otherwise be propagated into segments they are not valid for.

Dale


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>, Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [sipcore] Problems with GRUUs (was: Comments on draft-allen-dispatch-routing-out-of-dialog-request-00)
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Subject: Re: [sipcore] Problems with GRUUs (was: Comments on draft-allen-dispatch-routing-out-of-dialog-request-00)
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Hi,

>> Then GRUUs serve no practical purpose, because the above definition is=20
>> what just any plane-Jane RFC 3261 Contact URI already accomplishes. =20
>> The whole *point* of defining the GRUU mechanism was to have a Contact=20
>> URI which could be used to reach a specific endpoint from *anywhere*. =20
>> Right?
>>=20
>> If we're going to constrain the scope of what "anywhere" means, to=20
>> boil down to: "so long as we can talk to each other", then we've done=20
>> nothing but added a 'gr' param to a URI, afaict.
>>=20
>> Or am I misunderstanding you?
>
>You're misunderstanding me, but the distinction is tricky, and it's possib=
le that I haven't >forumulated it with sufficient accuracy.
>
>The critical point of a GRUU is that *if an entity can see that URI* then =
*that entity can use it for >contacting that UA*.  Now, in a flat, end-to-e=
nd network, that's nearly the same as the old >definition.  (In parts of th=
e network that aren't accessible to SIP at all, there is no entity that can=
 see >the URI, and so the fact that they can't contact the UA is irrelevant=
.)
>
>What makes this interesting is the situation where the network is partitio=
ned by SBCs, that is, where >SIP traffic can get from one garden to another=
 only by passing through an SBC.  Even if the GRUUs >for the selected domai=
n work fine in one garden (ex. the interior of the organization), they migh=
t >not be usable at all in the other garden (ex. the public internet).  But=
 the fact that the GRUU works >fine in one garden is an interesting fact, a=
nd they work just like one would hope within that garden.

...and that's the reason such GRUUs shouldn't be passed into the other gard=
en to begin with - and it's the reason why B2BUAs often remove/overwrite th=
em.

Regards,

Christer


From nobody Fri Mar 21 12:05:23 2014
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Subject: Re: [sipcore] [dispatch] Comments on draft-allen-dispatch-routing-out-of-dialog-request-00
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[moved from Dispatch]

> From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
>
> On Mar 16, 2014, at 8:54 PM, Andrew Allen <aallen@blackberry.com> wrote:
> > Having B2BUAs only include feature tags they understand to my view
> means no innovation and no new services. B2BUAs have then become the
> switches of old needing all to be updated or reconfigured to support
> any new services.
> 
> Yup, and that's exactly the right thing to happen.  It's what most
> IMS operators actually should *want* to happen as well.

> It's really not a bad thing to want service stability.

You can do it whichever way you prefer.  The B2BUA can copy all of the
feature tags if it is also transparent (proxy-like) enough that it
isn't goint to hinder any new features.  But if the B2BUA wants to
enforce stability, and goes to the effort to edit the SIP/SDP in
detail, then it should perform the corresponding editing of the
feature tags.

Dale


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Subject: Re: [sipcore] [dispatch] Comments on draft-allen-dispatch-routing-out-of-dialog-request-00
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[moved from Dispatch]

> From: Andrew Allen <aallen@blackberry.com>
> 
> In addition as I state in my draft modifying the Contact header
> field means either the media feature tags of the remote UA are lost
> or the B2BUA is saying if you send a request to this address
> (including potentially after this session has ended) you can expect
> that this feature is supported.

Yes, but ...  The Contact URI that the B2BUA uses is really a stand-in
for the "real" Contact URI.  That is, the B2BUA has to use *different*
Contact URIs depending for each original Contact URI.

But it has to be doing that already if out-of-dialog requests are
going to work at all.  If the B2BUA receives an OOD REFER, it has to
know which destination "on the other side" to send the REFER to, and
it has only the request-URI to tell it that.

But if you are using different Contact URIs for each actual
destination, then it isn't confusing to provide different media
feature tags for diffent Contact URIs.

Dale


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Subject: Re: [sipcore] [dispatch] Comments on draft-allen-dispatch-routing-out-of-dialog-request-00
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[moved from Dispatch]

> From: Hadriel Kaplan <hadriel.kaplan@oracle.com>

> There is no problem I know of, so long as you do NOT use GRUUs, or
> ignore the 'gr' param and just behave like they're normal contact
> URIs.

I'm curious what you mean by "just behave like they're normal contact
URIs".  My understanding is that the contact URI is used for whatever
it's used for *because* it's the contact URI.  If it's marked as a
GRUU, that presumably is a hint that your odds of success are higher,
but what would you do differently?

As far as I know, the one change is that if the contact URI is marked
as a GRUU, the UA is more likely to attempt to send REFERs for the
dialog out-of-dialog.  As you say in
draft-kaplan-dispatch-gruu-problematic-00:

   Furthermore, [RFC5589] prescribes sending a REFER outside of the 
   INVITE dialog if a GRUU is available, since with a GRUU the 
   Transferor "knows the Contact URI is routable outside the dialog".  

and in RFC 5589:

   As a result, if the Transferee supports the target-dialog extension
   and the Transferor knows the Contact URI is routable outside the
   dialog, the REFER SHOULD be sent in a new dialog.  If the nature of
   the Contact URI is not known or if support for the target-dialog
   extension is not known, the REFER SHOULD be sent inside the existing
   dialog.  A Transferee MUST be prepared to receive a REFER either
   inside or outside a dialog.  One way that a Transferor could know
   that a Contact URI is routable outside a dialog is by validation
   (e.g., sending an OPTIONS and receiving a response) or if it
   satisfies the properties described in the GRUU specification
   [SIP-GRUU].

Is this the entirety of what you mean by "behaving as if it's a GRUU"?

In your draft, you also write:

   Instead, a logically similar model has been successfully deployed 
   using "Global Contacts", which are Contact URIs having the 
   properties of a GRUU, but not being a GRUU in syntax (i.e., without 
   a 'gr' param, and not indicating an AoR).  This has the benefit that 
   downstream B2BUAs/UAs will not treat the Contact as a special GRUU 
   case. Global Contacts are not always valid after their dialog is 
   terminated, but it remains to be seen if this will cause problems in 
   the future.  Currently the main use-case for GRUU-like properties is 
   for call-transfer scenarios, for which dialog lifetime is 
   sufficient. 

Could you give us some detail as to what a "global contact" is and how
it works?

Dale


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Subject: Re: [sipcore] Problems with GRUUs
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Dale,

I don't know if I agree with your definitions here.

I see two distinct issues here:

- uniqueness of addresses (both DNS and IP)
- reachability

IMO, when we are talking about gruus, what we are really talking about 
is primarily global uniqueness. The registrar-provided gruus are 
intended to *help* with this for devices behind NATs, so that they can 
get a globally unique address.

Even with globally unique addresses there is no guarantee of 
reachability. Some devices may have a policy about who may reach them, 
and how. If their globally reachable address is used some other way they 
may block the request. And they may utilize intermediaries to help them 
enforce the policy. I don't think SIP considers such behavior wrong.

But, if UA-B intends that it be possible for UA-A to be able to send 
out-of-dialog requests to it, then when it provides a contact address, 
it should provide one that will *work* for UA-A to reach UA-B 
out-of-dialog. This should be part of the sip contract, and GRUU is an 
assistance to achieving that.

If UA-B doesn't want to be reached except in-dialog, then this doesn't 
apply. But then it must accept that many features won't work, because 
those may well require out-of-dialog access.

	Thanks,
	Paul

On 3/20/14 3:17 PM, Dale R. Worley wrote:
>> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
>>
>> I read this draft as well as the referenced draft
>> draft-kaplan-dispatch-gruu-problematic-00 (which somehow slipped by
>> without my noticing.)
>
> There is a thread that we all participated in starting at
> https://www.ietf.org/mail-archive/web/dispatch/current/msg02682.html.
> Unfortunately, the discussion wasn't very conclusive and there was no
> -01 version to clarify ambiguities or maintain interest.
>
> Let me comment on the discussion of
> draft-kaplan-dispatch-gruu-problematic-00 first:
>
> There was an inconclusive discussion of the meaning of "globally
> routable".  After thinking about it for a bit more, I think that we
> need to qualify the term "globally routable", and doing so will make a
> number of issues clearer further on.
>
> The central question is, what is this "global" of which we speak?
> After all, there are parts of the Internet from/to which SIP messages
> are blocked, and so in principle there are *no* truly
> globally-routable URIs.
>
> What we have to think about is area within which a URI is routable.
> To make this useful, we look an area, which we can call a "garden",
> that is closed with regard to SIP operations:
>
>      A. SIP dialogs cannot have one endpoint inside the garden and one
>      endpoint outside the garden.
>
>      B. Identifiers (including Call-Ids and Contact URIs) cannot leak from
>      inside the garden to outside the garden.
>
> (A garden can be a set of hosts or a set of host network interfaces,
> or possibly even finer distinctions.)
>
> (A draft of message used the term "universe", but SIP probably isn't
> ready for multiple universes connected by B2BUAs.)
>
> Note that a true B2BUA may have one interface inside a garden and
> one interface outside the garden, because all dialogs that it
> handles are terminated at one interface, and a coupled but distinct
> dialog is initiated at the other interface.
>
> When we label a URI as a GRUU, the needed property is that it is
> globally routed *within the garden within which the GRUU exists*.
>
> (My understanding is:)  In particular, the concatenation of all 3GPP
> provider networks is a garden, because dialogs may neither exit nor
> enter the the 3GPP walled garden, and the GRUUs that 3GPP entities
> construct are globally routable within that garden.  Thus, these are
> true GRUUs (within the 3GPP garden).
>
> Now, back to draft-kaplan-dispatch-gruu-problematic-00 itself:
>
>      3. Problems with Access Proxies
>
> This section of the draft discusses several problems, the first of
> which is where (1) a UA A accesses an authoritative proxy AP via an
> edge proxy EP, (2) UA A uses its GRUU as its Contact URI, and (3)
> the far-end UA B makes a request within the dialog.
>
> The dialog-forming request from A to B will typically have these
> headers:
>
>      A		EP		AP		B
>      ----------->
>       Contact: sip:a@example.com;gr=xxx
> 		--------------->
> 		  Contact: sip:a@example.com;gr=xxx
> 		  Record-Route: <sip:ep.example.com;lr>
> 				--------------->
> 				  Contact: sip:a@example.com;gr=xxx
> 				  Record-Route: <sip:ap.example.com;lr>
> 				  Record-Route: <sip:ep.example.com;lr>
>
> Because the GRUU must be translated by AP, a request from B to A
> within the dialog takes an unintended route:
>
>      A		EP		AP		B
> 				  <--------------
> 				  METHOD sip:a@example.com;gr=xxx
> 				  Route: <sip:ap.example.com;lr>
> 				  Route: <sip:ep.example.com;lr>
> 		  <---------------
> 		  METHOD sip:a@example.com;gr=xxx
> 		  Route: <sip:ep.example.com;lr>
> 		  --------------->
> 		  METHOD sip:a@example.com;gr=xxx
> 		  <---------------
> 		  METHOD sip:10.1.1.1
> 		  Route: <sip:ep@example.com;lr>
>       <------------
>       METHOD sip:10.1.1.1
>
> As the draft says:
>
>     The obvious solution is for the originating UA (Alice) to insert a
>     Record-Route of her actual/real registered Contact-URI in the
>     dialog-forming request.  Arguably this should have been the model
>     from the beginning for [RFC5627].
>
> The effect is that for in-dialog use, the contact URI contained in the
> first Record-Route is used as the ultimate target of the request, but
> for out-of-dialog use, the GRUU is used:
>
>      A		EP		AP		B
>      ----------->
>       Contact: sip:a@example.com;gr=xxx
>       Record-Route: <sip:10.1.1.1;lr>
> 		--------------->
> 		  Contact: sip:a@example.com;gr=xxx
> 		  Record-Route: <sip:ep.example.com;lr>
> 		  Record-Route: <sip:10.1.1.1;lr>
> 				--------------->
> 				  Contact: sip:a@example.com;gr=xxx
> 				  Record-Route: <sip:ap.example.com;lr>
> 				  Record-Route: <sip:ep.example.com;lr>
> 				  Record-Route: <sip:10.1.1.1;lr>
>
> A subsequenct in-dialog request is routed properly:
> 				  <--------------
> 				  METHOD sip:a@example.com;gr=xxx
> 				  Route: <sip:ap.example.com;lr>
> 				  Route: <sip:ep.example.com;lr>
> 				  Route: <sip:10.1.1.1;lr>
> 		  <---------------
> 		  METHOD sip:a@example.com;gr=xxx
> 		  Route: <sip:ep.example.com;lr>
> 		  Route: <sip:10.1.1.1;lr>
>       <------------
>       METHOD sip:a@example.com;gr=xxx
>       Route: <sip:10.1.1.1;lr>
>
> (UA A, upon receiving the request, recognizes the remaining Route URI
> as containing its own URI, and also recognizes the request-URI as
> being its own URI.)
>
> But an out-of-dialog request to A's Contact uses the GRUU:
> 				  <--------------
> 				  METHOD sip:a@example.com;gr=xxx
> 		  <---------------
> 		  METHOD sip:10.1.1.1
> 		  Route: <sip:ep.example.com;lr>
>       <------------
>        METHOD sip:10.1.1.1
>
> In regard to practical deployments where UAs use GRUUs but not
> implement this fix, the draft continues:
>
>     An uglier but sometimes easier to
>     deploy solution is for the Access Proxy to insert a Record-Route in
>     the dialog-forming request on the originating UA's (Alice's) behalf,
>     using the Via information as a pseudo-Contact.  This latter method
>     has issues if the originating UA is not directly communicating with
>     the Access Proxy (i.e., if there is another Proxy in-between).
>     Access Proxies which are B2BUAs with registration caching (e.g.,
>     SBC's) can insert a Record-Route of the matched registered contact,
>     instead, to avoid such issues.
>
> The second problem raised in this section is where the edge proxy is
> not a proxy but rather a B2BUA, which in a quasi-proxy manner replaces
> the Contact in A's dialog-forming request with its own Contact
> sip:alias-a@ep.example.com, but maintains a mapping from
> sip:alias-a@ep.example.com to the original Contact URI.
>
> Assuming that this is the only change in the scenario, it results in
> the same sort of poor routing:
>
>      A		EP		AP		B
>      ----------->
>       Contact: sip:a@example.com;gr=xxx
> 		--------------->
> 		  Contact: sip:alias-a@ep.example.com
> 				--------------->
> 				  Contact: sip:alias-a@ep.example.com
> 				  Record-Route: <sip:ap.example.com;lr>
> 				  <--------------
> 				  METHOD sip:alias-a@ep.example.com
> 		  <---------------
> 		  METHOD sip:alias-a@ep.example.com
> 		  --------------->
> 		  METHOD sip:a@example.com;gr=xxx
> 		  <---------------
> 		  METHOD sip:10.1.1.1
> 		  Route: <sip:ep@example.com;lr>
>       <------------
>       METHOD sip:10.1.1.1
>
> However, if UA A adds a protective Record-Route to its dialog-forming
> request (or if EP adds one for it), and that route is maintained as
> part of the mapping of sip:alias-a@ep.example.com, the desired
> behavior is obtained:
>
>      A		EP		AP		B
>      ----------->
>       Contact: sip:a@example.com;gr=xxx
>       Record-Route: <sip:10.1.1.1;lr>
> 		--------------->
> 		  Contact: sip:alias-a@ep.example.com
> 				--------------->
> 				  Contact: sip:alias-a@ep.example.com
> 				  Record-Route: <sip:ap.example.com;lr>
> 				  <--------------
> 				  METHOD sip:alias-a@ep.example.com
> 				  Route: <sip:ap.example.com;lr>
> 		  <---------------
> 		  METHOD sip:alias-a@ep.example.com
>       <------------
>       METHOD sip:a@example.com;gr=xxx
>       Route: <sip:10.1.1.1;lr>
>
> However, there is a third problem posed which strikes closer to the
> essential questions about GRUUs:
>
>     This type of scenario actually occurred in initial deployments.  In
>     the initial deployment case, the Access Proxy was not inserting a
>     Record-Route of itself, but was instead a B2BUA inserting a new
>     Contact-URI of itself, which it mapped back to the original (GRUU)
>     Contact-URI.  The authoritative Registrar-Proxy never sees the GRUU
>     Contact nor a Record-Route.  Therefore, in the initial deployment,
>     the B2BUA received its own Contact URI in the Request-URI of the
>     BYE, mapped it to "sip:alice@example.com;gr=foo", and failed to
>     resolve that because it did not have any special/new handling for
>     GRUUs.
>
> Using the call flow immediately above here for reference, this
> scenario is when EP translated sip:alias-a@ep.example.com into
> sip:a@example.com;gr=xxx, EP was unable to resolve the new URI at all.
> How could that come about?  After all, EP has an interface that can
> talk to the proxy for domain example.com, and presumably sees suitable
> DNS records.  By default, the result would be one of the annoying
> "looping" flows.
>
> What seems to be implied is that EP, *on its external interface*,
> cannot see any resolution for domain example.com, that is, there is no
> ingress SIP router for the domain example.com from that part of the
> internet.  And this is not an unusual possibility in deployment.  But
> comparing this scenario to my long-winded discussion of "gardens"
> above, what is going on is that {EP<-->AP<-->B} is one garden, and
> {A<-->EP} is a second garden -- and the GRUUs for example.com aren't
> routable in the second, "exterior" garden.
>
> Thus, the problem is not that the GRUU isn't globally routable per se,
> but rather that UA A was allowed to learn of a (purported) GRUU *that
> wasn't routable in the garden in which it sits*.
>
> So what we see is that there is an additional requirement on B2BUAs
> that sit on garden boundaries:  They have to make sure that a GRUU
> cannot leak into a garden within which it is not routable.
>
> In the above scenario, EP must be responsible for removing the GRUUs
> from registration responses to A, because the network on which A sits
> cannot route those GRUUs.  Once A cannot see its GRUU, it doesn't use
> it as its Contact URI.
>
> Of course, this makes the administrator of EP responsible for knowing
> which networks that it is attached to can route URIs for the domain in
> question, but presumably that's part of his job.  And for that matter,
> with access to a network, one can experimentally determine whether SIP
> requests addressed to a particular domain work or not, and where they
> are routed to (for the first hop).
>
> A further case is discussed:
>
>     Even if the Access Proxy had been Record-Routing, however, it would
>     have failed in certain cases.  A request from Alice to Bob need not
>     go to any Proxy with access to the Registration database.  For
>     example, Access Proxies can typically route calls for Emergency
>     numbers directly to gateways/peers, or Bob could simply be a
>     conference server known to the Access Proxy to which it can route
>     directly.  In such cases, no authoritative Registrar-Proxy would be
>     in the path to perform the mid-dialog Request-URI rewriting.
>     Neither [RFC3261] nor [RFC5627] require all requests from a UA to
>     route through that UA's authoritative Registrar-Proxy, and yet that
>     is the only way a GRUU model could work as currently defined.
>
> This situation works correctly for within-dialog requests because they
> do not require resolving the GRUU:
>
>      A		EP		AP		B
>      ----------->
>       Contact: sip:a@example.com;gr=xxx
>       Record-Route: <sip:10.1.1.1;lr>
> 		--------------->
> 		  Contact: sip:a@example.com;gr=xxx
> 		  Record-Route: <sip:ep.example.com;lr>
> 		  Record-Route: <sip:10.1.1.1;lr>
> 				--------------->
> 				  Contact: sip:a@example.com;gr=xxx
> 				  Record-Route: <sip:ep.example.com;lr>
> 				  Record-Route: <sip:10.1.1.1;lr>
>
> 		  <-----------------------------
> 		  METHOD sip:a@example.com;gr=xxx
> 		  Route: <sip:ep.example.com;lr>
> 		  Route: <sip:10.1.1.1;lr>
>       <------------
>       METHOD sip:a@example.com;gr=xxx
>       Route: <sip:10.1.1.1;lr>
>
> and if the destination is in a location where example.com is properly
> resolvable, out-of-dialog requests to A's Contact are routed to AP
> (due to their URI) and are resolved by AP:
>
> 				  <--------------
> 				  METHOD sip:a@example.com;gr=xxx
> 		  <---------------
> 		  METHOD sip:10.1.1.1
> 		  Route: <sip:ep.example.com;lr>
>       <------------
>        METHOD sip:10.1.1.1
>
> But if B is in a location where example.com is not resolvable, the
> out-of-dialog request would fail.  Assuming EP is a proxy, in that
> situation A and B are in the same garden (because a dialog can travel
> from one to the other), and since the garden containing A and B cannot
> resolve the purported GRUU, UA A should not have been allowed to see
> the GRUU.  If EP is a full B2BUA, then {EP<-->B} is a garden within
> which the GRUU is not routable, and the B2BUA is responsible for not
> presenting it as the Contact of the EP-to-B dialog.
>
> Dale
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


From nobody Fri Mar 21 14:37:14 2014
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Subject: [sipcore] SIPCORE Minutes, IETF89, March 2014
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The raw notes from the SIPCORE sessions, along with a summary of 
conclusions and action items have been posted for the IETF89 
proceedings. You can find them at:

https://datatracker.ietf.org/secr/proceedings/89/sipcore

I've included the summary below. Please note the action items!

	Thanks,
	Paul

=================== Conclusions & Action Items  ===================

* Errata on 3325:

Conclusion: No objection to interpretation given by chairs.

Action: Agree to clarify that it is allowed, but not required, to 
interpret these characters when not enclosed in <>.

* draft-johansson-dispatch-dane-sip:

Conclusion: Few present understood the issue. Many were confused.

Action: Olle to send use cases to the list.

* draft-yusef-sipcore-digest-scheme:

Conclusion: Don't fix Digest, fix Authentication

Action: Those who think fixing digest is insufficient should send use 
cases to the sipcore mailing list.

* draft-johansson-sip-dual-stack

Action: split the action item into two: one for the DNS part, and 
another for the remainder of Happy Eyeballs. Discuss the milestone text 
on the list.

Action: call for adoption of draft-johansson-dispatch-dane-sip on the 
list for the DNS part.

* RFC 6665 Dialog Re-Use with REFER

Conclusion: Much objection to need for GRUU to solve this problem.

Action: Robert Sparks and Adam Roach to write a new draft updating 
RFC3515 (REFER) for consistency with RFC6665.
Action: Continue discussion on list. (Especially those who object to 
described approach.)


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On 21 Mar 2014, at 22:37, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> The raw notes from the SIPCORE sessions, along with a summary of =
conclusions and action items have been posted for the IETF89 =
proceedings. You can find them at:
>=20
> https://datatracker.ietf.org/secr/proceedings/89/sipcore
>=20
> I've included the summary below. Please note the action items!
>=20
> 	Thanks,
> 	Paul
>=20
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D Conclusions =
& Action Items  =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> * Errata on 3325:
>=20
> Conclusion: No objection to interpretation given by chairs.
>=20
> Action: Agree to clarify that it is allowed, but not required, to =
interpret these characters when not enclosed in <>.
>=20
> * draft-johansson-dispatch-dane-sip:
>=20
> Conclusion: Few present understood the issue. Many were confused.
>=20
> Action: Olle to send use cases to the list.
I don't think the confusion was about use cases at all or whether we =
need DANE or not - it was about whether the DANE group documents was =
enough,
if this is just a bump in the TLS stack or if we actually need to do =
something in SIPcore. Some people claimed
we need to do zero, nothing, nada. Some people (including me) claims we =
do need to document this,
especially since the SIP Domain certificate RFC is an update to RFC 3261 =
and DANE usage changes
the certificate contents.

The DANE group is very clear that we do need to document our usage. Wes =
Hardaker pointed that out
several times during the discussion. It's documented in the DANE SRV =
drafts too.

>=20
> * draft-yusef-sipcore-digest-scheme:
>=20
> Conclusion: Don't fix Digest, fix Authentication
>=20
> Action: Those who think fixing digest is insufficient should send use =
cases to the sipcore mailing list.
>=20
> * draft-johansson-sip-dual-stack
>=20
> Action: split the action item into two: one for the DNS part, and =
another for the remainder of Happy Eyeballs. Discuss the milestone text =
on the list.
>=20
> Action: call for adoption of draft-johansson-dispatch-dane-sip on the =
list for the DNS part.
Our draft only includes the DNS part so the full draft can be adopted as =
is. THere's no draft covering the Happy Eyeballs part yet.

/O

>=20
> * RFC 6665 Dialog Re-Use with REFER
>=20
> Conclusion: Much objection to need for GRUU to solve this problem.
>=20
> Action: Robert Sparks and Adam Roach to write a new draft updating =
RFC3515 (REFER) for consistency with RFC6665.
> Action: Continue discussion on list. (Especially those who object to =
described approach.)
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Sat Mar 22 02:14:14 2014
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [sipcore] SIPCORE Minutes, IETF89, March 2014 - RFC 6665 Dialog Re-Use with REFER
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Subject: Re: [sipcore] SIPCORE Minutes, IETF89, March 2014 - RFC 6665 Dialog Re-Use with REFER
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Hi,

> * RFC 6665 Dialog Re-Use with REFER
>=20
> Conclusion: Much objection to need for GRUU to solve this problem.
>=20
> Action: Robert Sparks and Adam Roach to write a new draft updating RFC351=
5 (REFER) for consistency with RFC6665.
> Action: Continue discussion on list. (Especially those who object to=20
> described approach.)

I realize this is supposed to be a summary, but I think some parts are miss=
ing.

First, I am not sure what is meant by "need for GRUU to solve this problems=
".=20

Eventhough GRUU, and whether 3515 needs to be updated, was discussed, the o=
ther part of the discussion was not so much about GRUU - it was more about =
usage of in-dialog REFERs when the sender "knows" that receiver will use no=
refersub.=20

Related to that, I think it would be good to mention that there were sugges=
tions about deprecating implicit registrations all together, and defining a=
 new nofersub-must-support-AND-use option-tag, but that no conclusions were=
 made.

Thanks!

Regards,

Christer


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Subject: Re: [sipcore] Problems with GRUUs (was: Comments on draft-allen-dispatch-routing-out-of-dialog-request-00)
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On Mar 21, 2014, at 2:19 PM, Dale R. Worley <worley@ariadne.com> wrote:

>> From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
>>=20
>>> When we label a URI as a GRUU, the needed property is that it is
>>> globally routed *within the garden within which the GRUU exists*.
>>=20
>> Then GRUUs serve no practical purpose, because the above definition
>> is what just any plane-Jane RFC 3261 Contact URI already
>> accomplishes.  The whole *point* of defining the GRUU mechanism was
>> to have a Contact URI which could be used to reach a specific
>> endpoint from *anywhere*.  Right?
>>=20
>> If we're going to constrain the scope of what "anywhere" means, to
>> boil down to: "so long as we can talk to each other", then we've
>> done nothing but added a 'gr' param to a URI, afaict.
>>=20
>> Or am I misunderstanding you?
>=20
> You're misunderstanding me, but the distinction is tricky, and it's
> possible that I haven't forumulated it with sufficient accuracy.
>=20
> The critical point of a GRUU is that *if an entity can see that URI*
> then *that entity can use it for contacting that UA*.

OK, I'm following you now I think.  I think the difference is just that: =
such a GRUU contact could be used instead of an AoR target, whereas a =
3261 Contact couldn't (safely) be used as an AoR target. (ie, in =
out-of-dialog requests)

Yes?


> Functionally, the point is that SBCs have to be aware, for each
> network segment that it is connected to, whether GRUU URIs for its
> domain are valid there, and censor (or otherwise revise) GRUUs that
> would otherwise be propagated into segments they are not valid for.

That's not very practical. It requires the admin to know things which =
they are unlikely to know, or to know accurately enough to be safe and =
not cause problems in scenarios they did not foresee.  And it's not just =
SBCs - we're talking basically all B2BUAs.

-hadriel


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On Mar 21, 2014, at 3:31 PM, Dale R. Worley <worley@ariadne.com> wrote:

> [moved from Dispatch]
>=20
>> From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
>=20
>> There is no problem I know of, so long as you do NOT use GRUUs, or
>> ignore the 'gr' param and just behave like they're normal contact
>> URIs.
>=20
> I'm curious what you mean by "just behave like they're normal contact
> URIs".  My understanding is that the contact URI is used for whatever
> it's used for *because* it's the contact URI.  If it's marked as a
> GRUU, that presumably is a hint that your odds of success are higher,
> but what would you do differently?

You would send out-of-dialog REFERs, and you could (in theory) store the =
GRUU for future use indefinitely, such as in an address book.


> In your draft, you also write:
>=20
>   Instead, a logically similar model has been successfully deployed=20
>   using "Global Contacts", which are Contact URIs having the=20
>   properties of a GRUU, but not being a GRUU in syntax (i.e., without=20=

>   a 'gr' param, and not indicating an AoR).  This has the benefit that=20=

>   downstream B2BUAs/UAs will not treat the Contact as a special GRUU=20=

>   case. Global Contacts are not always valid after their dialog is=20
>   terminated, but it remains to be seen if this will cause problems in=20=

>   the future.  Currently the main use-case for GRUU-like properties is=20=

>   for call-transfer scenarios, for which dialog lifetime is=20
>   sufficient.=20
>=20
> Could you give us some detail as to what a "global contact" is and how
> it works?

It's just a Contact URI that encodes information in itself such that =
it's uniquely related to the original Contact for the original dialog.

So if an SBC gets a INVITE with a Contact URI of 'sip:alice@foo.com', =
then the new Contact URI the SBC uses in its out-going INVITE could be =
something like 'sip:akcjbaidch@sbc1.foo.com', whereas one from =
'sip:bob@foo.com' would produce 'sip:ucdhwbcu@sbc1.foo.com' or whatever. =
(although it may not be a hostname, but an IP instead)

The userinfo portion has sufficient information for the SBC to figure =
out which original Contact URI to use as a target for an out-of-dialog =
request (or to match to some internal dialog state).  In other words =
it's a way of minting a "garden-scoped" GRUU basically, but without =
using a 'gr' param so downstream nodes don't behave differently than =
before.

-hadriel


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On Mar 21, 2014, at 4:08 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> I see two distinct issues here:
>=20
> - uniqueness of addresses (both DNS and IP)
> - reachability
>=20
> IMO, when we are talking about gruus, what we are really talking about =
is primarily global uniqueness. The registrar-provided gruus are =
intended to *help* with this for devices behind NATs, so that they can =
get a globally unique address.
>=20
> Even with globally unique addresses there is no guarantee of =
reachability. Some devices may have a policy about who may reach them, =
and how. If their globally reachable address is used some other way they =
may block the request. And they may utilize intermediaries to help them =
enforce the policy. I don't think SIP considers such behavior wrong.
>=20
> But, if UA-B intends that it be possible for UA-A to be able to send =
out-of-dialog requests to it, then when it provides a contact address, =
it should provide one that will *work* for UA-A to reach UA-B =
out-of-dialog. This should be part of the sip contract, and GRUU is an =
assistance to achieving that.

And how could UA-B possibly know whether a URI it provides is usable by =
UA-A??  We're talking machines, not God. :)

Even UA-B's Registrar, and the human operator of UA-B's domain, don't =
actually know whether UA-A can use it if UA-A isn't in their domain.

-hadriel


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Subject: Re: [sipcore] Problems with GRUUs
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On 3/22/14 12:29 PM, Hadriel Kaplan wrote:
>
> On Mar 21, 2014, at 4:08 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> I see two distinct issues here:
>>
>> - uniqueness of addresses (both DNS and IP)
>> - reachability
>>
>> IMO, when we are talking about gruus, what we are really talking about is primarily global uniqueness. The registrar-provided gruus are intended to *help* with this for devices behind NATs, so that they can get a globally unique address.
>>
>> Even with globally unique addresses there is no guarantee of reachability. Some devices may have a policy about who may reach them, and how. If their globally reachable address is used some other way they may block the request. And they may utilize intermediaries to help them enforce the policy. I don't think SIP considers such behavior wrong.
>>
>> But, if UA-B intends that it be possible for UA-A to be able to send out-of-dialog requests to it, then when it provides a contact address, it should provide one that will *work* for UA-A to reach UA-B out-of-dialog. This should be part of the sip contract, and GRUU is an assistance to achieving that.
>
> And how could UA-B possibly know whether a URI it provides is usable by UA-A??  We're talking machines, not God. :)

OK, maybe I didn't phrase that well. UA-B can't be expected to do 
anything special to deal with restrictions that may be imposed on UA- by 
virtue of UA-A's environment.

What I meant was that UA-B needs to provide a contact that deals with 
any limitations imposed by UA-B's environment.

IOW, UA-B should be handing out a contact that can be successfully used 
out of dialog by UA-A if UA-A resides directly on the public internet.

> Even UA-B's Registrar, and the human operator of UA-B's domain, don't actually know whether UA-A can use it if UA-A isn't in their domain.

If we have two UA's each in restricted domains/gardens, then each side 
should be doing the necessary fixups. And each doesn't need to do it 
directly - it can be delegated to other servers in its environment. 
AFAIK that *is* (more or less) the model that SBCs are aiming for. But 
apparently preserving global out-of-dialog routability isn't always 
being observed.

IMO, saying "we only preserve in-dialog signaling" doesn't cut it.

I expect your servers *can* do this, but it is up to operators to 
configure them properly to do so. If so, maybe it is more of an issue of 
operator training - e.g. BCPs.

	Thanks,
	Paul


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Subject: [sipcore] Actions for DANE SIP
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On 3/22/14 3:33 AM, Olle E. Johansson wrote:

>> * draft-johansson-dispatch-dane-sip:
>>
>> Conclusion: Few present understood the issue. Many were confused.
>>
>> Action: Olle to send use cases to the list.
> I don't think the confusion was about use cases at all or whether we need DANE or not - it was about whether the DANE group documents was enough,
> if this is just a bump in the TLS stack or if we actually need to do something in SIPcore. Some people claimed
> we need to do zero, nothing, nada. Some people (including me) claims we do need to document this,
> especially since the SIP Domain certificate RFC is an update to RFC 3261 and DANE usage changes
> the certificate contents.
>
> The DANE group is very clear that we do need to document our usage. Wes Hardaker pointed that out
> several times during the discussion. It's documented in the DANE SRV drafts too.

OK. Maybe "use cases" isn't a proper characterization of what we need.
But clearly we have a problem around lack of understanding. And it isn't 
entirely clear exactly *what* people lack an understanding of.

My guess is that we have people with a range of understandings of DANE. 
And so a range of understandings of what we need to do.

IMO the starting point is to clarify why DANE is important to SIP. I see 
a couple of reasons, neither specific to sip, but relevant to sip:
- reliance on DNSSEC is more trustworthy than reliance on CAs
- server certificates are more easily managed than domain certs

I think your draft would benefit by having a section specifically 
addressing this. But it might be helpful to have some list discussion on 
it before you update the draft.

Once we get past "why DANE?", we then get to "what do we need to do 
before we can use DANE with sip?" Your draft addresses that, but 
obviously some people disagree on how much needs to be specified, and 
how much can be had by reference to existing generic documents. I doubt 
that big changes are needed in what you have already for this, but maybe 
it can be clarified to address the concerns that were raised - to make 
it crystal clear. Referencing the the explicit places where the generic 
DANE documents say that we must document the sip-specific usage would 
help here.

Does this match your understanding of what happened?
If so, I'll tweak the actions to agree.

	Thanks,
	Paul


From nobody Tue Mar 25 08:41:55 2014
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
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Subject: [sipcore] Passing of Francois Audet - Additional information
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--bcaec53f35f54a501b04f5702fe4
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Hi all,

I have some additional information to share that I thought would be of
interest to a number of people.

1) There have been some queries as to specifically how one can help the
family directly.  A Special Needs Trust has been established for Francois'
son, Olivier. This fund will be used to offset the costs of Olivier's
treatments that aren't covered by medical insurance or other funding.  If
you are interested in making a donation to this fund, please contact me
directly for further details.

2) The funeral is scheduled as follows:
Saturday, April 5 at 14:00 hours, at the funeral Cooperative in the
Outaouais, 95 boulevard Cit=E9-des-Jeunes, Gatineau (QC) J8Y 6 X 3

3) If you have any photos of Francois from meetings, etc, can you please
forward those to me?  I am putting together a scrapbook for Olivier. Also,
if you want to include a personal note, please send that.

Thanks for all your support,
Mary.



On Mon, Mar 17, 2014 at 10:56 AM, Mary Barnes <mary.ietf.barnes@gmail.com>w=
rote:

> A number of you all will remember Francois Audet, who was a colleague of
> mine at Nortel and a regular IETF attendee (until he went to Skype ;)  He
> was a co-author on the recently published 4244bis drafts.  He unfortunate=
ly
> lost his battle with cancer on Friday.  I have received the following
> information from a family friend via Facebook:
>
> Thanks for all your support messages, both on & offline.
> I've talked to Genevi=E8ve over the week-end, and she reminded me that th=
ere
> is just no real treatment of Brain Tumors, and that form of cancer is
> growing at some alarming rate.
> She agreed that the best support message would be to sponsor the research=
,
> for which donations can be made as a memorial gift for Francois Audet.
> I've selected a few organizations:
> - Stanford Brain Tumor Center (http://medicalgiving.stanford.edu/, this
> is where Fran=E7ois was treated. Make sure to indicate the Brain Tumor Ce=
nter
> as the recipient).
> - UCSF Brain Tumor Center (
> https://makeagift.ucsf.edu/site/SPageServer?pagename=3DA1_API_GordonMurra=
yGivingForm&ACode=3DB2934
> )
> - National Brain Tumor Society (http://www.braintumor.org/)
>
> Note that he also left behind a son, Olivier.
>
> That's all the information I have at this point.
>
> Regards,
> Mary.
>

--bcaec53f35f54a501b04f5702fe4
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<div dir=3D"ltr">Hi all,<div><br></div><div>I have some additional informat=
ion to share that I thought would be of interest to a number of people.</di=
v><div><br></div><div>1) There have been some queries as to specifically ho=
w one can help the family directly. =A0A Special Needs Trust has been estab=
lished for Francois&#39; son, Olivier. This fund will be used to offset the=
 costs of Olivier&#39;s treatments that aren&#39;t covered by medical insur=
ance or other funding. =A0If you are interested in making a donation to thi=
s fund, please contact me directly for further details.</div>
<div><br></div><div>2) The funeral is scheduled as follows:</div><div><span=
 style=3D"font-size:13.333333015441895px;font-style:normal;font-variant:nor=
mal;font-weight:normal;letter-spacing:normal;line-height:18px;text-align:le=
ft;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;=
background-color:rgb(255,255,255);display:inline!important;float:none"><fon=
t class=3D"" face=3D"arial, helvetica, sans-serif" color=3D"#000000">Saturd=
ay, April 5 at 14:00 hours, at the funeral Cooperative in the Outaouais, 95=
 boulevard Cit=E9-des-Jeunes, Gatineau (QC) J8Y 6 X 3</font></span><br>
</div><div><br></div><div>3) If you have any photos of Francois from meetin=
gs, etc, can you please forward those to me? =A0I am putting together a scr=
apbook for Olivier. Also, if you want to include a personal note, please se=
nd that. =A0</div>
<div><br></div><div>Thanks for all your support,</div><div>Mary.=A0</div><d=
iv>=A0=A0<br><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">=
On Mon, Mar 17, 2014 at 10:56 AM, Mary Barnes <span dir=3D"ltr">&lt;<a href=
=3D"mailto:mary.ietf.barnes@gmail.com" target=3D"_blank">mary.ietf.barnes@g=
mail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin-top:0px;margin-right:0px;=
margin-bottom:0px;margin-left:0.8ex;border-left-width:1px;border-left-color=
:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir=3D"ltr=
">
A number of you all will remember Francois Audet, who was a colleague of mi=
ne at Nortel and a regular IETF attendee (until he went to Skype ;) =A0He w=
as a co-author on the recently published 4244bis drafts. =A0He unfortunatel=
y lost his battle with cancer on Friday. =A0I have received the following i=
nformation from a family friend via Facebook:<div>

<br></div><div><div>Thanks for all your support messages, both on &amp; off=
line.</div><div>I&#39;ve talked to Genevi=E8ve over the week-end, and she r=
eminded me that there is just no real treatment of Brain Tumors, and that f=
orm of cancer is growing at some alarming rate.</div>

<div>She agreed that the best support message would be to sponsor the resea=
rch, for which donations can be made as a memorial gift for Francois Audet.=
</div><div>I&#39;ve selected a few organizations:</div><div>- Stanford Brai=
n Tumor Center (<a href=3D"http://medicalgiving.stanford.edu/" target=3D"_b=
lank">http://medicalgiving.stanford.edu/</a>, this is where Fran=E7ois was =
treated. Make sure to indicate the Brain Tumor Center as the recipient).</d=
iv>

<div>- UCSF Brain Tumor Center (<a href=3D"https://makeagift.ucsf.edu/site/=
SPageServer?pagename=3DA1_API_GordonMurrayGivingForm&amp;ACode=3DB2934" tar=
get=3D"_blank">https://makeagift.ucsf.edu/site/SPageServer?pagename=3DA1_AP=
I_GordonMurrayGivingForm&amp;ACode=3DB2934</a>)</div>

<div>- National Brain Tumor Society (<a href=3D"http://www.braintumor.org/"=
 target=3D"_blank">http://www.braintumor.org/</a>)</div></div><div><br></di=
v><div>Note that he also left behind a son, Olivier.=A0</div><div><br></div=
>
<div>That&#39;s all the information I have at this point.=A0</div>
<div><br></div><div>Regards,</div><div>Mary.</div></div>
</blockquote></div><br></div></div></div>

--bcaec53f35f54a501b04f5702fe4--


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Subject: [sipcore] Fwd: [RAI] SIPit 31 registration is open
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I didn't see this hit either SIPCORE or DISPATCH WG mailing lists and I
know that the RAI mailing list has far fewer subscribers than these other
two lists.

Obviously, don't reply to this message on these other lists.

---------- Forwarded message ----------
From: Robert Sparks <rjsparks@nostrum.com>
Date: Thu, Mar 20, 2014 at 12:42 PM
Subject: [RAI] SIPit 31 registration is open
To: rai@ietf.org


SIPit 31 will be held in Nice, France Sep 28 - Oct 3, 2014.
The event will be hosted by ETSI.

More information, and the registration page,  is available at
<http://www.etsi.org/news-events/events/750-sipit-31>
Please take advantage of the opportunity to register early.

In addition to the normal team-to-team testing at the event,
we will be concentrating on several specific interoperability areas:
* Nat and Firewall Traversal using Outbound, TURN, and STUN
* RTCWeb
* Advanced Video scenarios
* Use of TLS and DTLS

If you're planning to attend and have other areas you would like
have a multi-team focus session around, please let me know.

RjS

_______________________________________________
RAI mailing list
RAI@ietf.org
https://www.ietf.org/mailman/listinfo/rai

--bcaec53f35f5304c7904f5993cb7
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<div dir=3D"ltr">I didn&#39;t see this hit either SIPCORE or DISPATCH WG ma=
iling lists and I know that the RAI mailing list has far fewer subscribers =
than these other two lists.<div><br></div><div>Obviously, don&#39;t reply t=
o this message on these other lists.=A0<br>
<br><div class=3D"gmail_quote">---------- Forwarded message ----------<br>F=
rom: <b class=3D"gmail_sendername">Robert Sparks</b> <span dir=3D"ltr">&lt;=
<a href=3D"mailto:rjsparks@nostrum.com">rjsparks@nostrum.com</a>&gt;</span>=
<br>
Date: Thu, Mar 20, 2014 at 12:42 PM<br>Subject: [RAI] SIPit 31 registration=
 is open<br>To: <a href=3D"mailto:rai@ietf.org">rai@ietf.org</a><br><br><br=
>SIPit 31 will be held in Nice, France Sep 28 - Oct 3, 2014.<br>
The event will be hosted by ETSI.<br>
<br>
More information, and the registration page, =A0is available at<br>
&lt;<a href=3D"http://www.etsi.org/news-events/events/750-sipit-31" target=
=3D"_blank">http://www.etsi.org/news-<u></u>events/events/750-sipit-31</a>&=
gt;<br>
Please take advantage of the opportunity to register early.<br>
<br>
In addition to the normal team-to-team testing at the event,<br>
we will be concentrating on several specific interoperability areas:<br>
* Nat and Firewall Traversal using Outbound, TURN, and STUN<br>
* RTCWeb<br>
* Advanced Video scenarios<br>
* Use of TLS and DTLS<br>
<br>
If you&#39;re planning to attend and have other areas you would like<br>
have a multi-team focus session around, please let me know.<br>
<br>
RjS<br>
<br>
______________________________<u></u>_________________<br>
RAI mailing list<br>
<a href=3D"mailto:RAI@ietf.org" target=3D"_blank">RAI@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rai" target=3D"_blank">htt=
ps://www.ietf.org/mailman/<u></u>listinfo/rai</a><br>
</div><br></div></div>

--bcaec53f35f5304c7904f5993cb7--


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Subject: Re: [sipcore] [dispatch] Comments on draft-allen-dispatch-routing-out-of-dialog-request-00
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> On Mar 13, 2014, at 11:01 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> The problem is when call flows break even when all the participants are
> legitimate - when the providers are creating a walled garden that only works
> for the specific call flows they have anticipated and made specific
> provision for. That breaks innovation, and drives things like webrtc.


> On 3/13/14 2:10 PM, Hadriel Kaplan wrote:
>
> You realize the irony in that last statement right?  Webrtc is
> by-definition a walled-garden. :)

Hmmm, I read that as "when the providers are creating a walled garden
that only works for the specific call flows they have anticipated and
made specific provision for. That [sort of thinking] drives things
like webrtc."


> From: Michael Hammer <michael.hammer@yaanatech.com>
> 
> The real irony would be if the WebRTC servers interconnected to each other 
> using a widely-available IMS and thereby become more open.
> 
> One walled-garden turning the other walled-garden inside out.  :)


> From: Christer Holmberg <christer.holmberg@ericsson.com>
> 
> Well, your dream is about to become true, as 3GPP is currently
> defining WebRTC access to IMS :)

OK, *now* my head is starting to hurt!

Dale


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Subject: Re: [sipcore] [dispatch] draft-allen-dispatch-routing-out-of-dialog-request-00
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> From: Christer Holmberg <christer.holmberg@ericsson.com>
> 
> If the [deliverable] would be an SBC BCP document, I guess it could also
> be done in STRAW.

It makes sense to me that to the degree that the conclusion/consensus
is recommendations about SBC procedures, we should coordinate with
STRAW, or transfer the work entirely to STRAW.  (Are there any members
of the STRAW mailing list who aren't members of the SIPCORE mailing
list?)

Dale

